Waves Plug-in for Vocals and Monophonic Specifications

Waves Plug-in for Vocals and Monophonic Specifications
Audio Plug-Ins Guide
Version 10.0
Legal Notices
This guide is copyrighted ©2011 by Avid Technology, Inc.,
(hereafter “Avid”), with all rights reserved. Under copyright
laws, this guide may not be duplicated in whole or in part
without the written consent of Avid.
003, 96 I/O, 96i I/O, 192 Digital I/O, 192 I/O, 888|24 I/O,
882|20 I/O, 1622 I/O, 24-Bit ADAT Bridge I/O, AudioSuite,
Avid, Avid DNA, Avid Mojo, Avid Unity, Avid Unity ISIS,
Avid Xpress, AVoption, Axiom, Beat Detective,
Bomb Factory, Bruno, C|24, Command|8, Control|24,
D-Command, D-Control, D-Fi, D-fx, D-Show, D-Verb, DAE,
Digi 002, DigiBase, DigiDelivery, Digidesign,
Digidesign Audio Engine, Digidesign Intelligent Noise
Reduction, Digidesign TDM Bus, DigiDrive, DigiRack,
DigiTest, DigiTranslator, DINR, DV Toolkit, EditPack, Eleven,
HD Core, HD I/O, HD MADI, HD OMNI, HD Process, Hybrid,
Impact, Interplay, LoFi, M-Audio, MachineControl, Maxim,
Mbox, MediaComposer, MIDI I/O, MIX, MultiShell, Nitris,
OMF, OMF Interchange, PRE, ProControl, Pro Tools,
Pro Tools|HD, QuickPunch, Recti-Fi, Reel Tape, Reso,
Reverb One, ReVibe, RTAS, Sibelius, Smack!,
SoundReplacer, Sound Designer II, Strike, Structure,
SYNC HD, SYNC I/O, Synchronic, TL Aggro, TL AutoPan,
TL Drum Rehab, TL Everyphase, TL Fauxlder, TL In Tune,
TL MasterMeter, TL Metro, TL Space, TL Utilities, Transfuser,
Trillium Lane Labs, Vari-Fi, Velvet, X-Form, and XMON are
trademarks or registered trademarks of Avid Technology, Inc.
Xpand! is Registered in the U.S. Patent and Trademark Office.
All other trademarks are the property of their respective
owners.
Product features, specifications, system requirements, and
availability are subject to change without notice.
Guide Part Number 9329-65100-00 REV A 10/11
Documentation Feedback
At Avid, we are always looking for ways to improve our
documentation. If you have comments, corrections, or
suggestions regarding our documentation, email us at
[email protected]
Contents
Part I
Introduction to Pro Tools Plug-Ins
Chapter 1. Audio Plug-Ins Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Avid Audio Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Plug-In Formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Using Plug-Ins in Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Conventions Used in Pro Tools Documentation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
System Requirements and Compatibility for Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Contents of the Boxed Version of Your Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
About www.avid.com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Chapter 2. Installing Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Free Pro Tools Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Free VENUE Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Updating Older Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Using Pro Tools Plug-Ins with Avid Media Composer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Installing Plug-Ins for Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Installing Plug-Ins for VENUE Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Authorizing Paid Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Removing Plug-Ins for Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Removing Plug-Ins for VENUE Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Part II
EQ Plug-Ins
Chapter 3. AIR Kill EQ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
Kill EQ Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
Chapter 4. EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
EQ III Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Adjusting EQ III Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
EQ III I/O Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
Contents
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EQ III EQ Band Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
EQ III Frequency Graph Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
7 Band EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
2–4 Band EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
1 Band EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Chapter 5. JOEMEEK VC5 Meequalizer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
JOEMEEK Meequalizer Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
Chapter 6. Pultec Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
Pultec EQP-1A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
Pultec EQH-2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
Pultec MEQ-5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
Part III
Dynamics Plug-Ins
Chapter 7. BF-2A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
BF-2A Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
BF-2A Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
Chapter 8. BF-3A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
BF-3A Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
BF-3A Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Chapter 9. BF76 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
BF76 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
BF76 Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
Chapter 10. Channel Strip. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Sections and Panes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Output . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
FX Chain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
Dynamics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
EQ/Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
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Chapter 11. Dynamics III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Dynamics III Shared Features and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Compressor/Limiter III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
Expander/Gate III. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
De-Esser III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
Dynamics III Side-Chain Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Chapter 12. Fairchild Plug-Ins. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Fairchild 660 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Fairchild 670 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
Chapter 13. Impact . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Impact Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Using the Impact Compressor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
Chapter 14. JOEMEEK SC2 Compressor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
JOEMEEK Compressor Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
JOEMEEK Compressor Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Chapter 15. Maxim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
About Peak Limiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Maxim Controls and Meters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
Using Maxim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Maxim and Mastering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Chapter 16. Purple Audio MC77 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Chapter 17. Slightly Rude Compressor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
Slightly Rude Compressor Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
Slightly Rude Compressor Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
Chapter 18. Smack! . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
Smack! Controls and Meters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
Using the Smack! Compressor/Limiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Chapter 19. TL Aggro. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
TL Aggro Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
TL Aggro Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Using the TL Aggro Side-Chain Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Contents
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Part IV
Pitch Shift Plug-Ins
Chapter 20. AIR Frequency Shifter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
Frequency Shifter Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
Chapter 21. Pitch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
Pitch Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
Relative Pitch Entry (Musical Staff) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
Chapter 22. Pitch Shift . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
Pitch Shift Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
Chapter 23. Time Shift. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
Time Shift Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
AudioSuite Input Modes and Time Shift . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
AudioSuite Preview and Time Shift . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Time Shift as AudioSuite TCE Plug-In Preference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Processing Audio Using Time Shift . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Post Production Pull Up and Pull Down Tasks with Time Shift . . . . . . . . . . . . . . . . . . . . . . 134
Chapter 24. Vari-Fi. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
Chapter 25. X-Form . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
X-Form Displays and Controls Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
X-Form AudioSuite Input Modes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
AudioSuite TCE Plug-In Preference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Processing Audio Using X-Form . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
Using X-Form for Post Production Pull Up and Pull Down Tasks . . . . . . . . . . . . . . . . . . . . 143
Part V
Reverb Plug-Ins
Chapter 26. AIR Non-Linear Reverb. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
Chapter 27. AIR Reverb. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
Reverb Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
Chapter 28. AIR Spring Reverb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
Spring Reverb Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
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Chapter 29. D-Verb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
D-Verb Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
Chapter 30. Reverb One. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
A Reverb Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
Reverb One Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
Reverb One Graphs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 165
Other Reverb One Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Chapter 31. ReVibe. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Reverberation Concepts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
Using ReVibe . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
Adjusting ReVibe Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
ReVibe Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173
ReVibe Decay Color & EQ Section Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
ReVibe Contour Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
ReVibe Input/Output Meter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
ReVibe Online Help Button . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
ReVibe Room Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
Chapter 32. TL Space TDM and TL Space Native . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189
TL Space Feature Highlights . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
TL Space Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
TL Space and System Performance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 194
Impulse Response (IR) and TL Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
TL Space Presets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
TL Space Snapshots . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
TL Space Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
TL Space Display Area. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
TL Space IR Browser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
TL Space Primary Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208
TL Space Group Selectors and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 209
TL Space Info Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
Using TL Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 212
TL Space IR Library . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
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Part VI
Delay Plug-Ins
Chapter 33. AIR Dynamic Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
Dynamic Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
Chapter 34. AIR Multi-Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 221
Multi-Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 221
Chapter 35. Mod Delay II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223
Mod Delay II Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
Multichannel Mod Delay II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Selecting Audio for ModDelay II AudioSuite Processing . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Chapter 36. Mod Delay III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
Mod Delay III Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
Selections for Mod Delay III AudioSuite Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
Chapter 37. Moogerfooger Analog Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 231
Moogerfooger Analog Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 232
Chapter 38. Multi-Tap Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
Multi-Tap Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
Chapter 39. Ping-Pong Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Ping-Pong Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Chapter 40. Reel Tape Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
Reel Tape Common Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 238
Reel Tape Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 238
Chapter 41. Tel-Ray Variable Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 241
Tel-Ray Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 242
Tel-Ray Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 242
Chapter 42. TimeAdjuster . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
TimeAdjuster Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Using TimeAdjuster for Manual Delay Compensation . . . . . . . . . . . . . . . . . . . . . . . . . . . . 244
When to Compensate for Delays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 245
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Audio Plug-Ins Guide
Part VII
Modulation Plug-Ins
Chapter 43. AIR Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 249
AIR Chorus Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 249
Chapter 44. AIR Ensemble. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
Ensemble Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
Chapter 45. AIR Filter Gate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
Filter Gate Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
Chapter 46. AIR Flanger. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255
AIR Flanger Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255
Chapter 47. AIR Fuzz-Wah. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
Fuzz-Wah Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
Chapter 48. AIR Multi-Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 259
Multi-Chorus Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 259
Chapter 49. AIR Phaser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 261
Phaser Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 261
Chapter 50. AIR Talkbox . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
Talkbox Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
Chapter 51. AIR Vintage Filter. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Vintage Filter Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Chapter 52. Cosmonaut Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Cosmonaut Voice Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Chapter 53. Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Chorus Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Chapter 54. Flanger . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Flanger Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Chapter 55. Moogerfooger Lowpass Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Chapter 56. Moogerfooger 12-Stage Phaser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 277
Chapter 57. Moogerfooger Ring Modulator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
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Chapter 58. Reel Tape Flanger . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Reel Tape Common Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
Reel Tape Flanger Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
Chapter 59. Sci-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
Sci-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
Chapter 60. TL EveryPhase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 293
TL EveryPhase Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 293
TL EveryPhase Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 294
Using TL EveryPhase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 299
Chapter 61. Voce Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
Voce Chorus/Vibrato . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
Voce Spin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
Part VIII
Harmonic Plug-Ins
Chapter 62. AIR Distortion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
Distortion Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
Chapter 63. AIR Enhancer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Enhancer Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Chapter 64. AIR Lo Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
AIR Lo Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
Chapter 65. Lo-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Lo-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Chapter 66. Recti-Fi. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 323
Recti-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 324
Chapter 67. Reel Tape Saturation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
Reel Tape Common Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
Reel Tape Saturation Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 328
Chapter 68. SansAmp PSA-1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 331
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Audio Plug-Ins Guide
Part IX
Noise Reduction Plug-Ins
Chapter 69. DINR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
How Broadband Noise Reduction Works . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
BNR Spectral Graph . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 336
Broadband Noise Reduction Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 337
Using Broadband Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 341
Using BNR AudioSuite . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 344
Part X
Dither Plug-Ins
Chapter 70. Dither . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Dither Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Chapter 71. POW-r Dither . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
POW-r Dither Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
Part XI
Sound Field Plug-Ins
Chapter 72. AIR Stereo Width . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 355
Stereo Width Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 355
Chapter 73. Down Mixer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 357
Chapter 74. SignalTools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 359
SignalTools SurroundScope . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 359
SignalTools PhaseScope . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 360
SignalTools Display Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 361
SignalTools Level Meters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 362
Chapter 75. TL AutoPan. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
TL AutoPan Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
Using TL AutoPan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 370
Part XII
Instrument Plug-Ins
Chapter 76. Boom. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 375
Boom Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376
Inserting Boom on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
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Creating a Drum Pattern Using Boom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 380
Saving a Boom Pattern as a Preset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 380
Playing with Patterns in Boom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 381
Controlling Boom with MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 381
Playing Boom Patterns Using MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 382
Creating Boom Pattern Chains. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383
Using the MIDI Learn Function on Avid Virtual Instruments . . . . . . . . . . . . . . . . . . . . . . . . 383
Chapter 77. Bruno and Reso . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
Bruno/Reso Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
Bruno/Reso DSP Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 386
Inserting Bruno/Reso onto an Audio Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 386
Playing Bruno/Reso . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 386
Using an External Key Input with Bruno/Reso . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 387
Bruno Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
Reso Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
Chapter 78. Click . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 401
Click Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 401
Creating a Click Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 401
Chapter 79. DB-33 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 403
DB-33 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 403
Inserting DB-33 on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 408
Using the MIDI Learn Function on Avid Virtual Instruments . . . . . . . . . . . . . . . . . . . . . . . . 408
Chapter 80. Mini Grand . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 411
Mini Grand Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 411
Inserting Mini Grand on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 413
Using the MIDI Learn Function on Avid Virtual Instruments . . . . . . . . . . . . . . . . . . . . . . . . 414
Chapter 81. Structure Free . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 417
Structure Free Keyboard Section Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 417
Structure Free Patch List Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 419
Structure Free Main Page Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 421
Structure Free Browser Page Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 423
Using Structure Free . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 424
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Chapter 82. TL Drum Rehab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 429
TL Drum Rehab Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 430
TL Drum Rehab Controls and Displays Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 434
TL Drum Rehab Main Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 434
TL Drum Trigger Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 435
TL Drum Rehab Expert Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 440
Samples Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 444
TL Drum Rehab Preferences Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . 446
TL Drum Rehab Library Browser. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 446
Chapter 83. TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 451
Configuring Pro Tools for Use with TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 451
TL Metro Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 452
Synchronizing TL Metro to Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 453
Customizing TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 454
Chapter 84. Vacuum. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 457
Vacuum Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 458
Inserting Vacuum on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 464
Using the MIDI Learn Function on Avid Virtual Instruments . . . . . . . . . . . . . . . . . . . . . . . . 464
Chapter 85. Xpand!2. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 467
Xpand!2 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 468
Xpand!2 Patch Edit Controls Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 471
Xpand!2 Play Patch Edit Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 471
Xpand!2 Arp Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 472
Xpand!2 Mod Patch Edit Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 473
Inserting Xpand!2 on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 475
Using the MIDI Learn Function on Avid Virtual Instruments . . . . . . . . . . . . . . . . . . . . . . . . 475
Chapter 86. ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 477
ReWire Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
Using ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
MIDI Automation with ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 481
Quitting ReWire Client Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 482
Session Tempo and Meter Changes and ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
Looping Playback with ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
Automating Input Switching with ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 484
Contents
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Part XIII
Other Plug-Ins
Chapter 87. BF Essentials Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 487
BF Essential Clip Remover . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 487
BF Essential Correlation Meter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
BF Essential Meter Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
BF Essential Noise Meter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
Chapter 88. Signal Generator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
Signal Generator Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
AudioSuite Processing with Signal Generator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 490
Chapter 89. SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 491
Audio Replacement Techniques . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 491
SoundReplacer Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 492
Using SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 496
Getting Optimum Results with SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 497
Using the Audio Files Folder for Frequently Used SoundReplacer Files . . . . . . . . . . . . . . . 499
Chapter 90. Time Compression/Expansion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 501
Time Compression/ Expansion Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 501
Chapter 91. TL InTune. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 503
TL InTune Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 504
TL InTune Presets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
Using TL InTune . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 508
Chapter 92. TL MasterMeter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 509
TL Master Meter Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 510
Using TL MasterMeter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 513
TL MasterMeter Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 514
Chapter 93. TL Metro. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 517
Configuring Pro Tools for Use with TL Metro. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 517
TL Metro Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 518
Synchronizing TL Metro to Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 519
Customizing TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 520
Chapter 94. Trim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 523
Trim Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 523
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Chapter 95. Other AudioSuite Plug-In Utilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
DC Offset Removal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Duplicate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Gain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 526
Invert . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 526
Normalize . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 527
Reverse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
Part XIV
Eleven
Chapter 96. Eleven and Eleven Free. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 531
Chapter 97. Eleven Input Calibration and QuickStart . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 533
1: Connect your Guitar and Configure Source Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 534
2: Set Hardware and Levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 534
3: Set Up a Pro Tools Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 535
4. Set Up Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
5. Getting Started Playing Music with Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 537
Chapter 98. Using Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 539
Inserting Eleven on Tracks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 539
Adjusting Eleven’s Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 539
Using a Pro Tools Worksurface with Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 540
Using MIDI and MIDI Learn with Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 540
Eleven Settings (Presets) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 541
Master Section . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 541
Amp Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 543
Eleven Amp Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Eleven Cabinet Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 546
Eleven Cabinet Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 547
Tracks and Signal Routing for Guitar. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 548
Blending Eleven Cabinets and Amps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 552
Eleven Tips and Suggestions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 556
Eleven Signal Flow Notes. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 559
Contents
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Part XV
Synchronic
Chapter 99. Synchronic Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 563
Chapter 100. Synchronic Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 565
Synchronic Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 565
Playing Synchronic RTAS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 566
Performance and Edit Modes. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 569
Synchronic Performance Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 569
Synchronic Presets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 570
Chapter 101. Using Synchronic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 573
Adjusting Synchronic Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 573
Synchronic Sound Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 574
Synchronic Sound Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 574
Synchronic Sound Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 575
Synchronic Playback Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 581
Synchronic Playback Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 581
Synchronic Playback Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 582
Synchronic Effect Module . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 590
Synchronic Effect Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 590
Synchronic Effect Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 591
Synchronic XFade Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 596
Synchronic MIDI Module Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 598
Synchronic MIDI Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 598
Synchronic MIDI Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 600
Synchronic Keyboard Focus Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 601
Chapter 102. Using Synchronic as an AudioSuite Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . 603
Synchronic AudioSuite Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 603
Synchronic AudioSuite Workflow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 605
Chapter 103. Automating Synchronic RTAS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 609
Using Automation Playlists . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 609
Using MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 611
Chapter 104. Synchronic Plug-In Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Imported Audio Stored with Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 615
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Audio Plug-Ins Guide
Part I: Introduction to
Pro Tools Plug-Ins
Chapter 1: Audio Plug-Ins Overview
Plug-Ins are special-purpose software components that provide additional signal processing
and other functionality to Pro Tools ®. These include plug-ins that come with your Pro Tools
system, as well as many other plug-ins that can
be added to your system.
Additional plug-ins are available both from
Avid and our third-party developers. See
the documentation that came with the
plug-in for operational information.
Free Sound-Processing, Effects, and Utility
Plug-Ins:
• AIR Chorus
• AIR Distortion
• AIR Dynamic Delay
• AIR Enhancer
• AIR Ensemble
• AIR Filter Gate
• AIR Flanger
• AIR Frequency Shifter
Avid Audio Plug-Ins
®
Avid provides a comprehensive set of digital
signal processing tools for professional audio
production. A set of free virtual instrument and
sound processing, effects, and utility plug-ins
are included with Pro Tools. Other Avid plugins are available for purchase or rental from the
Avid store (http://shop.avid.com, or choose
Marketplace > Plug-Ins in Pro Tools).
• AIR Fuzz-Wah
• AIR Kill EQ
• AIR Lo-Fi
• AIR MultiChorus
• AIR Multi-Delay
• AIR Nonlinear Reverb
• AIR Phaser
• AIR Reverb
Free Avid Plug-Ins
• AIR Spring Reverb
A selection of free Avid plug-ins are included
with Pro Tools. These basic plug-ins provide a
comprehensive suite of digital signal processing
effects that include EQ, dynamics, delay, and
other essential audio processing tools.
• AIR Stereo Width
• AIR Talkbox
• AIR Vintage Filter
• Avid Channel Strip
• Avid Down Mixer
All of these plug-ins are installed when you
select the “Avid Effects” option when installing Pro Tools. For more information, see the
Pro Tools Installation Guide.
• BF 76
Chapter 1: Audio Plug-Ins Overview
3
• BF Essentials utility plug-ins
• SansAmp PSA-1
• Essential Clip Remover
• Signal Generator
• Essential Correlation Meter
• SignalTools
• Essential Meter Bridge
• SurroundScope
• Essential Noise Meter
• PhaseScope
• Click
• TimeAdjuster
• D-Fi plug-ins
• Time Compression/Expansion
• Lo-Fi ™
• Time Shift
• Recti-Fi ™
• TL AutoPan ™
• Sci-Fi
™
• Vari-Fi ™
• D-fx plug-ins
• Chorus
• Flanger
• Multi-Tap Delay
• Ping-Pong Delay
• Dither
• D-Verb
• Dynamics III
• Compressor/Limiter
• TL InTune ™
• TL MasterMeter ™
• TL Metro ™
• Trim
• Other AudioSuite Plug-In Utilities
• DC Offset Removal
• Duplicate
• Gain
• Invert
• Normalize
• Reverse
• Expander/Gate
• De-Esser
• Eleven Free ™ guitar amp modeling plug-in
• 2–4 Band
The following virtual instrument plug-ins are
included with Pro Tools, but require separate
installation using the Avid Virtual Instruments
installer (available on the Pro Tools DVD and
online).
• 1 Band
• Boom drum machine and sequencer
• EQ III
• 7 Band
• Maxim ™
• Mod Delay II
• DB-33 tonewheel organ emulator with
rotating speaker simulation
• Mod Delay III
• Mini Grand acoustic grand piano
• Pitch
• Structure Free sample player
• Pitch Shift
• Vacuum mono vacuum tube synthesizer
• POW-r Dither
• Xpand! 2 multitimbral synth and sample
workstation
• ReWire
4
Free Avid Virtual Instruments Plug-Ins
Audio Plug-Ins Guide
Avid Plug-Ins for Purchase
• TL EveryPhase ™
The following plug-ins are available separately
for purchase or rental:
• TL Space ™ TDM and TL Space Native
• Voce Spin
• Bomb Factory BF-3A
• Voce Chorus/Vibrato
• Bomb Factory BF-2A
• X-Form ™ high-quality time compression and
expansion plug-in
™
™
• Bruno & Reso cross-synthesis plug-ins
• Cosmonaut Voice
• DINR ™ intelligent noise reduction
• Eleven ™ guitar amplifier modeling plug-in
• Fairchild 660 and 670
• Impact ®
Plug-In Formats
There are three plug-in formats used in
Pro Tools:
• JOEMEEK SC2 Compressor
• AudioSuite ™ plug-ins (non-real-time,
file-based processing)
• JOEMEEK VC5 Meequalizer
• Native, real-time, host-based plug-ins:
• Moogerfooger plug-ins
• Moogerfooger Analog Delay
• Moogerfooger Ring Modulator
• Moogerfooger 12-Stage Phaser
• Moogerfooger Lowpass Filter
• Purple Audio MC77
• Reel Tape ™ plug-ins:
• Reel Tape Saturation
• Reel Tape Delay
• Reel Tape Flanger
• Reverb One ™
• ReVibe ®
• Slightly Rude Compressor
• AAX Native plug-ins
• RTAS ® plug-ins
• DSP, real-time, TDM plug-ins (Pro Tools|HD
and VENUE only)
AudioSuite Plug-Ins
AudioSuite plug-ins are used to process and
write (“render”) audio files on disk, rather than
nondestructively in real time. Depending on
how you configure a non-real-time AudioSuite
plug-in, it either creates an entirely new audio
file, or alters the original source audio file.
AudioSuite plug-ins can be used on all Pro Tools
systems and Avid software, as well as any thirdparty software that supports AudioSuite.
• Smack! ™
• SoundReplacer ™ drum and sound replacement
plug-in
AudioSuite plug-in files may use either the
“.aax” or “.dpm” file suffix.
• Synchronic ™ beat slicing and processing
plug-in
• Tel-Ray Variable Delay
• TL Aggro ™
• TL Drum Rehab ™
Chapter 1: Audio Plug-Ins Overview
5
AAX Plug-Ins
AAX (Avid Audio Extension) plug-ins provide
real-time plug-in processing using host-based
(“Native”) processing. The AAX plug-in format
also supports AudioSuite non-real-time, filebased rendered processing.
TDM Plug-Ins
(Pro Tools|HD and VENUE Systems Only)
TDM (Time Division Multiplexing) plug-ins
provide real-time DSP-based (“DSP”) processing with Pro Tools HD software on
Pro Tools|HD hardware.
AAX plug-in files use the “.aax” file suffix.
TDM plug-in files use the “.dpm” file suffix.
RTAS Plug-Ins
The number and variety of TDM plug-ins that
can be used simultaneously in a session are limited only by the amount of DSP available. You
can increase available DSP by installing additional cards (such as HD Accel_Core ™,
HD Accel ™, HD Core ™, or HD Process ™ cards) in
your computer.
RTAS (Real-Time AudioSuite) plug-ins provide
real-time plug-in processing using host-based
(“Native”) processing. They function as track
inserts, are applied to audio during playback,
and process audio non-destructively in real
time. Processing power for RTAS plug-ins comes
from your computer. The more powerful your
computer, the greater the number and variety of
RTAS plug-ins that you can use simultaneously.
Because of this dependence on the CPU or host
processing, the more RTAS plug-ins you use concurrently in a session, the greater the impact it
will have on other aspects of your system’s performance, such as maximum track count, number of available voices, the density of edits possible, and latency in automation and recording.
RTAS plug-ins can be used with all Pro Tools
systems, as well as third-party software that
supports RTAS.
RTAS plug-in files use the “.dpm” file suffix.
TDM plug-ins can also be used with VENUE live
console systems. DSP Mix Engine cards can be
added to a VENUE FOH Rack or Mix Rack for increased TDM plug-in capability.
Using Plug-Ins in Pro Tools
Refer to the Pro Tools Reference Guide for information on working with plug-ins, including:
• Inserting plug-ins on tracks
• Plug-In Window controls
• Adjusting plug-in controls
• Automating plug-ins
• Using side-chain inputs
• Using plug-in presets
• Clip indicators
6
Audio Plug-Ins Guide
Conventions Used in
Pro Tools Documentation
System Requirements and
Compatibility for Plug-Ins
Pro Tools documentation uses the following
conventions to indicate menu choices, keyboard
commands, and mouse commands:
To use Pro Tools plug-ins, you need the following:
• Any of the following systems:
• An Avid-qualified system running
Pro Tools or Pro Tools HD
Convention
Action
File > Save
Choose Save from the
File menu
Control+N
Hold down the Control
key and press the N key
• A qualified Avid Media Composer ® system
(AudioSuite and RTAS only)
Control-click
Hold down the Control
key and click the mouse
button
Right-click
Click with the right
mouse button
• An Avid-qualified system and a third-party
software application that supports the
RTAS, TDM, or AudioSuite plug-in standards
The names of Commands, Options, and Settings
that appear on-screen are in a different font.
The following symbols are used to highlight
important information:
User Tips are helpful hints for getting the
most from your Pro Tools system.
Important Notices include information that
could affect your Pro Tools session data or
the performance of your Pro Tools system.
Shortcuts show you useful keyboard or
mouse shortcuts.
Cross References point to related sections in
this guide and other Avid documentation.
• A qualified Avid VENUE system
(TDM only)
• USB Smart Key (iLok), for plug-ins that can be
purchased or rented
The iLok USB Smart Key is not supplied
with plug-ins or software options. You can
use the one included with certain Pro Tools
systems (such as Pro Tools|HD-series systems), or purchase one separately.
Avid can only assure compatibility and provide
support for hardware and software it has tested
and approved.
For complete system requirements and a list of
Avid-qualified computers, operating systems,
hard drives, and third-party devices, visit:
www.avid.com/compatibility
Third-Party Plug-In Support
For information on third-party plug-ins for
Pro Tools and VENUE systems, please refer to
the documentation that came with your plug-in.
Chapter 1: Audio Plug-Ins Overview
7
Contents of the Boxed
Version of Your Plug-In
If you bought your plug-in as a boxed version, it
includes the following:
• Installation disc (for selected plug-ins)
• Activation Card with an Activation Code
for authorizing plug-ins with an iLok USB
Smart Key
About www.avid.com
The Avid website (www.avid.com) is your best
online source for information to help you get the
most out of your Pro Tools system. The following are just a few of the services and features
available.
Product Registration Register your purchase
online.
Support and Downloads Contact Avid Customer
Success (technical support); download software
updates and the latest online manuals; browse
the Compatibility documents for system requirements; search the online Knowledge Base
or join the worldwide Pro Tools community on
the User Conference.
Training and Education Study on your own using
courses available online or find out how you can
learn in a classroom setting at a certified
Pro Tools training center.
Products and Developers Learn about Avid
products; download demo software or learn
about our Development Partners and their plugins, applications, and hardware.
News and Events Get the latest news from Avid
or sign up for a Pro Tools demo.
8
Audio Plug-Ins Guide
Chapter 2: Installing Plug-Ins
Installers for your plug-ins can be downloaded
from the Avid store (store.avid.com) or can be
found on the plug-in installer disc (included
boxed versions of selected plug-ins).
An installer may also be available on the
Pro Tools installer disc or on a software bundle
installer disc.
Free Pro Tools Plug-Ins
A suite of free Avid audio effects and virtual instruments plug-ins are included with Pro Tools.
Avid Effects A set of free audio effects plug-ins
Some free Avid audio plug-ins can be downloaded from the Avid website (www.avid.com)
for use with VENUE systems, Avid
Media Composer, as well as other applications
that support AAX, AudioSuite, RTAS, or TDM
plug-in formats.
Updating Older Plug-Ins
Because plug-in installers contain the latest versions of the plug-ins, use them to update any
plug-ins you already own. When installing
Pro Tools, the Pro Tools installer automatically
updates the core Pro Tools plug-ins.
that can be installed with Pro Tools.
Avid Virtual Instruments A set of free virtual in-
strument plug-ins (including 4.4 GB of sample
content) included on the Pro Tools installation
disc and also available separately online.
For more information about installing the
Avid Effects plug-ins and the Avid Virtual
Instruments plug-ins, see the Pro Tools Installation Guide :
Free VENUE Plug-Ins
VENUE-compatible plug-ins are pre-installed
on your VENUE system and are updated when
you update VENUE software. For more information about installing VENUE software, see the
documentation that came with your VENUE
system.
Be sure to use the most recent versions of
plug-ins. For more information, see the
Avid website (www.avid.com).
Using Pro Tools Plug-Ins with
Avid Media Composer
The plug-in installation, authorization, and uninstallation processes when using Pro Tools
plug-ins with Media Composer are the same as
in Pro Tools. For more information on using
Pro Tools plug-ins with Media Composer, see
“Installing Plug-Ins for Pro Tools” on page 10,
“Authorizing Paid Plug-Ins” on page 10, and
“Removing Plug-Ins for Pro Tools” on page 12.
Chapter 2: Installing Plug-Ins
9
Installing Plug-Ins for
Pro Tools
To install a plug-in:
1
Do one of the following:
• Download the installer for your computer
platform from the Avid website
(www.avid.com). After downloading,
you may need to uncompress the installer
(.SIT on Mac or .ZIP on Windows).
Authorizing Paid Plug-Ins
Pro Tools plug-ins are authorized using the iLok
USB Smart Key (iLok), manufactured by PACE
Anti-Piracy.
iLok USB Smart Key
– or –
• Insert the installer disc that came with the
boxed version of your plug-in into your
computer.
2
Double-click the plug-in installer application.
Follow the on-screen instructions to complete
the installation.
3
4 When installation is complete, click Quit (Mac)
or Finish (Windows).
When you open Pro Tools, you are prompted to
authorize your new plug-in (see “Using Pro
Tools Plug-Ins with Avid Media Composer” on
page 9).
Installing Plug-Ins for VENUE
Systems
Installers for VENUE plug-ins can be downloaded from www.avid.com. After downloading,
the installer must be transferred to either a USB
drive or a CD-ROM. Plug-Ins can then be installed using a USB drive connected to the USB
ports on any VENUE system, or using a CDROM inserted into the CD-ROM drive available
on FOH Rack or Mix Rack.
For complete instructions on installing plugins for VENUE systems, see the documentation that came with your VENUE system.
10
Not all Pro Tools plug-ins require authorization. For example, no authorization is
required for the free plug-ins included with
Pro Tools.
Audio Plug-Ins Guide
An iLok can hold hundreds of licenses for all of
your iLok-enabled software. Once an iLok is authorized for a given piece of software, you can
use the iLok to authorize that software on any
computer.
The iLok USB Smart Key is not supplied
with plug-ins or software options. You can
use the one included with certain Pro Tools
systems (such as Pro Tools|HD-series systems), or purchase one separately.
For more information, visit the iLok website
(www.iLok.com).
Authorizing Downloaded
Plug-Ins for Pro Tools
If you downloaded a plug-in from the Avid Store
(store.avid.com), you authorize it by downloading a license from iLok.com to an iLok.
For more information, visit the iLok website
(www.iLok.com).
Authorizing Boxed Versions of
Plug-Ins for Pro Tools
Authorizing Plug-Ins on VENUE
Systems
If you purchased a boxed version of software, it
comes with an Activation Code (on the included
Activation Card).
After installing a plug-in on a VENUE system,
the system re-creates the list of available plugins. Whenever the racks initialize, the system
checks authorizations for all installed plug-ins.
If no previous authorization for a plug-in is recognized, you will be prompted to authorize the
the plug-in.
To authorize a plug-in using an Activation Code:
If you do not have an iLok.com account, visit
www.iLok.com and sign up for an account.
1
Transfer the license for your plug-in to your
iLok.com account by doing the following:
2
• Visit www.avid.com/activation.
– and –
• Input your Activation Code (listed on your
Activation Card) and your iLok.com User
ID. Your iLok.com User ID is the name you
create for your iLok.com account.
Transfer the licenses from your iLok.com account to your iLok USB Smart Key by doing the
following:
3
• Insert the iLok into an available USB port
on your computer.
• Go to www.iLok.com and log in.
• Follow the on-screen instructions for transferring your licences to your iLok.
4
Launch Pro Tools.
If you have any installed unauthorized plugins or software options, you are prompted to authorize them. Follow the on-screen instructions
to complete the authorization process.
5
For complete instructions on authorizing
plug-ins for VENUE systems, see the
documentation that came with your VENUE
system.
VENUE supports challenge/response and iLok
USB Smart Key authorization, including pre-authorized iLoks and Activation Cards.
Challenge/Response Challenge/response authorization is only valid for the VENUE system the
plug-in is currently installed on. Challenge/response codes can be communicated using any
computer with Internet access.
iLok USB Smart Key Plug-Ins supporting web
authorizations through iLok.com can be authorized for your iLok USB Smart Key from any
computer with Internet access. This lets you
take your iLok and your plug-in authorizations
anywhere, to use plug-ins installed on any
system.
For more information, visit the iLok website
(www.iLok.com).
Chapter 2: Installing Plug-Ins
11
Removing Plug-Ins for
Pro Tools
Removing Plug-Ins for VENUE
Systems
If you need to remove a plug-in from your
Pro Tools system, follow the instructions for
your computer platform.
Plug-Ins installed on VENUE systems can be
disabled, uninstalled, or deleted. A plug-in that
has been disabled or uninstalled (but not deleted) can be reinstalled without the CD-ROM or
USB drive containing the plug-in installers. Deleted plug-ins, however, must be reinstalled
from installers located on either a USB drive or a
CD-ROM.
To remove a plug-in on Mac:
Locate and open the Plug-Ins folder on your
Startup drive, which will be in one of the following locations:
1
• Library/Application Support/
Digidesign/Plug-Ins
– or –
• Library/Application Support/Avid/Audio/
Plug-Ins
2
Do one of the following:
• Drag the plug-in to the Trash and empty the
Trash.
– or –
• Drag the plug-in to the Plug-Ins (Unused)
folder.
To remove a plug-in in Windows:
1
Choose Start > Control Panel.
2
Click Programs and Features.
3 Select the plug-in from the list of installed applications.
4
Click Uninstall.
Follow the on-screen instructions to remove
the plug-in.
5
12
Audio Plug-Ins Guide
For complete instructions on uninstalling
plug-ins for VENUE systems, see the
documentation that came with your VENUE
system.
Part II: EQ Plug-Ins
Chapter 3: AIR Kill EQ
AIR Kill EQ is an RTAS EQ plug-in.
Use the Kill EQ plug-in to zap out the Low, Mid,
or High broadband frequency range from an audio signal. This is a popular effect with DJs and
is commonly used in electronic music production (especially in dance music).
Kill EQ Controls
The Kill EQ plug-in provides a variety of controls for adjusting plug-in parameters.
Kill Switches
The High, Mid, and Low switches toggle their respective frequency bands on and off.
Gain
The Low, Mid, and High gain knobs control the
relative volume of the three frequency bands.
Freq
Kill EQ plug-in window
The Low and High freq controls set the crossover frequencies of the low and high pass filters.
The Sweep control changes both the low and
high-band cutoff frequencies simultaneously.
When the high and low bands are killed, manipulating this control creates a swept bandpass filter effect.
Output
The Output control sets the final output volume.
Chapter 3: AIR Kill EQ
15
16
Audio Plug-Ins Guide
Chapter 4: EQ III
The EQ III plug-in provides a high-quality
7 Band, 2–4 Band, or 1 Band EQ for adjusting
the frequency spectrum of audio material.
EQ III is available in the following formats:
• 7 Band: TDM, RTAS, and AudioSuite
• 2–4 Band: TDM and RTAS only
• 1 Band: TDM, RTAS, and AudioSuite
EQ III supports all Pro Tools session sample
rates: 192 kHz, 176.4 kHz, 96 kHz, 88.2 kHz,
48 kHz, and 44.1 kHz. EQ III operates as a
mono, multi-mono, or stereo plug-in.
EQ III can be operated from the following control surfaces:
• D-Command
• D-Control
• ProControl
EQ III has a Frequency Graph display that shows
the response curve for the current EQ settings
on a two-dimensional graph of frequency and
gain. The frequency graph display also lets you
modify frequency, gain and Q settings for individual EQ bands by dragging their corresponding points in the graph.
By choosing from the 7 Band, 2–4 Band, or
1 Band versions of the EQ III plug-in, you can
use only the number of EQ bands you need for
each track, conserving DSP capacity on
Pro Tools|HD systems.
EQ III Configurations
The EQ III plug-in appears as three separate
choices in the plug-in insert pop-up menu and
in the AudioSuite menu:
• C|24
• 1 Band (“1-Band EQ 3”)
• Control|24
• 2–4 Band (“4-Band EQ 3”)
• Digi 003
• 7 Band (“7-Band EQ 3”)
• Digi 002
• Command|8
• EUCON ™
• Mackie HUI-compatible controllers
Chapter 4: EQ III
17
1 Band EQ
The 1 Band EQ is available in TDM, RTAS, and
AudioSuite formats.
The 1 Band EQ has its own window, with six selectable filter types.
Adjusting EQ III Controls
You can adjust the EQ III plug-in controls using
different methods.
Dragging Plug-In Controls
The rotary controls on the EQ III plug-in can be
adjusted by dragging over them horizontally or
vertically. Dragging up or to the right increments the control. Dragging down or to the left
decrements the control.
1 Band EQ window
7 Band EQ and 2–4 Band EQ
The 7 Band EQ is available in TDM, RTAS, and
AudioSuite formats. The 2–4 Band EQ is available in TDM and RTAS formats only.
The 7 Band EQ and the 2–4 Band EQ share the
same window and identical controls, but with
the 2–4 Band EQ, a limited number of the seven
available bands can be active at the same time.
Dragging a plug-in control in EQ III
Typing Control Values
You can enter control values directly by clicking
in the corresponding text box, typing a value,
and pressing Enter (Windows) or Return (Mac).
Typing a control value
Inverting Filter Gain
(Peak EQ Bands Only)
7 Band EQ and 2–4 Band EQ window
18
Audio Plug-Ins Guide
Gain values can be inverted on any Peak EQ
band by Shift-clicking its control dot in the Frequency Graph display, or its Gain knob in the
plug-in window. This changes a gain boost to a
cut (+9 to –9) or a gain cut to a boost (–9 to +9).
Gain values cannot be inverted on Notch,
High Pass, Low Pass, or shelving bands.
Dragging in the Frequency
Graph Display
Resetting EQ III Controls to
Default Values
You can adjust the following by dragging the
control points directly in the Frequency Graph
display:
You can reset any on-screen control to its default value by Alt-clicking (Windows) or Option-clicking (Mac OS) directly on the control or
on its corresponding text box.
Frequency Dragging a control point to the right
increases the Frequency setting. Dragging a control point to the left decreases the Frequency
setting.
Gain Dragging a control point up increases the
Gain setting. Dragging a control point down decreases the Gain setting.
Q Start-dragging (Windows) or Control-dragging (Mac) a control point up increases the Q
setting. Start-dragging (Windows) or Controldragging (Mac) a control point down decreases
the Q setting.
Using EQ III in Band-Pass Mode
You can temporarily set any EQ III control to
Band-Pass monitoring mode. Band-Pass mode
cuts monitoring frequencies above and below
the Frequency setting, leaving a narrow band of
mid-range frequencies. It is especially useful for
adjusting limited bandwidth in order to solo and
fine-tune each individual filter before reverting
the control to notch filter or peaking filter type
operations.
Band-Pass mode does not affect EQ III Gain
controls.
To switch an EQ III control to Band-Pass mode:
 Hold Start+Shift (Windows) or Control+Shift
(Mac), and drag any rotary control or control
point horizontally or vertically.
Dragging a control point in the Frequency Graph
display
Adjusting Controls with Fine
Resolution
Controls and control points can be adjusted with
fine resolution by holding the Command key
(Mac) or the Control key (Windows) while adjusting the control.
EQ III interactive graph displaying Band-Pass mode
Chapter 4: EQ III
19
When monitoring in Band-Pass mode, the Frequency and Q controls function differently.
Frequency Sets the frequency above and below
which other frequencies are cut off, leaving a
narrow band of mid-range frequencies.
Q Sets the width of the narrow band of midrange frequencies centered around the Frequency setting.
EQ III I/O Controls
Certain Input and Output controls are found on
all EQ III configurations, except where noted
otherwise.
Input and Output Meters
Clip
Indicators
To switch an EQ III control out of Band-Pass mode:
 Release Start+Shift (Windows) or Control+Shift (Mac).
Controlling EQ III from a Control
Surface
EQ III can be controlled from any supported
control surface, including our D-Control, DCommand, ProControl, C|24, 003, Digi 002, or
Command|8. Refer to the guide that came with
the control surface for details.
Input
Polarity
Control Input
Gain
Control
Output Gain
Control
I/O controls and meters for 7 Band EQ and
2–4 Band EQ (top) and 1 Band EQ (bottom)
Input Gain Control
The Input Gain control sets the input gain of the
plug-in before EQ processing, letting you make
up gain or prevent clipping at the plug-in input
stage.
Output Gain Control
(7 Band EQ and 2–4 Band EQ Only)
The Output Gain control sets the output gain after EQ processing, letting you make up gain or
prevent clipping on the channel where the plugin is being used.
20
Audio Plug-Ins Guide
Input Polarity Control
The Input Polarity button inverts the polarity of
the input signal, to help compensate for phase
anomalies occurring in multi-microphone environments, or because of mis-wired balanced
connections.
Input and Output Meters
(7 Band EQ and 2–4 Band EQ Only)
The plasma-style Input and Output meters show
peak signal levels before and after EQ processing, and indicate them as follows:
Green Indicates nominal levels
EQ III EQ Band Controls
Individual EQ bands on each EQ III configuration have a combination of controls.
EQ Type Selector
On the 1 Band EQ, the EQ Type selector lets you
choose any one of six available filter types:
High Pass, Notch, High Shelf, Low Shelf, Peak,
and Low Pass.
On the 7 Band EQ and the 2–4 Band EQ, the
HPF, LPF, LF, and HF sections have EQ Type selectors to toggle between the two available filter
types in each section.
Yellow Indicates pre-clipping levels, starting at
–6 dB below full scale
Red Indicates full scale levels (clipping)
When using the stereo version of EQ III, the Input and Output meters display the sum of the
left and right channels.
The Clip indicators at the far right of each meter
indicate clipping at the input or output stage of
the plug-in. Clip indicators can be cleared by
clicking the indicator.
EQ Type Selectors
Chapter 4: EQ III
21
Band Enable Button
(7 Band EQ and 2–4 Band EQ Only)
The Band Enable button on each EQ band toggles the corresponding band in and out of circuit. When a Band Enable button is highlighted,
the band is in circuit. When a Band Enable button is dark gray, the band is bypassed and available for activation. On the 2–4 Band EQ, when a
Band Enable button is light gray, the band is bypassed and unavailable.
Frequency Control
Each EQ band has a Frequency control that sets
the center frequency (Peak, Shelf and Notch
EQs) or the cutoff frequency (High Pass and
Low Pass filters) for that band.
Frequency control
Q Control
Peak and Notch On Peak and Notch bands, the Q
Band Enable button
Band Gain Control
Each Peak and Shelf EQ band has a Gain control
for boosting or cutting the corresponding frequencies. Gain controls are not used on
High Pass, Low Pass, or Notch filters.
control changes the width of the EQ band.
Higher Q values represent narrower bandwidths.
Lower Q values represent wider bandwidths.
Shelf On Shelf bands, the Q control changes the
Q of the shelving filter. Higher Q values represent steeper shelving curves. Lower Q values
represent broader shelving curves.
Band Pass On High Pass and Low Pass bands,
the Q control lets you select from any of the following Slope values: 6 dB, 12 dB, 18 dB, or 24 dB
per octave.
Band Gain control
Q control
22
Audio Plug-Ins Guide
EQ III Frequency Graph Display
(7 Band EQ and 2–4 Band EQ Only)
The Frequency Graph display in the 7 Band EQ and the 2–4 Band EQ shows a color-coded control dot
that corresponds to the color of the Gain control for each band. The filter shape of each band is similarly color-coded. The white frequency response curve shows the contribution of each of the enabled
filters to the overall EQ curve.
Low
control dot
(red)
Mid
High
control dot control dot
(yellow)
(blue)
Frequency
response
curve
High Pass
control dot
(gray)
Low Mid
control dot
(brown)
High Mid
control dot
(green)
Low Pass
control dot
(gray)
Frequency Graph display for the 7 Band EQ
Chapter 4: EQ III
23
7 Band EQ III
The 7 Band EQ has the following available bands: High Pass/Low Notch, Low Pass/High Notch,
Low Shelf/Low Peak, Low Mid Peak, Mid Peak, High Mid Peak, and High Shelf/High Peak.
All seven bands are available for simultaneous use. In the factory default setting, the High Pass/Low
Notch and Low Pass/High Notch bands are out of circuit, the Low Shelf and High Shelf bands are selected and in circuit, and the Low Mid Peak, Mid Peak, High Mid Peak bands are in circuit.
Input/Output Level meters
Input/Output Level
and
Polarity controls
Frequency Graph
Display
High Pass/
Low Notch
Low Pass/
High Notch
Low
Shelf/Peak
7 Band EQ and 2–4 Band EQ window
24
Audio Plug-Ins Guide
Low Mid
Peak
Mid
Peak
High Mid
Peak
High
Shelf/Peak
7 Band EQ III High Pass/Low
Notch
7 Band EQ III Low Pass/High
Notch
The High Pass/Notch band is switchable between high pass filter and notch EQ functions.
By default, this band is set to High Pass Filter.
The Low Pass/Notch band is switchable between
low pass filter and notch EQ functions. By default, this band is set to Low Pass Filter.
High Pass Filter Attenuates all frequencies be-
Low Pass Filter Attenuates all frequencies above
low the Frequency setting at the selected slope
while letting all frequencies above pass through.
the Frequency setting at the selected slope while
letting all frequencies below pass through.
Low Notch EQ Attenuates a narrow band of fre-
High Notch EQ Attenuates a narrow band of fre-
quencies centered around the Frequency setting. The width of the attenuated band is determined by the Q setting.
quencies centered around the Frequency setting. The width of the attenuated band is determined by the Q setting.
High Pass Filter
button
Band
Enable
button
Low Notch EQ
button
Band
Enable
button
Frequency Slope
control control
Frequency Q
control control
Low Pass Filter
button
Band
Enable
button
High Notch EQ
button
Band
Enable
button
Frequency Slope
control control
Frequency Q
control control
High Pass filter (left) and Low Notch EQ (right)
Low Pass filter (left) and High Notch EQ (right)
The High Pass and Low Notch EQ controls and
their corresponding graph elements are displayed on-screen in gray. The following control
values are available:
The Low Pass and High Notch EQ controls and
their corresponding graph elements are displayed on-screen in gray. The following control
values are available:
Control
Value
Control
Value
Frequency Range
20 Hz to 8 kHz
Frequency Range
20 Hz to 8 kHz
Frequency Default
20 Hz
Frequency Default
20 Hz
HPF Slope Values
6, 12, 18, or 24 dB/oct
HPF Slope Values
6, 12, 18, or 24 dB/oct
Low Notch Q Range
0.1 to 10.0
Low Notch Q Range
0.1 to 10.0
Low Notch Q Default
1.0
Low Notch Q Default
1.0
Chapter 4: EQ III
25
7 Band EQ III Low Shelf/Low
Peak
The Low Shelf/Peak band is switchable between
low shelf EQ and low peak EQ functions. By default, this band is set to Low Shelf.
Low Shelf EQ Boosts or cuts frequencies at and
below the Frequency setting. The amount of
boost or cut is determined by the Gain setting.
The Q setting determines the shape of the shelving curve.
Low Peak EQ Boosts or cuts a band of frequen-
cies centered around the Frequency setting. The
width of the affected band is determined by the
Q setting.
Low Shelf EQ
button
Q
control
Band
Enable
button
Gain
control
Low Peak EQ
button
Q
control
Band
Enable
button
Gain
control
Frequency
Frequency
control
control
Low Shelf EQ (left) and Low Peak EQ (right)
26
Audio Plug-Ins Guide
The Low Shelf and Low Peak Gain controls and
their corresponding graph elements are displayed on-screen in red. The following control
values are available:
Control
Value
Frequency Range
20 Hz to 500 Hz
Frequency Default
100 Hz
Low Shelf Q Range
0.1 to 2.0
Low Peak Q Range
0.1 to 10.0
Q Default
1.0
Low Shelf Gain Range
–12 dB to +12 dB
Low Peak Gain Range
–18 dB to +18 dB
7 Band EQ III Low Mid Peak
7 Band EQ III Mid Peak
The Low Mid Peak band boosts or cuts frequencies centered around the Frequency setting. The
width of the band is determined by the Q setting.
The Mid Peak band boosts or cuts frequencies
centered around the Frequency setting. The
width of the band is determined by the Q setting.
Q
control
Q
control
Band
Enable
button
Frequency
control
Band
Enable
button
Frequency
control
Gain
control
Gain
control
Low Mid Peak EQ
Mid Peak EQ
The Low Mid Gain control and its corresponding
graph elements are displayed on-screen in
brown. The following control values are available:
The Mid Gain control and its corresponding
graph elements are displayed on-screen in yellow.
Control
Value
Frequency Range
40 Hz to 1 kHz
Frequency Default
200 Hz
Low Mid Peak Q
Range
0.1 to 10.0
Low Mid Peak Q
Default
1.0
Low Mid Peak Gain
Range
–18 dB to +18 dB
Control
Value
Frequency Range
125 Hz to 8 kHz
Frequency Default
1 kHz
Mid Peak Q Range
0.1 to 10.0
Mid Peak Q Default
1.0
Mid Peak Gain Range
–18 dB to +18 dB
Chapter 4: EQ III
27
7 Band EQ III High Mid Peak
The High Mid Peak band boosts or cuts frequencies centered around the Frequency setting. The
width of the band is determined by the Q setting.
Q
control
Band
Enable
button
Frequency
control
7 Band EQ III High Shelf/High
Peak
The High Shelf/Peak band is switchable between
high shelf EQ and high peak EQ functions. By
default, this band is set to High Shelf.
High Shelf EQ Boosts or cuts frequencies at and
above the Frequency setting. The amount of
boost or cut is determined by the Gain setting.
The Q setting determines the shape of the shelving curve.
High Peak EQ Boosts or cuts a band of frequen-
Gain
control
cies centered around the Frequency setting. The
width of the affected band is determined by the
Q setting.
High Shelf EQ
button
High Mid Peak EQ
The High Mid Gain control and its corresponding graph elements are displayed on-screen in
green. The following controls are available:
Control
Value
Frequency Range
200 Hz to 18 kHz
Frequency Default
2 kHz
Mid Peak Q Range
0.1 to 10.0
Mid Peak Q Default
1.0
Mid Peak Gain Range
–18 dB to +18 dB
Q
control
Band
Enable
button
Gain
control
High Peak EQ
button
Q
control
Band
Enable
button
Gain
control
Frequency
Frequency
control
control
High Shelf EQ (left) and High Peak EQ (right)
28
Audio Plug-Ins Guide
The High Shelf and High Peak Gain controls and
their corresponding graph elements are displayed on-screen in blue. The following control
values are available:
Control
Value
Frequency Range
1.8 kHz to 20 kHz
Frequency Default
6 kHz
High Shelf Q Range
0.1 to 2.0
High Peak Q Range
0.1 to 10.0
Q Default
1.0
High Shelf Gain
Range
–12 dB to +12 dB
High Peak Gain
Range
–18 dB to +18 dB
Filter Usage with 2–4 Band
EQ III
With a 2–4 Band EQ, a maximum of four filters
may be active simultaneously, with each of the
five Peak bands (Low Shelf/Peak, Low Mid Peak,
Mid Peak, High Mid Peak and High Shelf/Peak)
counting as one filter. Each of the Band-pass and
Notch filters (High Pass, Low Notch, Low Pass
and High Notch) counts as two filters.
When any combination of these filter types uses
the four-filter maximum on the 2–4 Band EQ,
the remaining bands become unavailable. This
is indicated by the Band Enable buttons turning
light gray. When filters become available again,
the Band Enable button on inactive bands turns
dark gray.
2–4 Band EQ III
The 2–4 Band EQ uses the same plug-in window
as the 7 Band EQ, but on the 2–4 Band EQ, but a
limited number of the seven available bands can
be active at the same time.
In the factory default setting, the High Pass/Low
Notch, Low Pass/High Notch and Mid Peak
bands are out of circuit, the Low Shelf and High
Shelf bands are selected and in circuit, and the
Low Mid Peak and High Mid Peak bands are in
circuit.
For Pro Tools HD, using a 2–4 Band EQ instead of a 7 Band EQ saves DSP resources.
Chapter 4: EQ III
29
Switching Between the
2–4 Band EQ and 7 Band EQ III
1 Band EQ III
When you switch an existing EQ III plug-in between the 2–4 Band and 7 Band versions, or
when you import settings between versions, the
change is subject to the following conditions:
The Frequency Graph display in the 1 Band EQ
shows a control dot that indicates the center frequency (Peak, Shelf and Notch Filters) or the
cutoff frequency (High Pass and Low Pass filters) for the currently selected filter type.
Changing from 2–4 Band to 7 Band
After switching from a 2–4 band EQ to a 7 Band
EQ, or importing settings from a 2–4 Band EQ,
all control settings from the 2–4 Band EQ are
preserved, and the bands in the 7 Band EQ inherit their enabled or bypassed state from the
2–4 Band plug-in.
Additional EQ bands can then be enabled to add
them to the settings inherited from the
2–4 Band plug-in.
Changing from 7 Band to 2–4 Band
Control dot
Frequency
response
curve
Frequency Graph display
Input Level and
Polarity controls
After switching from a 7 band EQ to a 2–4 Band
EQ, or importing settings from a 7 Band EQ, all
control settings from the 7 Band EQ are preserved in the 2–4 Band EQ, but all bands are
placed in a bypassed state.
Frequency Graph
display
Bands can then be enabled manually, up to the
2–4 Band EQ four-filter limit.
EQ Type Gain, Freq and
Q controls
selector
1 Band EQ window
The 1 Band EQ may be set to any one of six EQ
types: High Pass, Notch, High Shelf, Low Shelf,
Peak, and Low Pass, by clicking the corresponding icon in the EQ Type selector.
30
Audio Plug-Ins Guide
Band Controls
Notch Filter
The individual EQ types have some combination
of the following controls, as noted below.
The Notch Filter attenuates a narrow band of
frequencies centered around the Frequency setting. No gain control is available for this EQ
type. The width of the attenuated band is determined by the Q setting.
Control
Value
Frequency Range (All)
20 Hz to 20 kHz
Frequency Default (All)
1 kHz
Q Range (Low/High Shelf)
0.1 to 2.0
Q Range (Peak/Notch)
0.1 to 10.0
Q Default (All)
1.0
Gain Range (Low/High Shelf)
–12 dB to +12 dB
High Peak Gain Range
–18 dB to +18 dB
1 Band EQ set to Notch Filter
1 Band EQ III Types
High Shelf EQ
High Pass Filter
The High Shelf EQ boosts or cuts frequencies at
and above the Frequency setting. The amount of
boost or cut is determined by the Gain setting.
The Q setting determines the shape of the shelving curve.
The High Pass filter attenuates all frequencies
below the Frequency setting at the selected rate
(6 dB, 12 dB, 18 dB, or 24 dB per octave) while
letting all frequencies above pass through. No
gain control is available for this filter type.
1 Band EQ set to High Shelf EQ
1 Band EQ set to High Pass Filter
Chapter 4: EQ III
31
Low Shelf EQ
Low Pass Filter
The Low Shelf EQ boosts or cuts frequencies at
and below the Frequency setting. The amount of
boost or cut is determined by the Gain setting.
The Q setting determines the shape of the shelving curve.
The Low Pass filter attenuates all frequencies
above the cutoff frequency setting at the selected rate (6 dB, 12 dB, 18 dB, or 24 dB per octave) while letting all frequencies below pass
through. No gain control is available for this filter type.
1 Band EQ set to Low Shelf EQ
Peak EQ
The Peak EQ boosts or cuts a band of frequencies centered around the Frequency setting. The
width of the affected band is determined by the
Q setting.
1 Band EQ set to Peak EQ
32
Audio Plug-Ins Guide
1 Band EQ set to Low Pass Filter
Chapter 5: JOEMEEK VC5 Meequalizer
The JOEMEEK VC5 Meequalizer is an EQ
plug-in that is available in TDM, RTAS, and AudioSuite formats and offers simple controls with
incredibly warm, musical results.
JOEMEEK Meequalizer
Controls
Operation of the Meequalizer is dead simple,
and that’s the whole point.
Bass The Bass control adjusts low frequencies
±11.
Mid and Mid Freq The Mid and Mid Freq conJOEMEEK Meequalizer VC5 EQ
How the Meequalizer Works
Picture this: drums in the spare bedroom. Microphones, cables, and recording gear strewn
about the living room. A familiar scene—especially to legendary producer Joe Meek in 1962 as
he prepared to record yet another chart topping
hit.
Among countless other achievements, Joe Meek
built custom gear to get the sounds in his head
onto tape. One device was a treble and bass circuit with a sweepable mid control, built into a
tiny tobacco tin. The Meequalizer VC5 virtually
recreates the exact circuitry used by Joe Meek.
trols allow you to adjust mid frequencies, from
500Hz to 3.5KHz, ±11.
Treble The Treble control adjusts high frequencies ±11.
Gain The Gain control allows you to adjust the
output level ±11.
JOEMEEK Meezqualizer Tips and Tricks
Twelve O’Click
Alt-click (Windows) or Option-click (Mac) any
knob to reset any knob to its unity position
quickly.
Chapter 5: JOEMEEK VC5 Meequalizer
33
34
Audio Plug-Ins Guide
Chapter 6: Pultec Plug-Ins
Pultec plug-ins are a set of EQ plug-ins that are
available TDM, RTAS, and AudioSuite formats.
The following plug-ins are included:
• Pultec EQP-1A (see “Pultec EQP-1A” on
page 35)
• Pultec EQH-2 (see “Pultec EQH-2 ” on
page 36)
• Pultec MEQ-5 (see “Pultec MEQ-5 ” on
page 37)
How Pultec EQP-1A Works
Built in the early 1960s, The Pultec EQP-1A offers gentle shelving program equalization on
bass and highs, and offers a variable bandwidth
peak boost control. A custom (and secret) filter
network provides all its equalization functionality. Quality transformers interface it to realworld studio equipment. And a clean and welldesigned tube amplifier provides a fixed amount
of make-up gain.
Pultec EQP-1A
Pultec EQP-1A Controls
(TDM, RTAS, and AudioSuite)
Low Frequency Section Adjust low frequencies
The Pultec EQP-1A provides smooth, sweet EQ
and an extremely high quality tube audio signal
path. Use it on individual tracks, critical vocals,
or even across a stereo mix for mastering applications.
using the Boost and Atten knobs and the Low
Frequency switch, located at the left side of the
unit. All low-frequency equalization is a gentle
shelving type, 6 dB per octave.
High Frequency Boost Section Boost mid and
high frequencies using the Bandwidth and Boost
knobs and the High Frequency switch.
High Frequency Attenuate Section Cut high fre-
quencies using the Atten knob and the Atten Sel
switch located at the right side of the plug-in.
Pultec EQP-1A
Chapter 6: Pultec Plug-Ins
35
Pultec EQP-1A Tips and Tricks
Twelve O’Click
Alt-click (Windows) or Option-click (Mac) any
knob to reset it to its unity position.
“Q” and A
You may wonder why the Pultec EQP-1A has
separate knobs for boost and cut. The short answer is that they connect to different circuitry in
the unit.
Use caution, because the Sharp bandwidth setting results in up to 10 dB higher output than
Broad bandwidth at maximum Boost, just like
on the original. But don’t feel like you’re getting
cheated. Here at Bomb Factory, we consider
anything that encourages very careful and infrequent use of peaky boosts to be a Very Good
Thing.
Pultec EQH-2
(TDM, RTAS, and AudioSuite)
You can use the “extra” knob to your advantage.
Because the filters are not phase perfect, a Boost
setting of 3 and an Atten setting of 3 can make a
huge difference, even though a frequency plot
wouldn’t show much difference in tone. You’re
hearing the phase shift, not the tone shift.
The Pultec EQH-2 is a program equalizer similar
to the Pultec EQP-1A. It is designed to provide
smooth equalization across final mixes or individual tracks.
Our ears of very sensitive to phase, and using the
two knobs together, you can adjust phase at the
low end while also making tonal adjustments.
On the high end, you can set Boost to 10k and
Atten to 10k, then adjust Boost and Atten simultaneously. However, because Boost is a peak
equalizer and Atten is a shelving equalizer, the
results are much different, and you don’t get independent control of phase.
36
Pultec EQH-2
“Q” and Boost
How Pultec EQH-2 Works
In the high frequency boost section, the Bandwidth and Boost controls affect one another.
This is different from modern equalizers, where
adjusting Q typically doesn’t affect the amount
of equalization applied.
The Pultec EQH-2 offers three equalization sections: low frequency boost and attenuation,
midrange boost only, and 10k attenuation. Like
its EQP-1A sibling, it features high-quality
transformers and a tube gain stage. But unlike
the EQP-1A, the tube stage in the EQH-2 is a
push-pull design. As a result, the EQH-2 offers a
beefier tone.
Audio Plug-Ins Guide
Pultec EQH-2 Controls
Low Frequency Section Adjust low frequencies
using the top row of Boost and Atten knobs and
the CPS (cycles per second) switch. All low-frequency equalization is a gentle shelving type,
6 dB per octave.
High Frequency Boost Section Boost mid and
high frequencies using the KCS (kilocycles per
second) and Boost knobs on the second row.
High Frequency Attenuate Section Cut high fre-
quencies using the 10k Atten knob located at the
right side of the plug-in.
Pultec EQH-2 Tips and Tricks
Alt-click (Windows) or Option-click (Mac) any
knob to reset it to its unity position.
Pultec MEQ-5
(TDM, RTAS, and AudioSuite)
The Pultec MEQ-5 is the most unique equalizer
in the Pultec family. It is particularly useful on
individual tracks during mixdown.
Pultec MEQ-5 Controls
The Pultec MEQ-5 offers three equalization sections: low frequency boost, mid frequency
boost, and wide-range attenuation. Like all
Pultecs, it features quality transformers and a
tube gain stage.
How Pultec MEQ-5 is Used
Low Frequency Peak Boost low frequencies
(200, 300, 500, 700, 1000 Hz) using the upper
left controls.
Mid Frequency Peak Boost mid-frequencies
(1.5k, 2k, 3k, 4k, 5k) using the controls at the
upper right.
Wide-Range Dip Cut frequencies using Dip controls on the bottom row.
Pultec MEQ-5 Tips and Tricks
Guitars
Have multiple guitars that sound like mush in
the mix? The Pultec MEQ-5 is a classic tool for
achieving amazing guitar blends. Try boosting
one guitar and cutting another to achieve an octave of separation. For example, cut one guitar
using 1.5 (1500 Hz) Dip, then boost the other using 3 (3000 Hz) Peak. View the matched pairs of
presets (Guitar 1A and 1B, 2A and 2B, etc.) for
further examples of this technique.
Twelve O’Click
Alt-click (Windows) or Option-click (Mac) any
knob to reset it to its unity position.
Pultec MEQ-5
Chapter 6: Pultec Plug-Ins
37
38
Audio Plug-Ins Guide
Part III: Dynamics Plug-Ins
Chapter 7: BF-2A
The BF-2A is a vintage-style compressor plug-in
that is available in TDM, RTAS, and AudioSuite
formats.
Meticulously crafted to capture every nuance of
the legendary LA-2A tube studio compressor,
the Bomb Factory BF-2A provides the most authentic vintage compression sound available.
BF-2A
How BF-2A Works
Designed and manufactured in the early 1960s,
the LA-2A achieved wide acclaim for its smooth
compression action and extremely high quality
audio signal path.
Originally designed as a limiter for broadcast
audio, a Comp/Limit switch was added to LA-2A
compressors after serial number 572. The subsequent addition of a Comp (Compress) setting
made the LA-2A even more popular for use in
audio production. However, the switch was inconveniently located on the back of the unit next
to the terminal strips and tube sockets in the
original version. In the BF-2A plug-in, the
switch has been placed on the front panel, where
you can make better use of it.
The heart of the LA-2A is its patented T4B ElectroOptical Attenuator, which provides the compression action. The T4B consists of a photoconductive cell, which changes resistance when
light strikes it. It is attached to an electro-luminescent panel, which produces light in response
to voltage. Audio (voltage) is applied to the light
source, and what happens as the audio converts
to light and back to voltage gives the LA-2A its
unique compression action. (Yes, the Bomb Factory BF-2A preserves all the subtle characteristics of this unique electronic circuit.) After compression, gain brings the signal back to its
original level. The LA-2A’s gain comes from a
tube amplifier, which imparts further character
to the tone. In fact, it’s common to see engineers
using the LA-2A simply as a line amp, without
any compression applied to the signal.
One beautiful side effect of the LA-2A’s elegant
design is that it’s easy to hear the compression
action. When the BF-2A’s two knobs are set
properly, you know you got it right. It’s a great
unit for learning the art of compression!
Chapter 7: BF-2A
41
BF-2A Controls
The Peak Reduction and Gain controls combine
with the Comp/Limit switch to determine the
amount and sound of the compression. The following controls and meters are provided:
Gain Gain provides makeup gain to bring the
signal back after passing through peak reduction.
Peak Reduction Peak Reduction controls the
amount of signal entering the side-chain, which
in turn affects the amount of compression and
the threshold. The more Peak Reduction you
dial in, the more “squashed” the sound. Too little peak reduction and you will not hear any
compression action; too much and the sound becomes muffled and dead sounding.
Comp/Lim The Comp/Limit switch affects the
compression ratio. The common setting for audio production is Comp, which provides a maximum compression ratio of approximately 3:1.
In Limit mode, the unit behaves more like a
broadcast limiter, with a higher threshold and
compression ratio of approximately 12:1.
The BF-2A provides an extra parameter, a sidechain filter, that does not have a control on the
plug-in interface, but that can be accessed onscreen through Pro Tools automation controls.
In addition, the side-chain filter can be adjusted
directly from any supported control surface.
This side-chain filter reproduces the effect of an
adjustable resistor on the back panel of the
LA-2A. This control cuts the low frequencies
from the side-chain, or control signal, that determines the amount of gain reduction applied
by the compressor.
By increasing the value of the side-chain filter,
you filter out frequencies below 250 Hz from the
control signal, and decrease their effect on gain
reduction.
 A setting of zero means that the filter is not
applied to the side chain signal.
 A setting of 100 means that all frequencies below 250 Hz are filtered out of the side chain signal.
Meter Both Gain Reduction and Output meter-
To access the side-chain filter on-screen:
ing are provided. The Meter knob operates as
follows:
1
• When set to Gain Reduction, the meter needle moves backward from 0 to show the
amount of compression being applied to the
signal in dB.
• Whe set to Output, the needle indicates the
output level of the signal. The meter is calibrated with 0 VU indicating –18 dBFS.
42
Using the BF-2A Side-Chain
Filter
Audio Plug-Ins Guide
Click the Plug-In Automation button in the
Plug-In window to open the Automation Enable
window.
In the list of controls at the left, click to select
Side-Chain Filter and click Add (or, just doubleclick the desired control in the list).
2
3 Click OK to close the plug-in automation window.
4
In the Edit window, do one of the following:
• Click the Track View selector and select
Side-Chain Filter from the BF-2A submenu.
– or –
• Reveal an Automation lane for the track,
click the Automation Type selector and select Side-Chain Filter from the BF-2A submenu.
5 Edit the breakpoint automation for the BF-2A
side-chain filter. Control range is from 0 (the default setting where no filtering is applied to the
side-chain) to 100% (maximum side-chain filtering).
To access the side-chain filter from a control
surface:
Focus the BF-2A plug-in on your control surface.
1
Adjust the encoder or fader current targeting
the Side-Chain Filter parameter.
2
To automate your adjustments, be sure to
enable automation for that parameter as described above. See the Pro Tools Reference
Guide for complete track automation instructions.
BF-2A Tips and Tricks
AudioSuite Processing
When using the AudioSuite version of the
BF-2A, be sure to select an auxillary side-chain
input (normally the track you’re processing).
The default is “None” and if you leave it set like
this, there is nothing feeding the detector and
you will not hear any compression action.
Line Amp
Turn the Peak Reduction knob full counterclockwise (off) and use the Gain control to increase the signal level. Although the BF-2A does
not compress the sound with these settings, it
still adds its unique character to the tone.
Feed the BF-2A into the BF76
Or vice versa. Glynn Johns (who has worked
with the Stones, the Who, and others) popularized the early ‘70s British trick of combining a
slower compressor with a faster one. The effect
can produce very interesting sounds! Try applying Peak Reduction using the BF-2A, then
squash the missed attacks using the faster BF76.
Chapter 7: BF-2A
43
44
Audio Plug-Ins Guide
Chapter 8: BF-3A
The BF-3A is a vintage-style compressor plug-in
that is available in TDM, RTAS, and AudioSuite
formats.
Bomb Factory extends its award-winning Classic Compressors line with the BF-3A, based on
the classic LA-3A. A secret weapon of pros in the
know, the LA-3A adds a smoothness and sonic
texture that makes sounds jump right out of the
mix.
While the LA-2A’s gain comes from a tube amplifier, the LA-3A's gain comes from a solidstate (transistor) amplifier. This gives the LA3A a solid midrange and more aggressive tone.
Other subtle modifications change the behavior
of the T4B, causing it to respond differently—
particularly in response to percussive material.
The LA-3A is famous for its unique sonic imprint on guitar, piano, vocals and drums. Because it's so easy to control, you'll be getting
classic tones in no time with the BF-3A.
BF-3A Controls
The Peak Reduction and Output Gain controls
combine with the Comp/Limit switch to determine the amount and sound of the compression.
The following controls and meters are provided:
BF-3A
Peak Reduction Peak Reduction controls the
How BF-3A Works
amount of signal entering the side-chain. The
more Peak Reduction you dial in, the more
“squashed” and compressed the sound will be.
Too little peak reduction and you won’t hear any
compression action; too much and the sound becomes muffled and dead sounding.
Designed and manufactured in the late 1960s,
the original LA-3A shares many components in
common with the LA-2A compressor. Just like
the LA-2A, the heart of the LA-3A is the T4B
Electro-Optical Attenuator. This is a device that
converts audio to light and back and is largely
responsible for the compression character of the
unit.
Output Gain Output Gain provides makeup gain
to make the signal louder after passing through
the peak reduction.
Chapter 8: BF-3A
45
Comp/Lim The Comp/Limit switch affects the
compression ratio. The common setting for audio production is Comp, which provides a maximum compression ratio of approximately 3:1.
In Limit mode, the unit behaves more like a
broadcast limiter, with a higher threshold and
compression ratio of approximately 15:1.
Meter Both Gain Reduction and Output meter-
ing are provided. The Meter knob operates as
follows:
• When set to Gain Reduction, the meter needle moves backward from 0 to show the
amount of compression being applied to the
signal in dB.
• When set to Output, the needle indicates
the output level of the signal. The meter is
calibrated with 0 VU indicating –18 dBFS.
BF-3A Tips and Tricks
AudioSuite Processing
When using the AudioSuite version of the
BF-3A, be sure to select an auxillary side-chain
input (normally the track you are processing).
The default is “None” and if you leave it set like
this, there is nothing feeding the detector and
you will not hear any compression action.
Line Amp
Turn the Peak Reduction knob full counterclockwise (off) and use the Gain control to increase the signal level. Although the BF-3A does
not compress the sound with these settings, it
still adds its unique character to the tone.
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Audio Plug-Ins Guide
Chapter 9: BF76
The BF76 is a vintage-style compressor plug-in
that is available in TDM, RTAS, and AudioSuite
formats.
Modeled after the solid-state (transistor) 1176
studio compressor, the Bomb Factory BF76 preserves every sonic subtlety of this classic piece
of studio gear.
enjoy this sound—previously only available to
super-serious-pro-engineers working in expensive pro recording studios—in the privacy of
your own cubicle.
Deep inside the 1176
The Bomb Factory BF76
How the Bomb Factory BF76 Works
The 1176 Compressor, originally introduced in
the late 1970s, uses a FET (field-effect transistor). The 1176 also uses solid state amplification. The 1176 still provides an extremely high
quality audio signal path, but because of these
internal differences offers a much different
compression sound than other compressors.
Four selectable compression ratios are provided, along with controls allowing variable attack and release times.
Various explanations overheard in the control
room include “its 100:1 compression ratio!” or
equally adept quantitative analysis like “it
makes it super squishy sounding.” Now you can
BF76 Controls
BF76 provides the following controls:
Input The Input control sets the input signal
level to the compressor, which, in the 1176 design, determines both the threshold and amount
of peak reduction.
Output The Output control sets output level. Use
it to bring the signal back to unity after applying
gain reduction.
Attack and Release The Attack and Release con-
trols set the attack and release times of the compressor. Full counterclockwise is slowest, and
full clockwise is fastest. Attack times vary between 0.4 milliseconds to 5.7 milliseconds. Release times vary between 0.06 and 1.1 seconds.
Chapter 9: BF76
47
Ratio The Ratio Push switches select the compression ratio from 4:1 to 20:1.
Meter The Meter Push switches affect the meter-
ing.
• GR shows the amount of gain reduction.
• –18 and –24 show the output level (calibrated so that 0VU indicates –18dB FS and
–24dB FS respectively).
• The “Off ” switch turns off the meter.
BF76 Tips and Tricks
AudioSuite Processing
When using the AudioSuite version of the Bomb
Factory BF76, be sure to select a side-chain input (normally the track you are processing). The
default is “None” and if you leave it set like this,
there’s nothing feeding the detector and you
won’t hear any compression action.
Unexpected Visit from A&R Weevil Yields
Instant Hit Mix
A favorite feature on one megabuck mixing console is its stereo bus compressor. With the flick
of a switch, a punchy 8:1 compressor grabs the
current mix producing “instant radio hit.” Although Bomb Factory strongly disapproves of
anything which adds further chaos to the already opaque A&R decision-making process,
you may find that the Bomb Factory BF76 set to
8:1 ratio on a stereo mix provides a fast, easy alternate mix that can provide fresh ideas. It’s also
a handy way to make quick headphone submixes
when tracking overdubs.
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Audio Plug-Ins Guide
Give the Kids What They Want
Shift-click one of the Ratio Push switches to enable the “All Buttons In” mode. The compression ratio is still only 20:1, but the knee changes
drastically and the compressor starts (mis)behaving a little bit like an expander—watch the
meter for details. Hey, try it—sometimes it even
sounds good.
Selecting Proper Attack and Release Times
As on the original unit, setting either the attack
or release time too fast generates signal distortion. Again, this may or may not be the desired
effect. A good starting point for attack and release is “6” and “3” (the defaults), and you can
adjust as follows:
When compressing, use the slowest attack you
can that preserves the desired dynamic range.
Faster attacks remove the “punch” from the performance; slower attacks inhibit the compression you need to smooth things out.
When limiting, use the fastest attack time you
can before you start to hear signal distortion in
the low end. But don’t sweat it too much: on the
Bomb Factory BF76, the attack time ranges from
“incredibly fast” to “really damn fast” by modern standards. It can be hard to hear the difference.
Release times are more critical on the Bomb Factory BF76. To set release times, listen for loud
attacks and what happens immediately after the
peaks. Set the release time fast enough that you
don’t hear unnatural dynamic changes, but slow
enough that you don’t hear unnecessary pumping between two loud passages in rapid succession.
Chapter 10: Channel Strip
Avid Channel Strip is an AAX plug-in (Native
and AudioSuite) that provides EQ, Dynamics,
Filter, and Gain effects. The Avid Channel Strip
processing algorithms are based on the award
winning Euphonix System 5 console channel
strip effects.
Channel Strip supports 44.1 kHz, 48 kHz,
88.2 kHz, 96 kHz, 176.4 kHz and 192 kHz sample rates. Channel Strip supports mono, stereo,
and greater-than-stereo multichannel formats
up to 7.1 (Pro Tools HD and Pro Tools with
Complete Production Toolkit).
In addition to standard knob and fader controls,
Channel Strip also provides a graph to track the
gain transfer curve for the Expander/Gate, Compressor/Limiter, and Side Chain effects, and a
Frequency Graph display that shows the response curve for the current EQ settings on a
two-dimensional graph of frequency and gain.
The frequency graph display also lets you modify frequency, gain and Q settings for individual
EQ bands by dragging their corresponding
points in the graph.
Channel Strip provides different sections for
signal metering and gain adjustment, signal
path ordering, dynamics processing, and equalization and filtering.
Sections and Panes
Channel Strip plug-in, Compressor/Limiter tab shown
The Channel Strip plug-in window is organized
in several sections: Input, FX Chain, Output,
Dynamics, and EQ/Filters. The Dynamics and
EQ/Filters sections can be independently shown
or hidden. This lets you access controls or free
up screen space, depending on your needs.
Chapter 10: Channel Strip
49
When showing the Dynamics or EQ/Filters sections, several tabbed panes of controls are available for each section. You can click a tab to show
the controls for that tabbed pane. For Expand/Gate and Compressor/Limiter, and also
for the For the EQ and Filter effects, clicking the
corresponding control point on the graph display automatically shows the tab for Expander/Gate or the Compressor/Limiter, or the
corresponding EQ band or Filter.
Showing or Hiding the Dynamics
and EQ/Filters Sections
To hide (or show) the Dynamics or EQ/Filters
section of the plug-in window:
 Click the Show/Hide triangle to the left of the
section you want to show or hide.
Disabling or Enabling Channel
Strip Effects
You can independently disable effects in the Dynamics and EX/Filters sections of the Channel
Strip plug-in. For example, you may want to apply Comp/Limit processing to the signal, but not
Exp/Gate; or, you may want to only apply only a
high pass filter.
You can independently show or hide the Dynamics and EX/Filters sections of the Channel Strip
plug-in to use less screen space. These sections
are shown by default.
Dynamics section, Exp/Gate disabled
To enable effects in the Dynamics or EQ/Filters
section:
 Click the Enable/Disable button for the effect
you want to enable so that it is highlighted.
To disable effects in the Dynamics or EQ/Filters
section:
 Click the Enable/Disable button for the effect
you want to disable so that it is not highlighted.
Channel Strip plug-in, Dynamics section hidden
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Pro Tools Reference Guide
Listen Mode
The Side Chain tab in the Dynamics section, and
the EQ and Filter tabs in the EQ/Filter section
provide a Listen button.
 When enabled for the Side Chain, Listen mode
lets you hear the input signal that feeds the dynamic section. This can be either the external
key input or the internal side chain (including
the applied filter).
When enabled for any of the EQ bands, Listen
solos the corresponding EQ band and (temporarily) inverts the EQ Type so that you can tune
the Frequency and the Q for that EQ band.

Adjusting Controls with Fine
Resolution
Controls and control points can be adjusted with
fine resolution by holding the Command key
(Mac) or the Control key (Windows) while adjusting the control.
Input
The Input section provides input metering, and
controls for trimming the input signal and inverting its phase. It can also be toggled to show
post-processing gain reduction meters.
 When enabled for either of the Filter effects,
Listen solos the enabled Filter band and inverts
the Filter so that you can hear the audio signal
being fed into the filter.
To enable (or disable) Listen on the Side Chain
effect, EQ band, or a Filter effect:
 Click the Listen button for the Dynamics or
EQ/Filter tab you want so that it is highlighted.
Click it again so that it is not highlighted to disable it.
Input section (5.1 channel format shown)
Input Trim Control
The Input Trim control sets the input gain of the
plug-in before EQ processing, letting you make
up gain or prevent clipping at the plug-in input
stage.
To Trim the input signal, do one of the following:
Channel Strip plug-in, Side Chain Listen mode
enabled
 Click in the Input Trim field to type the desired Trim value (–36.0 dB to +36.0 dB).
– or –
Control-Shift-click (Mac) or Start-Shiftclick (Windows) and hold an EQ or Filter
control point in the Frequency Graph to
temporarily switch to Listen mode for that
EQ band or Filter effect.
Click Trim and drag up or down to adjust the
Input Trim setting.

Chapter 10: Channel Strip
51
Phase Invert
The Phase Invert button at the top of the Input
section inverts the phase (polarity) of the input
signal, to help compensate for phase anomalies
that can occur either in multi-microphone environments or because of mis-wired balanced connections.
To toggle between the Gain Reduction and Input
meters:
 Click the Input/Gain Reduction toggle in the
top right-hand corner of the Input section.
To enable (or disable) phase inversion on input:
 Click the Phase Invert button so that it is highlighted. Click it again so that it is not highlighted to disable it.
Input Meters
The Input meters show peak signal levels before
processing:
Dark Blue Indicates nominal levels from –INF to
–12 dB.
Light Blue Indicates pre-clipping levels, from
–12 dB to 0 dB.
Toggling between Input and Gain Reduction meters
Output
The Output section provides output metering
and controls for adjusting the level of the output
signal.
White Indicates full scale levels from 0 dB to
+6 dB.
Gain Reduction Meters
The Input meter can be switched to show Gain
Reduction metering for the processed signal
from 0 dB to –36 dB.
The Gain Reduction meters are usually displayed in yellow. When the Knee setting for either or both the Expander and the Compressor is
greater than 0 dB, the Gain Reduction meter displays the amount of the Knee level in amber over
the meter’s usual yellow display.
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Pro Tools Reference Guide
Output section (5.1 channel format shown)
Output Volume Control
The Output Volume control sets the output volume after processing, letting you make up gain
or prevent clipping on the channel where the
Channel Strip plug-in is being used. The Output
Volume control can be set to apply at the end of
the FX Chain (Post) or before the FX Chain
(PRE), see “FX Chain” on page 53.
To set the FX Chain:
Click the FX Chain show/hide button to reveal
the Process Order options.
1
To adjust the Output Volume, do one of the
following:
 Click in the Output Volume field to type the
desired value (–INF dB to +12 dB).
– or –
Click Vol and drag up or down to adjust the
Output Volume setting.

Showing the FX Chain Process Order
2 Click the desired effects chain ordering option
to select it. The available options include:
Output Meters
• EQ > FILT > DYN
The Output meters show peak signal levels after
processing:
• DYN > EQ > FILT
• EQ > DYN > FILT
• FILT > DYN > EQ
Dark Blue Indicates nominal levels from –INF to
–12 dB.
Light Blue Indicates pre-clipping levels, from
–12 dB to 0 dB.
White Indicates full scale levels from 0 dB to
+6 dB (which can result in distortion and clipping).
FX Chain
Channel Strip lets you determine the signal path
through the available Equalizer (EQ), Filter
(FILT), Dynamics (DYN), and Volume (VOL)
processing modules. This way you can determine the best signal path for the type of processing you want.
Select PRE or POST to place the Output Volume control at the beginning or at the end of the
effects signal chain.
3
Bypassing or Unbypassing
Individual Effects Modules
In the FX Chain display, you can deselect or select individual effects modules to bypass or unbypass the effect.
FX Chain, FILT bypassed
To bypass an effect module:

Click the module so that it is not highlighted.
To unbypass an effect module:

Click the module so that it is highlighted.
Chapter 10: Channel Strip
53
Dynamics
The Dynamics section of Channel Strip provides
Expander/Gate, Compressor/Limiter, and Side
Chain processing all in one. This section also
provides a dynamics graphic display for the
Compressor/Limiter and Expander/Gate
plug-ins. The display shows a curve that represents the level of the input signal (on the horizontal x–axis) and the amount of gain reduction
applied (on the vertical y–axis). The vertical line
represents the threshold.
The Dynamics Graph display—used with Expander/Gate and Compressor/Limiter processing—shows a curve that represents the level of
the input signal (on the horizontal x–axis) and
the amount of gain reduction applied (on the
vertical y–axis). The display shows two vertical
lines representing the Threshold setting for the
Expander/Gate and Compressor/Limiter, respectively.
The Dynamics Graph display also features an
animated red ball in the gain transfer curve display. This ball shows the amount of input gain
(x-axis) and gain reduction (y-axis) being applied to the incoming signal at any given moment. To indicate overshoots (when an incoming signal peak is too fast for the current
compression setting), the cursor temporarily
leaves the gain transfer curve.
Use this graph as a visual guideline to see how
much dynamics processing you are applying to
the incoming audio signal.
Dynamics Graph Gain Reduction
Resolution
Dynamics section, All tab shown
Channel Strip lets you view the gain reduction
scale on the Dynamics Graph display either in
3 dB increments from 0 dB to 18 dB or in 6 dB
increments from 0 dB to –36 dB.
Dynamics Graph
Input signal
level (x-axis)
Graph
Resolution
toggle
Output signal
level (y-axis)
Compressor/Limiter Threshold
Expander/Gate Threshold
Dynamics graph display
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Pro Tools Reference Guide
To change the Dynamics Graph Gain Reduction
resolution:

Click the Graph Resolution toggle.
Using the Dynamics Graph to Adjust
Controls
You can drag in the Dynamics Graph display to
adjust the corresponding Expander/Gate and
Compressor/Limiter controls. The cursor updates to show which control is being adjusted:
• Expander/Gate Ratio
• Expander/Gate Knee
The Dynamics Graph display shows the threshold as a vertical line.
Attack
The Attack control sets the attack time, or the
rate at which gain is reduced after the input signal crosses the threshold. Use this along with the
Ratio setting to control how soft the Expander’s
gain reduction curve is.
• Expander/Gate Threshold
• Gate Depth
Ratio
• Hysteresis
The Ratio control sets the amount of expansion.
For example, if this is set to 2:1, it will lower signals below the threshold by one half. At higher
ratio levels the Expander/Gate functions like a
gate by cutting off signals that fall below the
threshold. As you adjust the ratio control, refer
to the Dynamics Graph display to see how the
shape of the expansion curve changes.
• Compressor/Limiter Ratio
• Compressor/Limiter Knee
• Compressor/Limiter Threshold
• Limiter Depth
For the Expander/Gate and Compressor
Limiter effects, adjusting a control in the
Dynamics Graph display automatically
shows the pane that includes the adjusted
control if it is not already shown (except
when the All tab is shown).
Expander/Gate Controls
Depth
The Depth control sets the depth of the Expander/Gate when closed. Setting the gate to
higher range levels allows more and more of the
gated audio that falls below the threshold to
peek through the gate at all times.
Hold
Dynamics section, Expander/Gate tab
Threshold
The Threshold ( Thresh) control sets the level below which an input signal must fall to trigger expansion or gating. Signals that fall below the
threshold will be reduced in gain. Signals that
are above it will be unaffected.
The Hold control specifies the duration (in seconds or milliseconds) during which the Expander/Gate will stay in effect after the initial
attack occurs. This can be used as a function to
keep the Expander/Gate in effect for longer periods of time with a single crossing of the threshold. It can also be used to prevent gate chatter
that may occur if varying input levels near the
threshold cause the gate to close and open very
rapidly.
Chapter 10: Channel Strip
55
Release
The Release control sets how long it takes for
the gate to close after the input signal falls below
the threshold level and the hold time has passed.
Knee
The Knee control sets the rate at which the Expander/Gate reaches full effect once the threshold has been exceeded.
Hysteresis
The Hysteresis ( Hyst) control lets you adjust
whether or not the gate rapidly opens and closes
when the input signal is fluctuating near the
Threshold. This can help prevent undesirably
rapid gating of the signal. This control is only
available when Ratio is set to Gate, otherwise it
is greyed out.
Compressor/Limiter Controls
The smaller the value, the faster the attack. The
faster the attack, the more rapidly the Compressor/Limiter applies attenuation to the signal. If
you use fast attack times, you should generally
use a proportionally longer release time, particularly with material that contains many peaks in
close proximity.
Ratio
The Ratio control sets the compression ratio, or
the amount of compression applied as the input
signal exceeds the threshold. For example, a 2:1
compression ratio means that a 2 dB increase of
level above the threshold produces a 1 db increase in output. The compression ratio ranges
from 1:0:1 to 20:0:1.
Once the Ratio control passes 20:0:1 the Compressor/Limiter effect functions as a limiter
rather than a compressor.
At the limiter setting ( LMTR), for every decibel
that the incoming signal goes over the set
Threshold, 1 dB of gain reduction is applied.
Dynamics section, Compressor/Limiter tab
Threshold
The Threshold control sets the level that an input
signal must exceed to trigger compression or
limiting. Signals that exceed this level will be
compressed. Signals that are below it will be unaffected.
Compressor/Limiter Ratio set to LMTR
Once the Ratio control passes the LMTR setting,
it provides negative ratio settings from –20:0:1
to 0:1.
Attack
The Attack control sets the attack time, or the
rate at which gain is reduced after the input signal crosses the threshold.
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Pro Tools Reference Guide
Compressor/Limiter Ratio set to a negative value
With these settings, for every decibel that the incoming signal goes over the set Threshold, more
than 1 dB of gain reduction is applied according
to the negative Ratio setting. For example, at the
setting of –1.0:1, for each decibel over the set
threshold, 2 db of gain reduction is allied. Consequently, the output signal is both compressed
and made softer. You can use this as an creative
effect, or as a kind of ducking effect when used
with an external key input.
Depth
The Depth control sets the amount of gain reduction that is applied regardless of the input
signal. For example, if the Limiter is set at a
Threshold of –20 dB and Depth is set at 0 dB, up
to 20 dB of gain reduction is applied to the incoming signal (at 0 dB). If you set Depth to
–10 dB, no more than 10 dB of gain reduction is
applied to the incoming signal.
As you increase this control, it goes from applying “hard-knee” compression to “soft-knee”
compression:
• With hard-knee compression, compression
begins when the input signal exceeds the
threshold. This can sound abrupt and is
ideal for limiting.
• With soft-knee compression, gentle compression begins and increases gradually as
the input signal approaches the threshold,
and reaches full compression after exceeding the threshold. This creates smoother
compression.
Gain
The Gain control lets you boost overall output
gain to compensate for heavily compressed or
limited signals.
Side Chain Processing Controls
Release
The Release control sets the length of time it
takes for the Compressor/Limiter to be fully deactivated after the input signal drops below the
threshold.
Release times should be set long enough that if
signal levels repeatedly rise above the threshold,
the gain reduction “recovers” smoothly. If the
release time is too short, the gain can rapidly
fluctuate as the compressor repeatedly tries to
recover from the gain reduction. If the release
time is too long, a loud section of the audio material could cause gain reduction that continues
through soft sections of program material without recovering.
Knee
The Knee control sets the rate at which the compressor reaches full compression once the
threshold has been exceeded.
Dynamics section, Side Chain tab
Dynamics processors typically use the detected
amplitude of their input signal to trigger gain
reduction. This split-off signal is known as the
side-chain. Compressor/Limiter and Expander/Gate processing features external key
capabilities and filters for the side-chain.
With external key side-chain processing, you
trigger dynamics processing using an external
signal (such as a separate reference track or audio source) instead of the input signal. This external source is known as the key input.
Chapter 10: Channel Strip
57
With side-chain filters, you can make dynamics
processing more or less sensitive to certain frequencies. For example, you might configure the
side-chain so that certain lower frequencies on a
drum track trigger dynamics processing.
All-Linked If All-Linked is selected, dynamics
processing is applied equally to all channels
when the input signal reaches the threshold on
any input channel, except for the LFE channel (if
present). The LFE channel is processed independently based on its own input signal.
Source
The Source selector lets you set the source for
side chain processing: Internal, Key, or AllLinked.
Internal If Internal is selected, the plug-in uses
the amplitude of the input signal to trigger dynamics processing. With greater-than-stereo
multichannel processing, the input signal for
each stereo pair effects only those same channels, and likewise mono channels are effected
only by their own input signal. For example,
with an LCR multichannel format, the processing for the Center channel is only triggered
when the Center channel input signal reaches
the threshold. However, when the input signal
reaches the threshold on the Left or the Right
channel, processing is triggered for both the Left
and the Right channel.
Key If Key is selected, the plug-in uses the amplitude of a separate reference track or external
audio source to trigger dynamics processing.
The reference track used is selected using the
Plug-In Key Input selector in the Plug-In window header. With greater-than-stereo multichannel processing, the key signal triggers dynamics processing for all processed audio
channels equally.
Selecting the Source setting for Side Chain
processing
Detection
The Detection options include Peak or Avg (Average).
Peak Select the Peak option to apply side-chain
processing according to the detected peak amplitude.
Average Select the Average option to apply sidechain processing according to the detected average amplitude.
Filter Frequency
The Filter Frequency control lets you set the frequency for the selected Filter Type.
Filter Type
Four Filter Type options are available for sidechain processing:
Low Pass Select the Low Pass option to apply a
low pass filter to the side-chain processing at the
selected frequency.
High Pass Select the High Pass option to apply a
high pass filter to the side-chain processing at
the selected frequency.
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Pro Tools Reference Guide
Notch Select the Notch option to apply a notch
filter to the side-chain processing at the selected
frequency.
EQ/Filters
a band pass filter to the side-chain processing at
the selected frequency.
The EQ/Filters section of Channel Strip provides
a high-quality 4-band parametric equalizer for
adjusting the frequency spectrum of audio material.
Side Chain Processing Graph
EQ/Filters Graph
The Side Chain Processing Graph display shows
the frequency curve for the selected Filter Type at
the selected Filter Frequency.
The EQ/Filters section provides an interactive
Frequency Graph display that shows the response curve for the current EQ settings on a
two-dimensional graph of frequency and gain.
The Frequency Graph display also lets you modify frequency, gain, and Q settings for individual
EQ bands by dragging their corresponding
points in the graph. The Frequency Graph display also plots the frequency, Q, and filter shape
of the two filters (when either or both are enabled).
Band Pass Select the Band Pass option to apply
Frequency
(x-axis)
Graph Resolution
toggle
Gain
(y-axis)
Filter control point
EQ control point
EX/Filters section, High Mid Frequency tab shown
Chapter 10: Channel Strip
59
Frequency Graph Gain Resolution
Channel Strip lets you view the gain scale on the
Frequency Graph display either in 3 dB increments from –12 dB to +12 dB or in 6 dB increments from –24 dB to +24 dB.
Q Click within the curve of an EQ control point
and drag up or down to increase or decrease the
Q setting.
You can also Command-Control-click
(Mac) or Control-Alt-click (Windows) and
drag a control point up or down to increase
or decrease the Q setting.
To change the Frequency Graph Gain resolution:

Click the Graph Resolution toggle.
Using the Frequency Graph to Adjust
Controls
Low Frequency EQ Controls
You can adjust the following EQ controls by
dragging the control points directly in the Frequency Graph display:
Frequency Dragging a control point to the right
increases the Frequency setting. Dragging a control point to the left decreases the Frequency
setting.
You can press the Shift key while clicking
and dragging an EQ control point up or
down to adjust the Gain setting without
changing the Frequency. Likewise, press the
Shift key while clicking and dragging an EQ
control point left or right to adjust the Frequency setting without changing the Gain
setting.
Gain Dragging a control point up increases the
Gain setting. Dragging a control point down decreases the Gain setting.
Command-Shift-click (Mac) or ControlShift-click (windows) an EQ control point to
invert its Gain setting.
EQ/Filters section, Low Frequency tab
The LF tab provides controls for the low frequency band of the EQ. The low frequency band
can be set to be a Peak or Low Shelf EQ.
EQ Type
Select either the Peak or Low Shelf button to set
the EQ type for the low frequency band.
Frequency
The Frequency control lets you set the center
frequency for the low frequency band (Peak or
Shelf EQ).
Gain
The Gain control lets you boost or attenuate the
corresponding frequencies for the low frequency band.
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Pro Tools Reference Guide
Q
Q
With the low band EQ set to Peak, the Q control
changes the width of the EQ band. Higher Q values represent narrower bandwidths. Lower Q
values represent wider bandwidths.
The Q control changes the width of the low mid
peak EQ band. Higher Q values represent narrower bandwidths. Lower Q values represent
wider bandwidths.
With the low band EQ set to Shelf, the Q control
changes the Q of the shelving filter. Higher Q
values represent steeper shelving curves. Lower
Q values represent broader shelving curves.
High Mid Frequency EQ Controls
Low Mid Frequency EQ Controls
EQ/Filters section, High Mid Frequency tab
The HMF tab provides controls for the high mid
frequency band of the EQ. This band is a peak
EQ.
EQ/Filters section, Low Mid Frequency tab
Frequency
The LMF tab provides controls for the low mid
frequency band of the EQ. This band is a peak
EQ.
The Frequency control lets you set the center
frequency for the peak high mid frequency band.
Frequency
Gain
The Frequency control lets you set the center
frequency for the peak low mid frequency band.
The Gain control lets you boost or attenuate the
corresponding frequencies for the high mid frequency band.
Gain
Q
The Gain control lets you boost or attenuate the
corresponding frequencies for the low mid frequency band.
The Q control changes the width of the high mid
peak EQ band. Higher Q values represent narrower bandwidths. Lower Q values represent
wider bandwidths.
Chapter 10: Channel Strip
61
High Frequency EQ Controls
Filter 1 and Filter 2 Controls
EQ/Filters section, High Frequency tab
EQ/Filters section, Filter 1 tab shown
The High Frequency EQ tab provides controls
for the high frequency band of the EQ.
The Filter 1 and Filter 2 tabs provide the same
set of controls for each filter.
Filter Type
Filter Type
The High Frequency band can be set to be a Peak
or High Shelf EQ.
Both Filter 1 and Filter 2 can be set independently. Select from the following Filter Type
options:
Frequency
The Frequency control lets you set the center
frequency for the high frequency band (Peak or
Shelf EQ).
Gain
The Gain control lets you boost or attenuate the
corresponding frequencies for the high frequency band.
Q
With the high band EQ set to Peak, the Q control
changes the width of the EQ band. Higher Q values represent narrower bandwidths. Lower Q
values represent wider bandwidths.
With the high band EQ set to Shelf, the Q control
changes the Q of the shelving filter. Higher Q
values represent steeper shelving curves. Lower
Q values represent broader shelving curves.
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Pro Tools Reference Guide
• High Pass
• Low Pass
• Band Pass
• Notch
Frequency
The Frequency control lets you set the center
frequency for the selected Filter Type (from
20 Hz to 21.0 kHz).
Slope
When the Filter Type is set to Low Pass or High
Pass, the Slope control is available. The Slope
control lets you set the slope for the filter from
the selected Frequency to –INF (12 dB/O or
24 dB/O).
Q
When the Filter Type is set to Band Pass or
Notch, the Q control is available. The Q control
changes the width of the filter around the center
frequency band. Higher Q values represent narrower bandwidths. Lower Q values represent
wider bandwidths.
Chapter 10: Channel Strip
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Pro Tools Reference Guide
Chapter 11: Dynamics III
Dynamics III is a suite of three dynamics
plug-ins that are available in TDM, RTAS, and
AudioSuite formats:
• Compressor/Limiter (see “Compressor/Limiter III” on page 68)
• Expander/Gate (see “Expander/Gate III” on
page 71)
• De-Esser (see “De-Esser III” on page 73)
Dynamics III supports 44.1 kHz, 48 kHz,
88.2 kHz, 96 kHz, 176.4 kHz and 192 kHz sample rates. Compressor/Limiter and Expander/Gate modules work with mono, stereo,
and greater-than-stereo multichannel formats
up to 7.1. The De-Esser module works with
mono and stereo formats only.
In addition to standard controls in each module,
Dynamics III also provides a graph to track the
gain transfer curve in the Compressor/Limiter
and Expander/Gate plug-ins, and a frequency
graph to display which frequencies trigger the
De-Esser and which frequencies will be gain reduced.
Dynamics III Shared Features
and Controls
The Levels section, the LFE Enable button, and
the Dynamics Graph display of the user interface are shared between the Compressor/Limiter, Expander/Gate, and De-Esser plug-ins.
Dynamics III Levels Section
The indicators and controls in the Dynamic III
Levels section let you track input, output, and
gain reduction levels, as well as work with phase
invert and the threshold setting.
See “De-Esser III Level Meters” on page 74
for more information on De-Esser III Input/Output Level controls.
Phase
Invert
Input
meter
Peak hold
indicators
Threshold
arrow
Output meter
Gain
Reduction
meter
Peak hold
indicators
I/O Meter display (stereo instance shown)
Chapter 11: Dynamics III
65
Input and Output Meters
Gain Reduction Meter
The Input (In) and Output (Out) meters show
peak signal levels before and after dynamics
processing:
The Gain Reduction (GR) meter indicates the
amount the input signal is attenuated (in dB)
and shows different colors during dynamics
processing:
Green Indicates nominal levels.
Yellow Indicates pre-clipping levels, starting at
–6 dB below full scale.
Red Indicates full scale levels (clipping).
The clip indicators at the top of the Output meters indicate clipping at the input or output
stage of the plug-in. Clip indicators can be
cleared by clicking the indicator.
The Input and Output meters display differently depending on the type of track (mono,
stereo, or multichannel) on which the plugin has been inserted.
When Side-Chain Listen is enabled, the Output meter only displays the levels of the sidechain signal. See “Dynamics III Side-Chain
Listen” on page 76.
Toggling Multichannel Input and Output Meters
With multichannel track types LCRS and higher,
both Input and Output meters cannot be shown
at the same time. Click either the Input or Output button to display the appropriate level meter. The Input/Output meters display is toggled
to Output by default.g
Light Orange Indicates that gain reduction is
within the “knee” and has not reached the full
ratio of compression.
Dark Orange Indicates that gain reduction is being applied at the full ratio (for example, 2:1).
Threshold Arrow
The orange Threshold arrow next to the Input
meter indicates the current threshold, and can
be dragged up or down to adjust the threshold.
When a multichannel instance of the plug-in has
been configured to show only the Output meter,
the Threshold arrow is not displayed.
Phase Invert
The Phase Invert button at the top of the Levels
section inverts the phase (polarity) of the input
signal, to help compensate for phase anomalies
that can occur either in multi-microphone environments or because of mis-wired balanced connections.
Dynamics III LFE Enable
(Pro Tools HD and Pro Tools with Complete
Production Toolkit Only)
The LFE Enable button (located in the Options
section) is on by default, and enables plug-in
processing of the LFE (low frequency effects)
channel on a multichannel track formatted for
5.1, 6.1, or 7.1 surround formats. To disable LFE
processing, deselect this button.
Input (left) and Output (right) meter buttons
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Audio Plug-Ins Guide
The Compressor/Limiter and Expander/Gate
plug-ins also feature an animated, multi-color
cursor in their gain transfer curve displays.
LFE Enable button (Compressor/Limiter III shown)
The gain transfer curve of the Compressor/Limiter and Expander/Gate plug-ins shows a moving
ball cursor that shows the amount of input gain
(x-axis) and gain reduction (y-axis) being applied to the incoming signal.
The LFE Enable button is not available if the
plug-in is not inserted on an applicable
track.
Dynamics III Graph Display
The Dynamics Graph display—used with the
Compressor/Limiter and Expander/Gate plugins—shows a curve that represents the level of
the input signal (on the horizontal x–axis) and
the level of the output signal (on the vertical
y–axis). The orange vertical line represents the
threshold.
Use this graph as a visual guideline to see how
much dynamics processing you are applying.
Threshold
Output signal level (y-
Input signal
level (x-axis)
Gain transfer curve and cursor showing amount of
compression
To indicate overshoots (when an incoming signal peak is too fast for the current compression
setting) the cursor temporarily leaves the gain
transfer curve.
The cursor changes color to indicate the amount
of compression applied, as shown in the following table:
Cursor Color
Compression Amount
white
no compression
light orange
below full ratio
dark orange
full ratio amount
See “De-Esser III Frequency Graph Display”
on page 75 for information on using the DeEsser’s graph display.
Dynamics graph display
Chapter 11: Dynamics III
67
Dynamics III Side-Chain Section
For information on using the Side-Chain section
of the Compressor/Limiter or Expander/Gate,
see “Using Dynamics III Key Input for SideChain Processing” on page 78.
Compressor/Limiter III
Of course, compression has many creative uses
that break these rules.
The Compressor/Limiter plug-in applies either
compression or limiting to audio material, depending on the ratio of compression used.
About Limiting
Compressor/Limiter III
Limiting is used to remove only occasional
peaks because gain reduction on successive
peaks would be noticeable. If audio material
contains many peaks, the threshold should be
raised and the gain manually reduced so that
only occasional, extreme peaks are limited.
About Compression
Compression reduces the dynamic range of signals that exceed a chosen threshold by a specific
amount. The Threshold control sets the level
that the signal must exceed to trigger compression. The Attack control sets how quickly the
compressor responds to the “front” of an audio
signal once it crosses the selected threshold. The
Release control sets the amount of time that it
takes for the compressor’s gain to return to its
original level after the input signal drops below
the selected threshold.
68
To use compression most effectively, the attack
time should be set so that signals exceed the
threshold level long enough to cause an increase
in the average level. This helps ensure that gain
reduction does not decrease the overall volume
too drastically, or eliminate desired attack transients in the program material.
Audio Plug-Ins Guide
Limiting prevents signal peaks from ever exceeding a chosen threshold, and is generally
used to prevent short-term peaks from reaching
their full amplitude. Used judiciously, limiting
produces higher average levels, while avoiding
overload (clipping or distortion), by limiting
only some short-term transients in the source
audio. To prevent the ear from hearing the gain
changes, extremely short attack and release
times are used.
Limiting generally begins with the ratio set at
10:1 and higher. Large ratios effectively limit
the dynamic range of the signal to a specific
value by setting an absolute ceiling for the dynamic range.
Compressor/Limiter III
Input/Output Level Meters
The Input and Output meters show peak signal
levels before and after dynamics processing. See
“Dynamics III Levels Section” on page 65 for
more information.
Unlike scales on analog compressors, metering
scales on a digital device reflect a 0 dB value that
indicates full scale (fs)—the full-code signal
level. There is no headroom above 0 dB.
Compressor/Limiter III Graph
Display
The Dynamics Graph display lets you visually
see how much expansion or gating you are applying to your audio material. See “Dynamics III
Graph Display” on page 67.
Compressor/Limiter III
Threshold Control
The Threshold (Thresh) control sets the level
that an input signal must exceed to trigger compression or limiting. Signals that exceed this
level will be compressed. Signals that are below
it will be unaffected.
This control has an approximate range of –60 dB
to 0 dB, with a setting of 0 dB equivalent to no
compression or limiting. The default value for
the Threshold control is –24 dB.
An orange arrow on the Input meter indicates
the current threshold, and can also be dragged
up or down to adjust the threshold setting.
Threshold indicator on Dynamics Graph display
This control ranges from –60 dB (lowest gain) to
0 dB (highest gain).
Compressor/Limiter III Ratio
Control
The Ratio control sets the compression ratio, or
the amount of compression applied as the input
signal exceeds the threshold. For example, a 2:1
compression ratio means that a 2 dB increase of
level above the threshold produces a 1 db increase in output.
This control ranges from 1:1 (no compression)
to 100:1 (hard limiting).
Compressor/Limiter III Attack
Control
The Attack control sets the attack time, or the
rate at which gain is reduced after the input signal crosses the threshold.
Threshold arrow on input meter
The smaller the value, the faster the attack. The
faster the attack, the more rapidly the Compressor/Limiter applies attenuation to the signal. If
you use fast attack times, you should generally
use a proportionally longer release time, particularly with material that contains many peaks in
close proximity.
The Dynamics Graph display also shows the
threshold as an orange vertical line.
This control ranges from 10 s (fastest attack
time) to 300 ms (slowest attack time).
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69
Compressor/Limiter III Release
Control
The Release control sets the length of time it
takes for the Compressor/Limiter to be fully deactivated after the input signal drops below the
threshold.
Release times should be set long enough that if
signal levels repeatedly rise above the threshold,
the gain reduction “recovers” smoothly. If the
release time is too short, the gain can rapidly
fluctuate as the compressor repeatedly tries to
recover from the gain reduction. If the release
time is too long, a loud section of the audio material could cause gain reduction that continues
through soft sections of program material without recovering.
This control ranges from 5 ms (fastest release
time) to 4 seconds (slowest release time).
Compressor/Limiter III Knee
Control
The Knee control sets the rate at which the compressor reaches full compression once the
threshold has been exceeded.
As you increase this control, it goes from applying “hard-knee” compression to “soft-knee”
compression:
• With hard-knee compression, compression
begins when the input signal exceeds the
threshold. This can sound abrupt and is ideal
for limiting.
• With soft-knee compression, gentle compression begins and increases gradually as the input signal approaches the threshold, and
reaches full compression after exceeding the
threshold. This creates smoother compression.
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Audio Plug-Ins Guide
Graph examples of hard knee (left) and soft knee
(right) compression
For example, a Knee setting of 10 dB would be
the gain range over which the ratio gradually increased to the set ratio amount.
The Gain Reduction meter displays light orange
while gain reduction has not exceeded the knee
setting, and switches to dark orange when gain
reduction reaches the full ratio.
This control ranges from 0 db (hardest response) to 30 db (softest response).
Compressor/Limiter III Gain
Control
The Gain control lets you boost overall output
gain to compensate for heavily compressed or
limited signals.
This control ranges from 0 dB (no gain boost) to
+40 dB (loudest gain boost), with the default
value at 0 dB.
For more information on the LFE channel,
refer to the Pro Tools Reference Guide.
Compressor/Limiter III SideChain Section
The side-chain is the split-off signal used by the
plug-in’s detector to trigger dynamics processing. The Side-Chain section lets you toggle the
side-chain between the internal input signal or
an external key input, and tailor the equalization of the side-chain signal so that the triggering of dynamics processing becomes frequencysensitive. See “Dynamics III Side-Chain Input”
on page 75.
Expander/Gate III
The Expander/Gate plug-in applies expansion or
gating to audio material, depending on the ratio
setting.
About Gating
Gating silences signals that fall below a chosen
threshold. To enable gating, simply set the Ratio
and Range controls to their maximum values.
Expanders can be thought of as soft noise gates
since they provide a gentler way of reducing
noisy low-level signals than the typically abrupt
cutoff of a gate.
Expander/Gate III
Input/Output Level Meters
The Input and Output meters show peak signal
levels before and after dynamics processing. See
“Dynamics III Levels Section” on page 65 for
more information.
Expander/Gate III Dynamics
Graph Display
The Dynamics Graph display lets you visually
see how much expansion or gating you are applying to your audio material. See “Dynamics III
Graph Display” on page 67.
Expander/Gate III
About Expansion
Expansion decreases the gain of signals that fall
below a chosen threshold. They are particularly
useful for reducing noise or signal leakage that
creeps into recorded material as its level falls, as
often occurs in the case of headphone leakage.
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71
Expander/Gate III Look Ahead
Button
Normally, dynamics processing begins when the
level of the input signal crosses the threshold.
When the Look Ahead button is enabled, dynamics processing begins 2 milliseconds before
the level of the input signal crosses the threshold.
Threshold arrow on Input meter
The Dynamics Graph display also shows the
threshold as an orange vertical line.
Look Ahead control
The Look Ahead control is useful for avoiding
the loss of transients that may have been otherwise cut off or trimmed in a signal.
Expander/Gate III Threshold
Control
The Threshold (Thresh) control sets the level below which an input signal must fall to trigger expansion or gating. Signals that fall below the
threshold will be reduced in gain. Signals that
are above it will be unaffected.
An orange arrow on the Input meter indicates
the current threshold, and can also be dragged
up or down to adjust the threshold setting.
Threshold indicator on Dynamics Graph display
This control has an approximate range of –60 dB
to 0 dB, with a setting of 0 dB equivalent to no
compression or limiting. The default value for
the Threshold control is –24 dB.
Expander/Gate III Ratio Control
The Ratio control sets the amount of expansion.
For example, if this is set to 2:1, it will lower signals below the threshold by one half. At higher
ratio levels (such as 30:1 or 40:1) the Expander/Gate functions like a gate by cutting off
signals that fall below the threshold. As you adjust the ratio control, refer to the built-in graph
to see how the shape of the expansion curve
changes.
This control ranges from 1:1 (no expansion) to
100:1 (gating).
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Audio Plug-Ins Guide
Expander/Gate III Attack
Control
Expander/Gate III Range
Control
The Attack control sets the attack time, or the
rate at which gain is reduced after the input signal crosses the threshold. Use this along with the
Ratio setting to control how soft the Expander’s
gain reduction curve is.
The Range control sets the depth of the Expander/Gate when closed. Setting the gate to
higher range levels allows more and more of the
gated audio that falls below the threshold to
peek through the gate at all times.
This control ranges from 10 s (fastest attack
time) to 300 ms (slowest attack time).
This control ranges from –80 dB (lowest depth)
to 0 dB (highest depth).
Expander/Gate III Hold Control
Expander/Gate III Side-Chain
Section
The Hold control specifies the duration (in seconds or milliseconds) during which the Expander/Gate will stay in effect after the initial
attack occurs. This can be used as a function to
keep the Expander/Gate in effect for longer periods of time with a single crossing of the threshold. It can also be used to prevent gate chatter
that may occur if varying input levels near the
threshold cause the gate to close and open very
rapidly.
This control ranges from 5 ms (shortest hold) to
4 seconds (longest hold).
Expander/Gate III Release
Control
The Release control sets how long it takes for the
gate to close after the input signal falls below the
threshold level and the hold time has passed.
This control ranges from 5 ms (fastest release
time) to 4 seconds (slowest release time).
The side-chain is the split-off signal used by the
plug-in’s detector to trigger dynamics processing. The Side-Chain section lets you toggle the
side-chain between the internal input signal or
an external key input, and tailor the equalization of the side-chain signal so that the triggering of dynamics processing becomes frequencysensitive. See “Dynamics III Side-Chain Input”
on page 75.
De-Esser III
The De-Esser reduces sibilants and other high
frequency noises that can occur in vocals, voiceovers, and wind instruments such as flutes.
These sounds can cause peaks in an audio signal
and lead to distortion.
The De-Esser reduces these unwanted sounds
using fast-acting compression. The Threshold
control sets the level above which compression
starts, and the Frequency (Freq) control sets the
frequency band in which the De-Esser operates.
Chapter 11: Dynamics III
73
Input
meter
Output meter
Gain
Reduction
meter
De-Esser III
Using De-Essing Effectively
To use de-essing most effectively, insert the DeEsser after compressor or limiter plug-ins.
The Frequency control should be set to remove
sibilants (typically the 4–10 kHz range) and not
other parts of the signal. This helps prevent deessing from changing the original character of
the audio material in an undesired manner.
Similarly, the Range control should be set to a
level low enough so that de-essing is triggered
only by sibilants. If the Range is set too high, a
loud, non-sibilant section of audio material
could cause unwanted gain reduction or cause
sibilants to be over-attenuated.
To improve de-essing of material that has both
very loud and very soft passages, automate the
Range control so that it is lower on soft sections.
.
The De-Esser has no control to directly adjust the threshold level (the level that an input signal must exceed to trigger de-essing).
The amount of de-essing will vary with the
input signal.
De-Esser III Level Meters
These controls let you track input, output, and
gain reduction levels.
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Audio Plug-Ins Guide
De-Esser III I/O Meter display
Input and Output Meters
The Input and Output meters show peak signal
levels before and after dynamics processing:
Green Indicates nominal levels.
Yellow Indicates pre-clipping levels, starting at
–6 dB below full scale.
Red Indicates full scale levels (clipping).
The Clip indicators at the top of each meter indicate clipping at the input or output stage of
the plug-in. Clip indicators can be cleared by
clicking the indicator.
De-Esser III Gain Reduction
Meter
The Gain Reduction meter indicates the amount
the input signal is attenuated, in dB. This meter
shows different colors during de-essing:
Light Orange Indicates that gain reduction is being applied, but has not reached the maximum
level set by the Range control.
Dark Orange Indicates that gain reduction has
reached the maximum level set by the Range
control.
De-Esser III Frequency Control
The Frequency (Freq) control sets the frequency
band in which the De-Esser operates. When HF
Only is disabled, gain is reduced in frequencies
within the specified range. When HF Only is enabled, the gain of frequencies above the specified value will be reduced.
This control ranges from 500 Hz (lowest frequency) to 16 kHz (highest frequency).
De-Esser III Range Control
The Range control defines the maximum
amount of gain reduction possible when a signal
is detected at the frequency set by the Frequency
control.
This control ranges from –40 dB (maximum deessing) to 0 dB (no de-essing).
De-Esser III Frequency Graph
Display
The De-Esser Frequency Graph display shows a
curve that represents the level of gain reduction
(on the y-axis) for the range of the output signal's frequency (on the x-axis). The white line
represents the current Frequency setting, and
the animated orange line represents the level of
gain reduction being applied to the signal.
Use this graph as a visual guideline to see how
much dynamics processing you are applying at
different points in the frequency spectrum.
Current gain reduction
Frequency
Gain
(y-axis)
Range
De-Esser III HF Only Control
When the HF Only button is enabled, gain reduction is applied only to the active frequency
band set by the Frequency control. When the HF
Only button is disabled, the De-Esser applies
gain reduction to the entire signal.
De-Esser III Listen Control
When enabled, the Listen button lets you monitor the sibilant peaks used by the De-Esser as a
side-chain to trigger compression. This is useful
for listening only to the sibilance for fine-tuning
De-Esser controls. To monitor the whole output
signal without this filtering, deselect the Listen
button.
Frequency
(x-axis)
De-Esser graph display
Dynamics III Side-Chain Input
(Compressor/Limiter and Expander/Gate Only)
Dynamics processors typically use the detected
amplitude of their input signal to trigger gain
reduction. This split-off signal is known as the
side-chain. The Compressor/Limiter and Expander/Gate plug-ins feature external key capabilities and filters for the side-chain.
Chapter 11: Dynamics III
75
With external key side-chain processing, you
trigger dynamics processing using an external
signal (such as a separate reference track or audio source) instead of the input signal. This external source is known as the key input.
With side-chain filters, you can make dynamics
processing more or less sensitive to certain frequencies. For example, you might configure the
side-chain so that certain lower frequencies on a
drum track trigger dynamics processing.
Dynamics III Side-Chain
Controls
The controls in the Side-Chain section let you
toggle the side-chain between the internal input
signal or an external key input, listen to the
side-chain, and tailor the equalization of the
side-chain signal so that the triggering of dynamics processing becomes frequency-sensitive.
External Key button
Dynamics III Side-Chain Listen
When enabled, this control lets you listen to the
internal or external side-chain input by itself, as
well as monitor its levels with the Output meter.
This is especially useful for fine-tuning the
plug-in’s filter settings or external key input.
Side-Chain Listen button
Side-Chain Listen is not saved with other
plug-in presets.
Dynamics III Side-Chain HF and LF
Filter Enable Buttons
The HF Filter Enable and LF Filter Enable buttons toggle the corresponding filter in or out of
the side-chain. When this button is highlighted,
the filter is applied to the side-chain signal.
When this button is dark gray, the filter is bypassed and available for activation.
Compressor/Limiter and Expander/Gate Side-Chain
Dynamics III Side-Chain External Key
The External Key toggles external side-chain
processing on or off. When this button is highlighted, the plug-in uses the amplitude of a separate reference track or external audio source to
trigger dynamics processing. When this button
is dark gray, the External Key is disabled and the
plug-in uses the amplitude of the input signal to
trigger dynamics processing.
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Audio Plug-Ins Guide
HF and LF Filter Enable buttons
Dynamics III Side-Chain HighFrequency (HF) Filter Type
The HF filter section lets you filter higher frequencies out of the side-chain signal so that only
certain bands of high frequencies or lower frequencies pass through to trigger dynamics processing. The HF side-chain filter is switchable
between Band Pass and Low Pass filters.
Band Pass Filter Makes triggering of dynamics
processing more sensitive to frequencies within
the narrow band centered around the Frequency
setting, and rolling off at a slope of 12 dB per octave.
Band-Pass filter
Low Pass Filter Makes triggering of dynamics
processing more sensitive to frequencies below
the Frequency setting rolling off at a slope of
12 dB per octave.
HF frequency controls
Dynamics III Side-Chain Low-Frequency
(LF) Filter Type
The LF filter section lets you filter lower frequencies out of the side-chain signal so that only
certain bands of low frequencies or higher frequencies are allowed to pass through to trigger
dynamics processing. The LF side-chain is switchable between Band Pass and High Pass filters.
Band-Pass Filter Makes triggering of dynamics
processing more sensitive to frequencies within
the narrow band centered around the Frequency
setting, and rolling off at a slope of 12 dB per octave.
Band-Pass filter
High Pass Filter Makes triggering of dynamics
Low Pass filter
Dynamics III Side-Chain HF Frequency
Control
processing more sensitive to frequencies above
the Frequency setting rolling off at a slope of
12 dB per octave.
The HF frequency control sets the frequency position for the Band Pass or Low Pass filter, and
ranges from 80 Hz to 20 kHz.
Chapter 11: Dynamics III
77
High Pass filter
Dynamics III Side-Chain LF Frequency
Control
Selecting a Key Input
Click External Key to activate external sidechain processing.
2
The Frequency control sets the frequency position for the Band-Pass or High Pass filter, and
ranges from 25 Hz to 4 kHz.
External Key
3 To listen to the signal that will be used to control side-chain input, click Side-Chain Listen to
enable it (highlighted).
LF frequency controls
Using Dynamics III Key Input for
Side-Chain Processing
To use a filtered or unfiltered external key input to
trigger dynamics processing:
Click the Key Input selector and select the input or bus carrying the audio from the reference
track or external audio source.
1
Side-Chain Listen
To filter the key input so that only specific frequencies trigger the plug-in, use the HF and LF
controls to select the desired frequency range.
4
Begin playback. The plug-in uses the input or
bus that you chose as an external key input to
trigger its effect.
5
Adjust the plug-in’s Threshold (Thresh) control to fine-tune external key input triggering.
6
7 Adjust other controls to achieve the desired effect.
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Audio Plug-Ins Guide
Using a Filtered Input Signal for
Side-Chain Processing with
Dynamics III
To use the filtered input signal to trigger dynamics
processing:
1 Ensure the Key Input selector is set to No Key
Input.
Key Input selector
2 Ensure that the External Key button is disabled
(dark gray).
Side-Chain section
3 To listen to the signal that will be used to control side-chain input, click Side-Chain Listen to
enable it (highlighted).
Side-Chain section
To filter the side-chain input so that only specific frequencies within the input signal trigger
the plug-in, use the HF and LF controls to select
the desired frequency range.
4
Begin playback. The plug-in uses the filtered
input signal to trigger dynamics processing.
5
To fine-tune side-chain triggering, adjust the
plug-in controls.
6
Chapter 11: Dynamics III
79
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Audio Plug-Ins Guide
Chapter 12: Fairchild Plug-Ins
The Fairchild plug-ins are a pair of vintage compressor plug-ins that are available in TDM,
RTAS, and AudioSuite formats. The following
plug-ins are included:
• Fairchild 660 (see “Fairchild 660” on
page 81)
• Fairchild 670 (see “Fairchild 670” on
page 83)
Fairchild 660
(TDM, RTAS, and AudioSuite)
Re-introducing the undisputed champion in
price, weight, and performance: the $35,000,
one-hundred pound, Fairchild 660.
Bomb Factory’s no-compromise replica captures every detail of this studio classic.
How the Fairchild 660 Works
Designed in the early 1950s, the Fairchild 660 is
a variable-mu tube limiter. Variable-mu designs
use an unusual form of vacuum tube that is capable of changing its gain dynamically.
The result? In addition to featuring a tube audio
stage like the LA-2A, the Fairchild actually
achieves gain reduction through the use of
tubes!
The heart of the Fairchild limiter—a 6386 triode—is one such variable-mu tube. In fact, four
of these tubes are used in parallel. A key part of
the Fairchild design, it ensures that the output
doesn’t get darker as the unit goes further into
gain reduction, and also reduces distortion as
the tubes are biased further into Class-B operation.
Tubes, wires, and iron
Bomb Factory Fairchild 660
Chapter 12: Fairchild Plug-Ins
81
Fairchild 660 Controls
Fairchild 660 Tips and Tricks
Adjust the Input Gain and Threshold controls
together until you get the sound you want. Like
many classic compressors, after a little bit of
tweaking, you’ll know immediately when you
get it right.
5,6,7,8…
Input Gain Input Gain sets the input level to the
unit and the compression threshold, just like the
Input control on an 1176. Full clockwise is loudest.
Threshold Threshold adjusts the gain to the
sidechain, just like the Peak Reduction control
on an LA-2A.
Time Constant Selects the attack and release
times for the compressor. One is fastest, and six
is slowest. Seven and eight are Bomb Factory
custom presets.
The Fairchild manual documents Time Constant
settings 5 and 6 as user presets—although you
have to go inside with a soldering iron to change
them. We used the “factory default” values.
Bonus Settings
Settings 7 and 8 do not exist on real-world
units—well, at least most of them. These settings are taken from a real-world Fairchild modification invented by Dave Amels many years before he designed the plug-in version.
What do they do? Settings 7 and 8 offer versions
of Time Constant 2 with a gentler release useful
for compressing vocals and other program material where you desire more subtlety in the
compression action. Give them a try—you’ve already heard them on hit songs on the radio.
Pump It Up
With a carefully adjusted Input Gain and
Threshold, you can use Time Constant 1 to
achieve a cool pumping effect on drums. The
sound gets darker and fuller, and sits beautifully
in a mix.
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Fairchild 670
(TDM and RTAS)
Bomb Factory’s no-compromise replica captures every detail of the Fairchild 670. The
Fairchild 670 is a dual-channel unit and, as
such, is only available on stereo tracks.
Note that the companion Fairchild 660 also supports stereo operation. Bomb Factory modeled
both a Fairchild 660 and a Fairchild 670 from
scratch using two different hardware units. This
gives you a choice of two different-sounding
Fairchild units to try on your stereo tracks!
How the Fairchild 670 Works
The Fairchild 670’s internal design is similar to
the Fairchild 660. However, the Fairchild 670 offers two channels of compression instead of one.
Combined with the AGC control, this gives you
even more compression options on stereo
tracks.
Fairchild 670 Controls
Adjust the Input Gain and Threshold controls
together on both channels until you get the
sound you want. Like many classic compressors,
after a little bit of tweaking, you’ll know immediately when you get it right.
Input Gain Sets the input level to the unit and
the compression threshold, just like the Input
control on an 1176. Full clockwise is loudest.
Threshold Adjusts the gain to the sidechain, just
like the Peak Reduction control on an LA-2A.
Time Constant Selects the attack and release
times. One is fastest, and six is slowest. Seven
and eight are Bomb Factory custom presets. See
“Fairchild 670” on page 83 for details on these
custom settings.
AGC Lets you select Left/Right processing or
Lat/Vert processing of the two channels.
Left/Right works like a dual-mono compressor
with separate controls for the left and right
channels. In Lat/Vert mode the top row of controls affects the in-phase (Lat) information and
the bottom row of controls affects the out of
phase (Vert) information. Although originally
designed for vinyl mastering where excess Vert
(vertical) information could cause the needle to
jump out of the groove, you can use the Lat/Vert
mode to achieve some amazing effects – especially on drums.
Fairchild 670 Tips and Tricks
Fairchild 670
To exactly match the settings between channels,
hold down the Alt key (Windows) or the Shift
key (Mac) while dragging the mouse. This is useful when trying to preserve the existing
Left/Right balance on stereo material.
Chapter 12: Fairchild Plug-Ins
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Chapter 13: Impact
Impact is a high-quality TDM dynamics processing plug-in.
Impact Controls
The Impact plug-in provides critical control
over the dynamic range of audio signals, with
the look and sound of a mixing console’s stereobus compressor.
Impact Ratio Control
Impact provides support for 192 kHz,
176.4 kHz, 96 kHz, 88.2 kHz, 48 kHz, and
44.1 kHz sessions.
Impact provides support for mono, stereo, and
all Pro Tools-supported multichannel audio formats.
Ratio sets the compression ratio. If the ratio is
set to 2:1 for example, it will compress changes
in signals above the threshold by one half. This
control provides four fixed compression ratios,
2:1, 4:1, 10:1, and 20:1. Selecting 2:1 applies very
light compression; selecting 20:1 applies heavy
compression, bordering on limiting.
Impact requires one or more
Pro Tools|HD Accel cards.
Ratio
Impact plug-in
Chapter 13: Impact
85
Impact Attack Control
Impact Release Control
Attack sets the compressor attack time. To use
compression most effectively, the attack time
should be set so that signals exceed the threshold level long enough to cause an increase in the
average level. This helps ensure that gain reduction does not decrease the overall volume. The
range of this control is from 0.1 ms to 30.0 ms.
Release controls the length of time it takes for
the compressor to be fully deactivated after the
input signal drops below the threshold level. In
general, this setting should be longer than the
attack time and long enough that if signal levels
repeatedly rise above the threshold, they cause
gain reduction only once. If the release time is
too long, a loud segment of audio material could
cause gain reduction to persist through a lowvolume segment (if one follows). Setting this
control to its maximum value, Auto, selects a release time that is program dependent, based on
the audio being processed. The range of this
control is from 20 milliseconds to 2.5 seconds.
Attack
Impact Threshold Control
Threshold sets the decibel level that a signal
must exceed for Impact to begin applying compression. Signals that exceed the Threshold will
be compressed by the amount of gain reduction
set with the Ratio control. Signals that are below
the Threshold will be unaffected. The range of
the Threshold control is from –70 dB to –0 dB.
A setting of –0 dB is equivalent to no compression.
Threshold
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Release
Impact Make-up Control
Impact Ext Control (Side-Chain)
Make-Up adjusts the overall output gain. Because large amounts of compression can restrict
dynamic range, the Make-Up control is useful
for compensating for heavily compressed signals and making up the resulting difference in
level. When Impact is used on stereo or multichannel tracks, the Make-Up control determines
master output levels for all channels. The range
of this control is from 0 dB of attenuation to
+40 dB of gain.
External On/Off enables and disables side-chain
processing. With side-chain processing you can
trigger compression from a separate reference
track or external audio source. The source used
for triggering side-chain processing is referred
to as the Key Input.
See “Using the Impact Compressor” on
page 88 for instructions on setting up and
using a key input.
External On/Off
Make-Up
Applying large amounts of Make-Up gain
will boost the level of any noise or hiss present in audio material, making it more audible.
Impact Listen On/Off Control
Key Listen On/Off enables and disables auditioning of the Key Input (the reference track or
external audio source used for triggering sidechain processing). This is useful for fine-tuning
Impact’s compression settings to the Key Input.
Listen On/Off
Chapter 13: Impact
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Impact Gain Reduction Meter
The Gain Reduction meter is an analog-style
meter that indicates the amount of gain reduction in dB. The range of this meter is from 0 dB
to 40 dB. The gain reduction meter displays the
amount of gain reduction linearly from 0–20 db,
and non-linearly from 20–40 dB.
A red clip indicator appears at the top of each
meter. Clicking a clip indicator clears it. Altclicking (Windows) or Option-clicking (Mac)
clears the clip indicators on all channels.
Clip indicator in Input/Output meters (mono shown)
Gain Reduction meter
Impact Meters
The Input/Output meters indicate input and
output signal levels in dB. When Impact is used
in mono or stereo, both input and output meters
are displayed. When Impact is used in a multichannel format, only output meters are displayed by default. You can toggle the meter display to show only input meters by clicking the
blue-green rectangle at the lower right of the
meter display.
Toggling the meter display in the Output meters
(5.1 surround format shown)
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Audio Plug-Ins Guide
Input/Output meters (stereo shown)
Using the Impact
Compressor
Compressors reduce the dynamic range of audio
signals that exceed a user-selectable threshold
by a specific amount. This is accomplished by
reducing output levels as input levels increase
above the threshold.
The amount of output level reduction that Impact applies as input levels increase is referred
to as the compression ratio. This parameter is adjustable in discrete increments. If you set the
compression ratio to 2 (a ratio of 2:1), for each
2 dB that the signal exceeds the threshold, the
output will increase only by 1 dB. With a setting
of 4 (a ratio of 4:1), an 8 dB increase in input
will produce only a 2 dB increase in output.
Side-Chain Processing
Compressors generally use the detected amplitude of their input signal as a control source.
However, you can also use other signals, such as
a separate reference track or an external audio
signal as a control source. This is known as sidechain processing.
Side-chain processing lets you control Impact
compression using an independent audio signal
(typically, another Pro Tools track). In this way
you can compress the audio of one track using
the dynamics of a different audio track.
To use a Key Input signal for side-chain
processing:
1 Click the Send button and select a bus path for
the audio track or Auxiliary Input you want to
use as the side-chain signal.
From Impact’s Key Input menu, select the input or bus path carrying the audio you want to
use as the side-chain signal to trigger Impact
compression. The Key Input source must be
monophonic.
2
The reference track or external audio source
used for triggering side-chain processing is referred to as the Key Input.
Using the Impact Side-Chain
Input
Impact provides side-chain processing capabilities. Compressors typically use the detected amplitude of their input signal to cause gain reduction. This split-off signal is called the side-chain.
However, an external signal (referred to as the
Key Input) can be used to trigger compression.
A typical use for side-chain processing is to control the dynamics of one audio signal using the
dynamics of another signal (referred to as the
Key Input). For example, you could use a lead
vocal track to trigger compression of a background vocal track so that their dynamics
match.
Selecting a Key Input
To activate external side-chain processing,
click Ext.
3
Begin playback. Impact uses the input or bus
that you selected as a Key Input to trigger its effect.
4
If you want to hear the audio source you have
selected as the side-chain input, click Listen. (To
stop listening to the side-chain input, click Listen again).
5
Remember to disable Listen to resume normal plug-in monitoring.
Adjust Impact’s Threshold parameter to finetune Key Input triggering.
6
Adjust other parameters to achieve the desired
effect.
7
Chapter 13: Impact
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Chapter 14: JOEMEEK SC2 Compressor
The JOEMEEK SC2 Compressor is a dynamics
processing plug-in is available in TDM, RTAS,
and AudioSuite formats.
In use by top producers the world over,
JOEMEEK compression is the secret weapon
that gives your sound the character and excitement it deserves!
JOEMEEK Compressor
Controls
The SC2 Compressor provides the following
controls:
Input Gain Input Gain adjusts the input level to
the compressor.
Compression The Compression control affects
the gain to the side-chain of the compressor. Use
it along with Slope to adjust the amount of compression.
Output Gain Output Gain provides makeup gain
after compression.
JOEMEEK SC2 Compressor
How the JOEMEEK SC2 Compressor Works
Legendary producer Joe Meek used to say, “If it
sounds right, it is right.” Nowhere is this more
apparent than in Joe Meek’s masterful use of
non-linear, sometimes severe compression in
his productions.
The JOEMEEK Compressor is designed purely as
an effects compressor. Its purpose is to change
the way the ear perceives sound; its action
changes the clarity, balance and even rhythmic
feel of music.
Slope Slope is similar to the compression ratio
controls found on other compressors. However,
on the JOEMEEK, the actual ratio varies based
on program material so the term Slope is used
instead. In practice, 1 is very gentle compression and 2 or 3 are typically right for voice and
submixes. The higher numbers are better for instruments and extreme sounds. (At the suggestion of the original designers, Bomb Factory
added the 5 setting found on the later-model JOEMEEK SC2.2. Use 5 to create severe pumping
effects.)
Attack Attack sets the time that the compressor
takes to act. Slower attacks are typically used
when the sound of the compression needs to be
less obvious.
Chapter 14: JOEMEEK SC2 Compressor
91
Release Release sets the time during which signal returns to normal after compression. With
longer release times, the compression is less noticeable.
To hear it, use a drum track, set Slope to 5, and
Attack and Release to Fast. Used sparingly, this
effect can contribute to musical drive in your
tracks.
Attack/Release Times
JOEMEEK Compressor Tips
and Tricks
Not Perfect. Just Right
Standard engineering practice says that a compressor should work logarithmically. For a certain increase of volume, the output volume
should rise proportionally less, with a result
that the more you put in, the more it’s pushed
down.
The JOEMEEK compressor doesn’t work this
way. As volume increases at the input, a point is
reached where the compressor starts to work
and the gain through the amplifier is reduced. If
the input level keeps rising, gradually the gain
reduction becomes less effective and the amplifier goes back to being a linear amplifier except
with the volume turned down.
This is by design, and is based on an understanding of how the human ear behaves! The result is that the listener is fooled into thinking
that the JOEMEEK compressed sound is louder
than it really is—but without the strange psycho-acoustic effect of “deadness” that other
compressors suffer from.
Overshoot
At fast Attack settings, it is possible to make the
JOEMEEK “overshoot” on percussive program
material. This means that the compression electronics are driven hard before the light cells respond to the increased level. The cells catch up
and overcompress momentarily giving a tiny dip
immediately following the start of the note.
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Audio Plug-Ins Guide
It may be difficult to understand the interactions between the Attack and Release controls,
because the JOEMEEK Compressor behaves very
differently than typical compressors. Experimentation is the best option, but an explanation
may help you understand what’s going on.
The JOEMEEK Compressor uses a compound release circuit that reacts quickly to short bursts
of volume, and less quickly to sustained volume.
While the unit was being prototyped and designed, the values and ranges of these timings
were chosen by experimentation using wide
ranges of program material.
Because of these intentional effects produced by
the compressor, the JOEMEEK makes a perfect
tool for general enhancement of tracks to
“brighten,” “tighten,” “clarify,” and catch the
attention of the listener, functions that are difficult or impossible to achieve with conventional compressor designs.
Chapter 15: Maxim
Maxim is a unique and powerful peak-limiting
and sound maximizing plug-in that is available
in TDM, RTAS, and AudioSuite formats. Maxim
is ideal for critical mastering applications, as
well as standard peak-limiting tasks.
 Online Help (accessed by clicking a control
name) provides descriptions of each control.
Maxim offers several critical advantages over
traditional hardware-based limiters. Most significantly, Maxim takes full advantage of the
random-access nature of disk-based recording
to anticipate peaks in audio material and preserve their attack transients when performing
reduction.
This makes Maxim more transparent than conventional limiters, since it preserves the character of the original audio signal without clipping
peaks or introducing distortion.
The multichannel TDM version of Maxim is
not supported at 192 kHz. Use the multimono TDM or RTAS version instead.
Maxim features include:
 “Perfect attack-limiting” through look-ahead
analysis accurately preserves transient attacks
and the character of original program material.
 A full-color histogram plots input dB history
during playback and provides visual feedback
for setting threshold level.
 A user-adjustable ceiling lets material be
level-optimized for recording.
Maxim
About Peak Limiting
Peak limiting is an important element of audio
production. It is the process of preventing signal
peaks in audio material from clipping by limiting their dynamic range to an absolute, user-selectable ceiling and not letting them exceed this
ceiling.
Limiters let you select a threshold in decibels. If
an audio signal peak exceeds this threshold, gain
reduction is applied, and the audio is attenuated
by a user-selectable amount.
 Dither for noise shaping during the final mixdown.
Chapter 15: Maxim
93
Limiting has two main uses in the audio production cycle:
• Adjusting the dynamic range of an entire final
mixdown for premastering purposes
• Adjusting the dynamic range of individual instruments for creative purposes
Limiting a Mixdown
The purpose of applying limiting during final
mixdown is to flatten any large peaks remaining
in the audio material to have a higher average
signal level in the final mix. By flattening peaks
that would otherwise clip, it is possible to increase the overall level of the rest of the mix.
This results in higher average audio levels, potentially better signal to noise ratio, and a
smoother mix.
Limiting Individual Instruments
The primary purpose of applying limiting to individual instruments is to alter their dynamic
range in subtle or not-so-subtle ways. A common application of this type of limiting is to
modify the character of drums. Many engineers
do this by applying heavy limiting to flatten the
snap of the attack portion of a drum hit. By adjusting the release time of the limiter it is possible to bring up room tone contained in the decay
portion of the drum sound.
In some cases, this type of limiting can actually
change a drum’s character from a very dry
sound to a relatively wet sound if there is
enough room tone present. This method is not
without its drawbacks, however, since it can also
bring noise levels up in the source audio if present.
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How Maxim Differs From Conventional Limiters
Maxim is superior to conventional limiters in
several ways. Unlike traditional limiters, Maxim
has the ability to anticipate signal peaks and respond instantaneously with a true zero attack
time.
Maxim does this by buffering audio with a 1024sample delay while looking ahead and analyzing
audio material on disk before applying limiting.
Maxim can then instantly apply limiting before
a peak builds up. The result is extremely transparent limiting that faithfully preserves the attack transients and retains the overall character
of the original unprocessed signal.
In addition, Maxim provides a histogram, that
displays the distribution of waveform peaks in
the audio signal. This provides a convenient visual reference for comparing the density of
waveform peaks at different decibel levels and
choosing how much limiting to apply to the material.
The TDM version of Maxim introduces 1028
samples of delay at 48 kHz into any processed signal. The RTAS version of Maxim
introduces 1024 samples of delay. These delays will increase proportionally at higher
sample rates. To preserve phase synchronicity between multiple audio sources when
Maxim is only applied to one of these
sources, use Delay Compensation, or the
Time Adjuster plug-in to compensate.
Maxim Controls and Meters
Maxim Input Level Meter
This meter displays the amplitude of input signals prior to limiting. Unlike conventional meters, Maxim’s Input meter displays the top 24 dB
of dynamic range of audio signals, which is
where limiting is typically performed. This provides you with much greater metering resolution within this range so that you can work with
greater precision.
By dragging the Threshold slider downwards,
you can visually adjust the level at which limiting will occur. Maxim displays the affected
range in orange.
dB level of
waveform
peaks
Maxim Histogram
The Histogram displays the distribution of
waveform peaks in the audio signal. This graph
is based on audio playback. If you select and
play a short loop, the histogram is based on that
data. If you select and play a longer section, the
Histogram is based on that. Maxim holds peak
data until you click the Histogram to clear it.
The Histogram provides a visual reference for
comparing the density of waveform peaks at different decibel levels. You can then base limiting
decisions on this data.
The X axis of the Histogram shows the number
of waveform peaks occurring at specific dB levels. The Y axis shows the specific dB level at
which these peaks occur. The more waveform
peaks that occur at a specific dB level, the longer
the X-axis line. If there appears to be a pronounced spike at a certain dB level (4 dB for example), it means that there are a relatively large
number of waveform peaks occurring at that
level. You can then use this information to decide how much limiting to apply to the signal.
density of
waveform
peaks at
each level
Histogram
Maxim Threshold Slider
This slider sets the threshold level for limiting.
Signals that exceed this level will be limited. Signals below it will be unaffected. Limited signal
peaks are attenuated to match the threshold
level, so the value that you set here will determine the amount of reduction applied.
Maxim Output Meter
This meter displays the amplitude of the output
signal. The value that appears here represents
the processed signal after the threshold, ceiling,
and mixing settings have been applied.
Maxim Ceiling Slider
This slider determines the maximum output
level. After limiting is performed you can use
this slider to adjust the final output gain. The
value that you set here will be the absolute ceiling level for limited peaks.
Chapter 15: Maxim
95
Maxim Attenuation Meter
Maxim Mix Slider
This meter displays the amount of gain reduction being applied over the course of playback,
with the maximum peak displayed in the numeric readout at the bottom of the meter. For
example, if the numerical display at the bottom
of the Attenuation meter displays a value of
4 dB, it means that 4 dB of limiting has occurred. Since this is a peak-hold readout, you
can temporarily walk away from a session during playback and still know the maximum gain
reduction value when you come back. To clear
the numeric readout, click it with the mouse.
This slider sets the ratio of dry signal to limited
signal. In general, if you are applying Maxim to
a main output mix, you will probably want to set
this control to 100% wet. If you are applying
heavy limiting to an individual track or element
in a mix to modify its character, this control is
particularly useful since it lets you add precisely
the desired amount of the processed effect to the
original signal.
Release Slider
This slider sets how long it takes for Maxim to
ease off of its attenuation after the input signal
drops below the threshold level. Because Maxim
has an attack time of zero milliseconds, the release slider has a very noticeable effect on the
character of limiting. In general, if you are using
heavy limiting, you should use proportionally
longer release times in order to avoid pumping
that may occur when Maxim is forced to jump
back and forth between limited and unlimited
signal levels. Lengthening the release time has
the effect of smoothing out these changes in
level by introducing a lag in the ramp-up or
ramp-down time of attenuation. Use short release times on material with peaks that are relatively few in number and that do not occur in
close proximity to each other. The Release control has a default value of 1 millisecond.
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Audio Plug-Ins Guide
Maxim Link Button
When depressed, this button (located between
the Threshold and Ceiling numeric readouts)
links the Threshold and Ceiling controls. These
two sliders will then move proportionally together. As you lower the Threshold control, the
Ceiling control is lowered as well. When these
controls are linked you can conveniently compare the effect of limiting at unity gain by clicking the Bypass button.
Link button
Maxim Dither Button
Maxim Bit Resolution Button
When selected, this applies dither. Dither is a
form of randomized noise used to minimize
quantization artifacts in digital audio systems.
Quantization artifacts are most audible when
the audio signal is near the low end of its dynamic range, such as during a quiet passage or
fade-out.
These buttons select dither bit resolution. In
general, set this control to the maximum bit resolution of your destination media.
Applying dither helps reduce quantization noise
that can occur when you are mixing from a
24-bit source to a 16-bit destination, such as
CD-R or DAT. If you are using Maxim on a Master Fader during mixdown, Maxim’s built-in
dither function saves you the trouble and DSP
resources of having to use a separate Dither
plug-in.
If Dither is disabled, the Noise Shaping and Bit
Resolution controls will have no effect.
Maxim Noise Shaping Control
 16-bit is recommended for output to digital
devices such as DAT recorders and CD recorders
since they have a maximum resolution of 16 bits.
 18-bit is recommended for output to digital
devices that have a maximum resolution of
18 bits.
 20-bit is recommended for output to digital
devices that support a full 20-bit recording data
path. Use this setting for output to analog devices using an 882|20 I/O audio interface. It is
also recommended for use with digital effects
devices that support 20-bit input and output,
since it provides for a lower noise floor and
greater dynamic range when mixing 20-bit signals directly into Pro Tools.
When selected, this applies noise-shaped dither.
Noise shaping biases the dither noise to less audible high frequencies so that it is not as readily
perceived by the ear. Dither must be enabled in
order to use Noise Shaping.
Chapter 15: Maxim
97
Using Maxim
Following are suggestions for using Maxim most
effectively.
To use Maxim:
1
Insert Maxim on the desired track.
Select the portion of the track containing the
most prominent audio peaks.
2
Loop playback and look at the data displayed
by the Histogram and Attenuator meter.
In general, a value of 0.5 dB or so is a good maximum ceiling. Don’t set the ceiling to zero, since
the digital-to-analog convertors on some DATs
and CD players will clip at or slightly below zero.
If you are using Maxim on an output mix
that will be faded out, enable the dithering
options you want to improve the signal performance of the material as it fades to lower
amplitudes.
3
Select the Link button to link the Threshold
and Ceiling controls. You can then adjust these
controls together proportionally and, using the
Bypass button, compare the audio with and
without limiting.
4
Adjust the Threshold downwards until you
hear and see limiting occur, then bring the
Threshold back up slightly until you have
roughly the amount of limiting you want.
5
Periodically click and clear the Attenuation
meter to check attenuation. In general, applying
2 dB to 4 dB of attenuation to occasional peaks
in pop-oriented material is appropriate.
6
Use the Bypass button to compare the processed and unprocessed sound and to check if
the results are acceptable.
7
8 Avoid pumping effects with heavier limiting by
setting the Release slider to longer values.
When you get the effect you want, deselect the
Link button and raise the output level with the
Ceiling slider to maximize signal levels without
clipping.
9
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Audio Plug-Ins Guide
Maxim and Mastering
If you intend to deliver audio material as a
24-bit audio file on disk for professional mastering, be aware that many mastering engineers
prefer material delivered without dither or level
optimization.
Mastering engineers typically want to receive
audio material as undisturbed as possible in order to have leeway to adjust the level of the material relative to other material on a CD. In such
cases, it is advisable to apply only the limiting
that you find creatively appropriate—adding a
little punch to certain instruments in the mix,
for example.
However, if you intend to output the material to
DAT or CD-R, use appropriate limiting and add
dither. Doing so will optimize the dynamic
range and preserve the activity of the lower, or
least significant bits in the audio signal,
smoothly dithering them into the 16-bit output.
Chapter 16: Purple Audio MC77
Purple MC77 is a dynamics processing plug-in
that is available in TDM, RTAS, and AudioSuite
formats.
The Purple Audio MC77 is a spot-on digital replica of Andrew Roberts’s acclaimed MC77 Limiting amplifier, which in turn is an update of his
classic MC76 hardware unit. Representing a different take on the 1176-style FET limiter, the
Bomb Factory Purple Audio MC77 preserves every audio nuance and sonic subtlety of the classic originals.
The Bomb Factory MC77
How the Bomb Factory Purple Audio MC77
Works
The Bomb Factory Purple Audio MC77 has controls identical in name to those of the Bomb Factory BF76, and which function similarly. For
more information, see Chapter 9, “BF76.”
Chapter 16: Purple Audio MC77
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Chapter 17: Slightly Rude Compressor
Slightly Rude Compressor is a dynamics processing plug-in that is available in TDM, RTAS,
and AudioSuite formats.
The Slightly Rude Compressor is the first completely custom compressor designed by Bomb
Factory. Used conservatively, it sounds beautiful on vocals, drums, guitars, and piano. Pushed
hard, it's unique and aggressive. The stereo version is specifically designed to solve the problems that often plague digital mixes.
Slightly Rude Compressor
Controls
The Slightly Rude Compressor is not based on
any specific piece of vintage hardware. It is a
completely custom design that features all of
Bomb Factory’s knowledge and expertise in the
realm of digital compression.
Slightly Rude Compressor provides the following controls:
s
Input Amount Sets the input level to the unit and
the compression threshold, just like the Input
control on an 1176. Full clockwise is loudest.
Make-Up Gain Adds gain after compression. It
works just like the Gain control on an LA-2A.
Release Time Adjusts the release time; full
clockwise is fastest and provides the most
“pump.”
Rudeness Affects the sound of the compression
action.
Slightly Rude/Super Rude Switch Affects the
Slightly Rude Compressor
sound of the compression action.
Chapter 17: Slightly Rude Compressor
101
Slightly Rude Compressor
Tips and Tricks
For a classic sound, use the “Slightly Rude” setting and keep the Rudeness control below the
half-way point. Settings above 50% will increase
the aggressiveness of the compressed sound.
To achieve more dynamic effects, switch to the
“Super Rude” mode. In this mode, the Rudeness
knob controls the amount of overshoot in the
compressor. This results in a distinctive processed sound on percussive material, especially
on piano and drums.
Try chaining the Slightly Rude Compressor either before or after other compressors. Using
the Bomb Factory Fairchild 660 (or 670) or
Bomb Factory BF76 before or after the Slightly
Rude Compressor will give you an amazing variety of compression options—especially if you
experiment with the Super Rude mode.
Also be sure to try the Slightly Rude Compressor
on full mixes or stereo submixes! It adds the
“glue” that helps hold mixes together, something that’s often hard to achieve in the digital
domain.
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Audio Plug-Ins Guide
Chapter 18: Smack!
Smack! is a dynamics processing plug-in that is available in TDM, RTAS, and AudioSuite formats.
Smack! Plug-In (TDM version shown)
The Smack! compressor/limiter plug-in has the
following features.
• Three modes of compression:
• Norm mode emulates FET compressors,
which can have faster attack and release
times than electro-optical compressors.
This mode lets you fine-tune compression
precisely by adjusting the attack, release,
and ratio controls.
• Warm mode is based on Norm mode, but
has release characteristics more like those
of electro-optical limiters.
• Opto mode emulates classic electro-optical
limiters, which tend to have gentler attack
and release characteristics than FET compressors. The attack, release and ratio controls are not adjustable in this mode.
• “Key Input” side-chain processing, which lets
you trigger compression using the dynamics of
another signal.
• Side-Chain EQ filter, which lets you tailor the
compression to be frequency-sensitive.
• High Pass filter, which lets you remove
“thumps” or “pops” from your audio.
• Distortion control, which lets you add different types of subtle harmonic distortion to the
output signal.
Smack! has no control to directly adjust the
threshold level (the level that an input signal must exceed to trigger compression).
The amount of compression will vary with
the input signal, which is adjustable by the
Input control.
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Smack! Controls and Meters
Smack! includes controls for multiple compression modes and a VU meter.
Smack! Compression Mode
Buttons
Smack! has three modes of compression: Norm
(Normal), Opto, and Warm. Use the corresponding button to select a mode.
Norm, Warm, and Opto mode buttons
Norm Mode Button
Enable the Norm button to emulate FET compressors, which can have significantly faster attack and release times than opto-electricalbased compressors. It can be used for a wide
range of program material and, with extreme
settings, can be used for sound effects such as
“pumping.”
In Norm mode, you can precisely adjust the Ratio, Attack, and Release controls to fine-tune the
compression characteristics.
,
As with Norm mode, Warm mode can be used
for a wide range of program material including
vocals or low-frequency instruments such as
tom-toms or bass guitar. Extreme settings can
be used to produce “pumping” effects. Like
Norm mode, Warm mode lets you precisely adjust the Ratio, Attack, and Release controls to
fine-tune the compression characteristics.
Opto Mode Button
Enable the Opto button to emulate opto-electro
compressors. Opto mode produces “soft knee”
compression with gentle attack and release
characteristics, and is ideal for compressing
thin vocals, bass guitars, kick drums, and snare
drums. In Opto mode, only the Input and Output
controls are available for adjusting the amount
of compression. The Attack, Release, and Ratio
controls are greyed out and cannot be manually
adjusted.
Smack! Input Control
In all Smack! compression modes, Input adjusts
the level of input gain to the compressor. For
more compression, increase the amount of input
gain. For less compression, reduce the amount
of input gain.
Some sustained low-frequency tones can
cause waveform distortion in Norm mode.
The release characteristics of Warm mode
(which is based on Norm mode) can be used
to remedy this distortion by reducing waveform modulation.
Setting the Input and Output controls to 5 is
equal to unity gain at a compression ratio of
1:1.
Warm Mode Button
Enable the Warm button for compression that is
based on Norm mode, but which has programdependent release characteristics. These characteristics, often described as “transparent” or
“smooth,” can be less noticeable to the listener
and can reduce waveform distortion caused by
some sustained low-frequency tones.
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Input
Smack! Attack Control
In Norm and Warm modes, Attack controls the
rate at which gain is reduced after the input signal crosses the threshold.
This control is greyed out in Opto mode.
Set this control to 0 for the fastest attack time,
or to 10 for the slowest attack time. Depending
on the program material and the parameters
used, this represents an approximate range of
100 s to 80 milliseconds.
As you increase the Ratio control, Smack! goes
from applying “soft-knee” compression to
“hard-knee” compression, as follows:
• With soft-knee compression, gentle compression begins and increases gradually as the input signal approaches the threshold. This
creates smoother compression.
• In hard-knee compression, compression begins when the input signal exceeds the threshold. This can sound abrupt, and is ideal for
limiting or de-essing.
Smack! compression ratios range from subtle
compression to hard limiting. At ratios of 10:1
and higher, Smack! functions as a limiter. Selecting the Smack! setting lowers the threshold
slightly and applies hard limiting, which keeps
the output level constant regardless of the input
level. (This setting can also be used for extreme
compression effects.)
Attack
Smack! Ratio Control
In the Norm and Warm modes, Ratio controls
the compression ratio, or the amount of compression applied as the input signal exceeds the
threshold. For example, a 2:1 compression ratio
means that a 2 dB increase of level above the
threshold produces a 1 dB increase in output.
This control is greyed out in Opto mode.
Smack! has no control to directly adjust the
threshold level (the level that an input signal must exceed to trigger compression).
The amount of compression will vary with
the input signal, which is adjustable by the
Input control.
Ratio
Smack! Release Control
In Norm and Warm modes, Release controls the
length of time it takes for the compressor to be
fully deactivated after the input signal drops below the threshold level. If the release time is too
short, distortion can occur on low-frequency
signals.
This control is greyed out in Opto mode.
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Set this control to 0 for the fastest release time,
or to 10 for the slowest release time. Depending
on the program material and the parameters
used, this represents an approximate range of
15 ms to 1 second for Norm mode (or the primary release of Warm mode).
Smack! Side-Chain EQ Filter
The side-chain is the signal path that a compressor uses to determine the amount of gain reduction it applies to the signal being compressed.
This signal path is derived from the input signal
or Key Input, depending on the user's selection.
When enabled, the Side-Chain EQ filter lets the
user tailor the equalization of the side-chain signal so that the compression becomes frequencysensitive.
Release
Smack! Output Control
In all Smack! compression modes, Output adjusts the overall output gain, which lets you
compensate for heavily compressed signals by
making up the resulting difference in gain.
When you apply Smack! to stereo or multichannel tracks, the Output control determines master output levels for all channels.
See “Using the Smack! Side-Chain Input” on
page 109 for more information on using the
Side-Chain EQ on a Key Input.
The Side-Chain EQ filter has the following settings:
High Pass Makes the compressor's detector less
sensitive to low frequencies in the input signal
or Key Input by rolling off at a rate of 6 dB per
octave. For example, you might use this setting
on a mix to prevent a bass guitar or bass drum
from causing too much gain reduction.
Set this control to 0 for no output gain (silence),
or to 10 for the loudest output gain. This represents an approximate range of +40 dB.
Setting the Input and Output controls to 5 is
equal to unity gain at a compression ratio of
1:1.
High Pass Side-Chain EQ
Output
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Band-Emphasis Makes the compressor's detec-
tor more sensitive to mid-to high frequencies in
the input signal or Key Input by boosting those
frequencies in the side-chain signal. For example, you might use this setting to reduce sibilance in vocal tracks.
The amount of distortion that Smack! applies to
the input signal depends on both the level of the
input signal and the amount of compression being applied.
Odd Applies mostly odd (and some even) harmonics to the distortion.
Even Applies mostly even (and some odd) har-
monics to the distortion.
O+E Applies an equal blend of odd and even har-
monic distortion.
The Output control has no effect on the level
of distortion applied to the signal.
Band-Emphasis Side-Chain EQ
Combined Enables the High Pass and peak settings simultaneously to make the compressor's
detector more sensitive to high frequencies and
less sensitive to low frequencies.
Distortion
Smack! HPF Toggle Switch
When enabled, the HPF (high pass filter) toggle
switch gently rolls off audio frequencies lower
than 60 Hz in the output signal at a rate of 6 dB
per octave.
Combined Side-Chain EQ
Off Disables the Side-Chain EQ control.
This is especially useful for removing “thumps”
or “pops” from vocals, bass, or kick-drums.
Smack! Distortion Control
When enabled, Distortion adds subtle secondorder and third-order harmonic distortion to
the output signal.
HPF Toggle Switch
• Odd harmonics produce waveforms that are
more square-shaped and are often described as “harsh” sounding.
• Even harmonics produce waveforms with
more rounded edges and are often described as “smooth” sounding.
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Smack! VU Meter
The VU meter displays the amount of input
level, output level, or gain reduction from compression, depending on the current Meter Mode
button setting. It is calibrated to a reference
level of –14 dBFS = 0 VU.
Input
Clipping
indicator
Internal
Clipping
indicator
Meter Mode
button
Output
Clipping
indicator
The Internal Clipping indicator (labelled “INT
CLIP”) turns red when the signal exceeds the
available headroom. Clicking the Internal Clipping indicator clears it. Alt-clicking (Windows)
or Option-clicking (Mac) clears the clip indicators on all channels.
Using the Smack!
Compressor/Limiter
Smack! supports 44.1 kHz, 48 kHz, 88.2 kHz,
96 kHz, 176.4 kHz and 192 kHz sample rates. It
works with mono, stereo, and greater-than-stereo multichannel formats up to 7.1.
Input meter
Gain meter
Output
meter
VU Meter
Meter Mode Button and Clip Indicators
The Meter Mode button toggles between displaying three display modes, as follows:
In Displays the input signal level, referenced to
–14 dBFS = 0 VU.
Out Displays the output signal gain, referenced
to –14 dBFS = 0 VU.
GR Displays the amount of gain reduction ap-
plied by the compressor.
Input and Output Meters
The Input and Output meters indicate input and
output signal levels in dBFS (dB relative to full
scale or maximum output).
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Sample rates of 176.4 and 192 kHz with the
TDM version of Smack! require an HD Accel
card, and only work with mono, stereo, and
greater-than-stereo multichannel formats
up to 7.0. These higher sample rates are not
supported by HD Core ™ and HD Process ™
cards
In general, when working with stereo and
greater-than-stereo tracks, use the multichannel
version of Smack!.
Multi-mono plug-ins, such as dynamicsbased or reverb plug-ins, may not function
as you expect. Use the multichannel version
of a multi-mono plug-in when available.
The TDM version of Smack! introduces 5 samples of delay. The RTAS version of Smack! introduces 1 sample of delay.
Using the Smack! Side-Chain
Input
Smack provides side-chain processing capabilities. Compressors typically use the detected amplitude of their input signal to cause gain reduction. This split-off signal is called the side-chain.
However, an external signal (referred to as the
Key Input) can be used to trigger compression.
A typical use for external side-chain processing
is to control the dynamics of one audio signal
using the dynamics of another signal. For example, you could use a lead vocal track to duck the
level of a background vocal track so that the
background vocals do not interfere with the lead
vocals.
RTAS plug-ins do not provide side-chain
processing when used on TDM-based systems. If you want to use side-chain processing, use the TDM versions of plug-ins on
TDM-based systems.
To use an external Key Input to trigger
compression:
1 Insert Smack! on a track you want to compress
using external side-chain processing.
On the audio track or Auxiliary Input that you
want to specify as the Key Input (the signal that
will be used to trigger compression), click the
Send button and select the bus path to the track
that will use side-chain processing.
2
The Key Input must be monophonic.
In the track that you are compressing, click the
instance of Smack! in the Inserts pop-up menu.
3
In the Smack! plug-in window, click the Key
Input menu, and select the input or bus path that
you have designated as the Key Input.
4
Begin playback. Smack! uses the input or bus
that you selected as a Key Input to trigger its effect.
5
To fine-tune the amount of compression, adjust the send level from the Key Input track.
6
The Side-Chain EQ filter lets you tailor the
equalization of the side-chain signal so that
the compression becomes frequency-sensitive. See “Smack! Side-Chain EQ Filter” on
page 106 for more information.
When you are using a Key Input to trigger
compression, the Input control has no effect
on the amount of compression.
To tailor the side-chain signal so that the detector is frequency-sensitive, use the Side-Chain
EQ filter (see “Smack! Side-Chain EQ Filter” on
page 106 for more information).
7
Adjust other parameters to achieve the desired
effect.
8
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Audio Plug-Ins Guide
Chapter 19: TL Aggro
TL Aggro plug-in
TL Aggro is a dynamics processing plug-in that
is modeled on vintage FET compressors and is
available in TDM and RTAS formats. At moderate settings, TL Aggro is designed to sound
smooth and transparent, perfect for vocals and
acoustic instruments. Crank TL Aggro up for
maximum aggressiveness and it instantly adds
character and intensity to guitars and drum
tracks.
TL Aggro Overview
This sections explains the basics of analog compression, and how the TL Aggro works.
Analog Compression
Compression is a common audio processing
technique that is essential to many recording
styles. A compressor is a specialized type of amplifier that acts to reduce the dynamic range between the quietest and loudest peaks of an audio
signal. When dynamic range is compressed, this
highlights quieter parts of an audio signal while
taming the loudest parts. Heavy use of compression on percussion, instruments, and vocals is a
staple in musical genres such as rock and pop.
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Before the introduction of digital technology in
the studio, compressors were typically designed
around a set of analog components. Various
compressor circuit designs are known for their
distinctive sound and characteristics. Popular
analog compressors are often designed around
optical isolator, VCA (voltage controlled amplifier), or FET (field effect transistor) based circuits that produce the compression effect.
TL Aggro
TL Aggro implements a unique compressor topology based on a traditional analog FET design,
with several updates for the digital age.
The following figure shows the different modules of TL Aggro and how they interact with the
audio signal.
TL Aggro signal flow, processing, and controls
TL Aggro uses a reverse feedback system common to many analog compressors. In essence,
this means that the compressor is not compressing the input signal but rather analyzing and
compressing the already compressed output signal. Sound weird? It is. Reverse-feedback is a
strange and paradoxical concept. It can lead to
strange and chaotic behavior if not well-tamed.
In fact, at least one well known and popular
hardware compressor that uses a reverse feedBack topology becomes marginally unstable at
extreme compression settings. Despite this
sometimes unpredictable behavior, the reverse
feedBack model produces a desirable and unique
compression sound.
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Audio Plug-Ins Guide
TL Aggro adds modern digital conveniences to
the reverse feedBack model. Precise bass compensation provides for improved tracking of
bass heavy instruments or a complete stereo
mix. TL Aggro provides linked stereo operation
to preserve stereo imaging as well as full sidechain support. A tube drive module adds additional tube-style distortion if desired.
TL Aggro uses a program dependent release
which provides more natural sounding compression. In essence, the program dependent release works to slow down the release time of
compressor so that it more smoothly rides the
average loudness of the audio material.
The most unique feature of TL Aggro is its
Threshold control. Most reverse-feedback compressors do not implement a Threshold control
typical to non-FET compressors. Instead, they
provide an input control that increases the
amount of compression as the unit is driven
harder. However, an input control adjustment is
often less intuitive than a Threshold control.
Implementing a Threshold control into the operation of TL Aggro has two specific side-effects. At the extreme setting of a high threshold,
high ratio, fast attack, and a slow release,
TL Aggro can overshoot in compression and become “sticky” with a high gain reduction. Sonically, this sounds like “pops” in the output signal. In more technical terms, TL Aggro is
becoming marginally unstable. In this scenario
you can alleviate the problem by doing one or
more of the following:
• Lower the Threshold
• Reduce the Ratio
• Reduce the Attack
• Increase the Release
The second side effect is that for a given set of
Ratio and Attack settings, the compressor has a
finite range of available gain reduction. At some
cutoff point on the Threshold knob, you might
find that compressor ceases to apply anymore
compression to the signal. To acquire more
compression range, increase the Ratio slider, or
alternatively increase the Attack speed.
The reverse-feedback model combined with the
Threshold control and additional features like
Bass Compensation and Tube Drive gives
TL Aggro a wide range of compression styles
once you understand how it operates. The ability to adjust threshold gives TL Aggro a distinctive advantage over traditional reverse feedBack
designs, both in terms of functionality and sonic
character.
TL Aggro Controls
TL Aggro controls are grouped together in the
plug-in interface as follows: compression controls, bass compensation controls, tube drive
controls, and meters.
TL Aggro Compression Controls
TL Aggro provides the standard compression
controls Threshold, Ration, Attack, Release, and
Post Gain.
Compressions controls
Threshold
The Threshold control sets the amplitude level
at which the compressor begins to affect the input signal. The values indicated on the Threshold knob are in negative dB. At the default 0 dB
setting, TL Aggro will pass the audio signal
through at unity gain and will have no effect on
the audio. As the Threshold knob is turned
clockwise (click and drag up), the threshold will
be lowered deeper into the input signal and result in more gain reduction as the compressor
becomes sensitive to more of the incoming audio signal.
Ratio
The Ratio control indicates the degree at which
TL Aggro is reducing dynamic range. The Ratio
slider increases the amount of compression as
the slider is pushed upwards, by increasing the
amount of gain reduction in the output signal
relative to the input signal. Additionally, as the
ratio is increased, the “knee” of compression
curve is made tighter. At lower ratio settings,
TL Aggro has a gentle knee in the compression
curve.
Attack and Release
The Attack control controls the amount of time
it takes TL Aggro to begin compression once the
audio signal has reached the threshold. Slow attack times tend to promote overall brightness
and high frequency audio within the compressed audio signal.
Conversely, the Release control controls the
time it takes TL Aggro to return to unity gain
once the audio signal has fallen back below the
threshold. TL Aggro uses a program dependent
release which slows down the release time to
more smoothly ride the average loudness of the
audio material.
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Turning the Attack and Release knobs clockwise
increases the reaction speed of the compressor.
1 is the slowest setting and 10 is the fastest setting.
breathing. For example, Bass Compensation
sounds great on bass guitar or when you have
TL Aggro on your master fader as stereo bus
compressor.
Post Gain
Additionally, TL Aggro provides a cutoff frequency control to tailor the sound of the bass
compensation. This acts as a high pass filter and
the values indicated above the Bass Compensation slider are in Hertz. As the slider increases
from left to right, the compressor will be even
less reactive to low frequencies.
The Post Gain control lets you make up for the
signal gain lost through compression. The values indicated on the knob are in dB. At maximum setting, 36 dB of gain can be applied to the
compressed signal.
TL Aggro Bass Compensation
Controls
The Bass Compensation section of TL Aggro affects the compressor’s side-chain circuitry. By
default, Bass Compensation is enabled as indicated by the illuminated green light. To disable,
toggle the switch in the section by clicking it.
The green lamp will turn off to indicate that
Bass Compensation has been switched out of the
side-chain signal path.
For example, place a stereo TL Aggro on a full
stereo drum mix. Set the compressor for moderate to high gain reduction levels, enable the Bass
Compensation, and slide the frequency control
from left to right. As the cutoff frequency is increased, you will hear more and more of the kick
drum “punch” through the mix and become
louder relative to snare or cymbals.
TL Aggro Tube Drive Control
The Tube Drive module adds subtle even order
distortion after the compression processing,
simulating the effect of a vacuum tube amplifier.
This provides a difference in the sonic signature
of TL Aggro and is most noticeable on audio
with harmonic content such as piano and acoustic guitar.
Bass Compensation controls
When Bass Compensation is enabled, the compressor becomes less sensitive to bass frequencies in the input signal. This models the sensitivity of the human ear, which is also much less
sensitive to low frequencies. For most signal
sources, enabling Bass Compensation will reduce the total amount of gain reduction that
TL Aggro induces, but the result will often be
more natural sounding with less pumping and
Tube Drive control
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Audio Plug-Ins Guide
To engage the Tube Drive, turn the Tube Drive
rocker switch to on by clicking it. The Tube
Drive rocker switch and tube light up when Tube
Drive processing is on. The amount of distortion
increases with the output level.
TL Aggro Meters
TL Aggro provides LED and Needle meters
LED Meters, In and Out
LED Meters
The LED meters display the peak input and output levels. The LED meters are normalized to
0 dB at digital full-scale.
Note that when TL Aggro is inserted on a mono
track, only the left LED meters will display levels.
Needle Meter
The Needle meter shows input, output, and gain
reduction levels, selectable by the buttons directly to the left of the meter. By default, the GR
(gain reduction) button is selected and the meter displays the amount of gain reduction
TL Aggro is applying on the input.
ited to give it more natural motion. At fast release settings, the instantaneous gain reduction
might be less than what it is presented by the
needle.
In Input (IN) or Output (OUT) mode, the needle
meter displays an average of the signals roughly
approximating the RMS (root-mean-square)
strength of the signal. The grey scale on the meter represents the input and output levels in
negative dB This gives you a better representation of the overall loudness of the signal with respect to the LED meters.
Using the TL Aggro
Side-Chain Input
Using a Side-Chain Input to TL Aggro lets you
direct audio from another track or hardware input in your Pro Tools session to drive the input
of the TL Aggro compressor. This is usually
achieved by sending the audio from the desired
channel to a bus and setting the side-chain input
on TL Aggro to the same bus.
On versions of Pro Tools prior to 7.0, RTAS
plug-ins do not provide side-chain processing on TDM systems. Use the TDM version
of TL Aggro if you require side-chain processing on a TDM system.
For more information on using Side-Chain
Input, see the Pro Tools Reference Guide.
When in GR mode, the needle instantaneously
reacts to peak reductions that occur. The red
scale of the meter indicates compression in dB.
This gives you an accurate representation of the
total amount of gain reduction being applied.
However, the release speed of the needle is lim-
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Audio Plug-Ins Guide
Part IV: Pitch Shift Plug-Ins
Chapter 20: AIR Frequency Shifter
AIR Frequency Shifter is an RTAS pitch-shifting
plug-in.
Use the Frequency Shifter plug-in to shift the
audio signal’s individual frequencies inharmonically, creating a unique effect.
Shifter Section
The Shifter section provides control over the direction of frequency shift, and feedback of the
signal through the algorithm.
Mode The Mode control sets the direction of the
frequency shifting effect.
Up Shifts frequencies up.
Down Shifts frequencies down.
Up & Down Shifts frequencies equally up and
Frequency Shifter Plug-In window
down, and the two shifted signals are heard simultaneously.
Frequency Shifter Controls
Stereo Shifts the right channel frequencies up,
and the left channel down.
The Dynamic Delay plug-in provides a variety of
controls for adjusting plug-in parameters.
Feedback
Frequency
The Feedback control lets you run the signal
through the pitch shifting algorithm multiple
times, creating a cascading, layered effect.
The Frequency control sets the amount of frequency shifting.
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet (pitchshifted) signal. At 50%, there are equal amounts
of dry and wet signal. At 0%, the output is all dry
and at 100% it is all wet.
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Audio Plug-Ins Guide
Chapter 21: Pitch
Pitch is a pitch-shifting plug-in that is available
in TDM and AudioSuite formats.
The Pitch plug-in is designed for a variety of audio production applications ranging from pitch
correction of musical material to sound design.
Pitch processing uses the technique of varying
sample playback rate to achieve pitch transposition. Because changing audio sample playback
rate results in the digital equivalent of varispeeding with tape, this is an unsatisfactory
method since it changes the overall duration of
the material.
Pitch transposition with the Pitch plug-in involves a much more complex technique: digitally adding or subtracting portions of the audio
waveform itself, while using de-glitching crossfades to minimize undesirable artifacts. The result is a processed signal that is transposed in
pitch, but still retains the same overall length as
the original, unprocessed signal.
The Pitch plug-in was formerly called
DPP-1. It is fully compatible with all settings and presets created for DPP-1.
Pitch plug-in
Pitch Controls
The Pitch plug-in provides the following controls:
Input Level This control attenuates the input
level of the Pitch plug-in to help prevent internal clipping.
Signal Present Indicator LED This LED indicates
the presence of an input signal.
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121
Clip Indicator This indicator indicates whether
clipping has occurred on output. It is a clip-hold
indicator. If clipping occurs at any time, the clip
light will remain on. To clear the Clip indicator,
click it. Long delay times and high feedback
times increase the likelihood of clipping.
–8va and +8va Buttons Clicking the –8va button
adjusts pitch down one octave from the current
setting of the coarse and fine pitch controls.
Clicking the +8va button adjusts pitch up one
octave from the current setting of the coarse and
fine pitch controls.
Mix This control adjusts the ratio of dry signal to
effected signal in the output. In general, this
control should be set to 100% wet, unless you are
using the Pitch plug-in in-line on an Insert for
an individual track or element in a mix. This
control can be adjusted over its entire range
with little or no change in output level.
Delay This control sets the delay time between
the original signal and the pitch-shifted signal.
It has a maximum setting of 125 milliseconds.
You can use the Delay control in conjunction
with the Feedback control to generate a single
pitch-shifted echo, or a series of echoes that
climb in pitch.
Relative Pitch Entry (Musical
Staff)
Clicking on any note on this musical staff selects
a relative pitch transposition value that will be
applied to an audio signal. If the C above middle
C is illuminated (the staff is in treble clef), it indicates that no pitch transposition has been selected. If a pitch transposition is selected, the
note interval corresponding to the selected
transposition value is indicated in yellow. Altclicking (Windows) or Option-clicking (Mac) on
the staff will set the coarse pitch change value to
zero.
Feedback This control controls the amount and
type of feedback (positive or negative) applied
from the output of the delay portion of the Pitch
plug-in back into its input. It also controls the
number of repetitions of the delayed signal. You
can use it to produce effects that spiral up or
down in pitch, with each successive echo shifted
in pitch.
Coarse This control adjusts the pitch of a signal
in semitones over a two octave range. Pitch
changes are indicated both in the Semitones
field and in the Musical Staff section below this
slider. Using the –8va and +8va buttons in conjunction with the Coarse slider provides a full 4octave range of adjustment.
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Audio Plug-Ins Guide
Relative pitch entry
Fine This control controls the pitch of a signal in
cents (hundredths of a semitone) over a 100 cent
range. The range of this slider is –49 to +50
cents. Precise pitch change values are indicated
in the Fine field. The flat, natural, and sharp
signs below this slider indicate deviation from
the nearest semitone.
Ratio This control indicates the ratio of transposition between the original pitch and the selected transposition value.
Crossfade This control adjusts the crossfade
length in milliseconds to optimize performance
of the Pitch plug-in according to the type of audio material you are processing. The Pitch plugin performs pitch transposition by replicating
or subtracting portions of audio material and
very quickly crossfading between these alterations in the waveform of the audio material.
Crossfade length affects the amount of smoothing performed on audio material to prevent audio artifacts such as clicks from occurring as the
audio is looped to generate the pitch shift.
In general, small, narrow-range pitch shifts require longer crossfades and large shifts require
smaller ones. The disadvantage of a long crossfade time is that it will smooth the signal, including any transients. While this is sometimes
desirable for audio material such as vocals, it is
not appropriate for material with sharp transients such as drums or percussion.
The default setting for this control is Auto. At
this setting, crossfade times are set automatically, according to the settings of the Coarse and
Fine pitch controls. The Auto setting is appropriate for most applications. However, you can
manually adjust and optimize crossfade times
using the Crossfade slider if necessary. For audio material with sharper attack transients, use
shorter crossfade times. For audio material with
softer attack transients, use longer crossfade
times.
Minimum Pitch This controls sets the minimum
fundamental pitch that the Pitch plug-in will
recognize when performing pitch transposition.
Use this to optimize the Pitch plug-in’s performance by adjusting this control based on the
lowest fundamental pitch of the audio material
that you want to process.
On audio material with a low fundamental pitch
frequency content (such as an electric bass) setting this control to a lower frequency (such as
30 Hz) will improve the Pitch plug-in’s performance. The most important thing to remember
when using this control is that the fundamental
frequency of audio material you want to process
must be above the frequency you set here.
The range of this slider is from 15 Hz to 1000 Hz.
The default setting is 60 Hz. Adjustment is tied
to the current setting of the Maximum Pitch
control so that the minimum range is never less
than one octave, and the maximum range never
more than five octaves.
Maximum Pitch This control adjusts the maxi-
mum fundamental pitch that the Pitch plug-in
will recognize when performing pitch transposition. To optimize the Pitch plug-in’s performance, adjust this setting (and the Minimum
Pitch setting) based on the highest fundamental
pitch of the audio material that you want to process. The range of this slider is from 30 Hz to
4000 Hz. The default setting is 240 Hz.
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Chapter 22: Pitch Shift
Pitch Shift is an AudioSuite plug-in that provides pitch-based processing.
The Pitch Shift plug-in adjusts the pitch of any
source audio file with or without a change in its
duration. This is a very powerful function that
transposes audio a full octave up or down in
pitch with or without altering playback speed.
Pitch Shift Controls
This Pitch Shift plug-in provides the following
controls:
Gain Adjusts input level, in 10ths of a dB. Dragging the slider to the right increases gain, dragging to the left decreases gain.
Coarse and Fine Adjusts amount of pitch shift.
The Coarse slider transposes in semitones (half
steps). The Fine slider transposes in cents (hundredths of a semitone).
Ratio Adjusts the amount of transposition
(pitch change). Moving the slider to the right
raises the pitch of the processed file, while moving the slider to the left decreases its pitch.
Crossfade Use this to manually adjust crossfade
Pitch Shift plug-in
length in milliseconds to optimize performance
of the Pitch Shift plug-in according to the type
of audio material you are processing. This plugin achieves pitch transposition by processing
very small portions of the selected audio material and very quickly crossfading between these
alterations in the waveform of the audio material.
Crossfade length affects the amount of smoothing performed on audio material. This prevents
audio artifacts such as clicks from occurring. In
general, smaller pitch transpositions require
longer crossfades; wider pitch transpositions re-
Chapter 22: Pitch Shift
125
quire smaller crossfades. Long crossfade times
may over-smooth a signal and its transients.
This is may not be desirable on drums and other
material with sharp transients.
Use the Crossfade slider to adjust and optimize
crossfade times. For audio material with sharper
attack transients, use smaller crossfade times.
For audio material with softer attack transients,
use longer crossfade times.
Min Pitch Sets the lowest pitch used in the plugin’s Pitch Shift processing. The control has a
range of 40 Hz to 1000 Hz. Use it to focus the
Pitch Shift process according to the audio’s
spectral shape.
Use lower values when processing lower frequency audio material. Use higher values when
processing higher frequency audio material.
Accuracy Sets the processing resources allocated to audio quality (Sound) or timing
(Rhythm). Set the slider toward Sound for better
audio quality and fewer audio artifacts. Set the
slider toward Rhythm for a more consistent
tempo.
Time Correction Disabling this option has the ef-
fect of “permanently varispeeding” your audio
file. The file’s duration will be compressed or
extended according to the settings of the Coarse
and Fine pitch controls. When Time Correction
is enabled, fidelity can be affected. For example,
time expansion as a result of Time Correction
when lowering pitch can cause the audio to
sound granulated.
Pitch Shift Reference Pitch
Controls
The following controls let you use a reference
pitch as an audible reference when pitch-shifting audio material.
Reference Pitch Activates the sine wave-based
reference tone.
Note Adjusts the frequency of the reference tone
in semitones (half steps).
Detune Provides finer adjustment of the fre-
quency of the reference tone in cents (100ths of
a semitone).
Level Adjusts the volume of the reference tone
in dB.
Using Reference Pitch
To use Reference Pitch:
Select the audio material you want to use as a
pitch reference. Click the preview button to begin playback of the selected audio.
1
Click the Reference Pitch button to activate the
reference sine wave tone.
2
Adjust the Note and Detune settings to match
the reference tone to the pitch of the audio playback. Adjust the Gain setting to change the relative volume of the reference tone. It may also be
helpful to toggle the Reference Pitch on and off
to compare pitch.
3
4
Select the audio material to be pitch shifted.
Adjust the Coarse and Fine Pitch Shift controls
to match the pitch of the audio playback to the
reference pitch.
5
Click Render to apply pitch shift to the selection.
6
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Audio Plug-Ins Guide
Chapter 23: Time Shift
Time Shift is an AudioSuite plug-in that provides high quality time compression and expansion (TCE) algorithms and formant correct
pitch-shifting.
Time Shift Controls
Time Shift is ideal for music production, sound
design, and post production applications. Use it
to manipulate audio loops for tempo matching
or to transpose vocal tracks using formant correct pitch shifting. You can also use it in audio
post production for pull up and pull down conversions as well as for adjusting audio to specific
time or SMPTE durations for synchronization
purposes.
Audio Use the controls in the Audio section to
Time Shift controls in the interface for are organized in the following four sections:
select the most appropriate time compression
and expansion algorithm (mode) for the type of
material you want to process, and to attenuate
the gain of the processed audio to aid clipping.
Time Use the controls in the Time section to
specify the amount of time compression or expansion you want to apply.
Formant or Transient Use the controls in the
Formant or Transient section to adjust either
the amount of formant shift or the transient detection parameters depending upon which mode
you have selected in the Audio section. The Formant section is only available when Monophonic is selected as the Audio Mode. The Transient section is available with slightly different
controls depending on whether Polyphonic or
Rhythmic is selected as the Audio Mode.
Pitch Use the controls in the Pitch section to apply pitch shifting. Pitch shifting can also be formant correct if you select the Monophonic audio
setting.
Time Shift plug-in
Chapter 23: Time Shift
127
Time Shift Audio Controls
The Audio section of Time Shift provides controls for specifying the type of audio you want to
process and gain attenuation of the processed
signal to avoid clipping.
Time Shift plug-in, Audio section
Mode
The Audio Mode pop-up menu determines the
following types of TCE and pitch shift algorithm
for processing audio:
Monophonic Select Monophonic for processing
monophonic sounds (such as a vocal melody).
Polyphonic Select Polyphonic for processing
complex sounds (such as a multipart musical selection).
Rhythmic Select Rhythmic for processing per-
cussive sounds (such as a mix or drum loop).
Rhythmic mode uses transient analysis for
time shifting. If you select audio with no apparent transients, or set the Transient
Threshold control to a setting above any detected transients, Time Shift assumes a “virtual-transient” every three seconds to be
able to process the file. Consequently, the
file should be 20 bpm or higher (one beat every three seconds) to achieve desirable results. For material that has no apparent
transients, use Monophonic or Polyphonic
mode.
Varispeed Select Varispeed to link time and
pitch change for tape-like pitch and speed
change effects, and post production workflows.
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Audio Plug-Ins Guide
Range
The Audio Range pop-up menu determines the
following frequency ranges for analysis:
Low For low-range material, such as a bass guitar, select Low.
Mid For mid-range material, such as male vocals, select Mid. In Monophonic mode, Mid is
the default setting and is usually matches the
range of most monophonic material.
High For material with a high fundamental frequency such as female vocals, select high.
Wide For more complex material that covers a
broad frequency spectrum, select Wide. In Polyphonic mode, Wide is the default setting and is
usually best for all material when using the
Polyphonic audio type.
The range pop-up menu is unavailable in
Rhythmic mode and Varispeed mode.
Gain
The Audio Gain control attenuates the input
level to avoid clipping. Adjust the Gain control
from 0.0 dB to –6.0 dB to avoid clipping in the
processed signal.
Clip Indicator
The Clip indicator indicates clipping in the processed signal. When using time compression or
pitch shifting above the original pitch, it is possible for clipping to occur. The Clip indicator
lights when the processed signal is clipping. If
the processed signal clips, undo the AudioSuite
process and attenuate the input gain using the
Gain control. Then, re-process the selection.
Level Indicator
The Level indicator displays the level of the output signal using a plasma LED, which uses the
full range of plasma level metering colors.
Time Shift Time Controls
The Time section of Time Shift provides controls for specifying the amount of time compression or expansion as well as the timebase used
for calculating TCE. Adjust the Time control to
change the target duration for the processed audio.
Time Shift plug-in, Time section
Original Displays the Start and End times, and
Length of the edit selection. Times are displayed
in units of the timebase selected in the Units
pop-up menu.
Processed Displays the target End time and
Length of the processed signal. Times are displayed in units of the timebase selected in the
Units pop-up menu. You can click the Processed
End and Length fields to type the desired values.
These values update automatically when adjusting the Time control.
Tempo Displays the Original Tempo and Processed Tempo in beats per minute (bpm). You
can click the Original Tempo and Processed
Tempo fields to type the desired values. The
Processed Tempo value updates automatically
when adjusting the Time control.
Unit Select the desired timebase for the Original
and Processed time fields: Bars|Beats, Min:Sec,
Timecode, Feet+Frames, or Samples.
Time Shift does not receive Bars|Beat and
Feet+Frame information from Pro Tools 7.0
or 7.1. Consequently, Bars|Beats and
Feet+Frames are displayed as “N/A.”
Speed Displays the target time compression or
expansion as a percentage of the original. Adjust
the Time control or click the Speed field and
type the desired value. Time can be changed
from 25.00% to 400.00% of the original speed
(or 4 to 1/4 times the original duration). The default setting is 100.00%, or no change. 25.00%
results in 4 times the original duration and
400.00% results in 1/4 of the original duration.
The Speed field only displays up to 2 decimal
places, but lets you type in as many decimal
places as you want (up to the IEEE standard).
While the display rounds to 2 decimal places,
the actual time shift is applied based on the
number you typed. This is especially useful for
typing post production pull up and pull down
factors (see “Post Production Pull Up and Pull
Down Tasks with Time Shift” on page 134).
Time Shift Formant Controls
The Formant section of Time Shift lets you shift
the formant shape of the selected audio independently of the fundamental frequency. This is
useful for achieving formant correct pitch shifting. It can also be used as an effect. For example,
you can formant shift a male vocal up by five
semitones and it will take on the characteristics
of a female voice.
Chapter 23: Time Shift
129
The Formant section is only available when
Monophonic is selected as the Audio Type. The
Formant section provides a single control for
transposing the formants of the selected audio
by –24.00 semitones (–2 octaves) to +24.00
semitones (+2 octaves), with fine resolution in
cents. Adjust the Formant Shift control or click
the Shift field and type the desired value.
Time Shift Transient Controls
The Transient section is only available when
Polyphonic or Rhythmic is selected as the Audio
Type, and provides slightly different controls
for each.
When Polyphonic is selected as the Audio Type,
the Transient section provides controls for setting the transient detection threshold and for
adjusting the analysis window length for processing audio.
Time Shift plug-in, Formant section
Audio with a fundamental pitch has an
overtone series, or set of higher harmonics.
The strength of these higher harmonics creates a formant shape, which is apparent if
viewed using a spectrum analyzer. The overtone series, or harmonics, have the same
spacing related to the pitch and have the
same general shape regardless of what the
fundamental pitch is. It is this formant
shape that gives the audio its overall characteristic sound or timbre. When pitch shifting
audio, the formant shape is shifted with the
rest of the material, which can result in an
unnatural sound. Keeping this shape constant is critical to formant correct pitch
shifting and achieving a natural sounding
result.
Time Shift plug-in, Transient section with Polyphonic
selected as the Audio Type
When Rhythmic is selected as the Audio Type,
the Transient section provides controls for setting the transient detection threshold, and for
adjusting the decay rate of the transients in the
processed audio when time stretching.
Time Shift plug-in, Transient section with Rhythmic
selected as the Audio Type
Follow The follow button enables an envelope
follower that simulates the original acoustics of
the audio being stretched. Click the Follow button to enable or disable envelope following. Follow is only available when Polyphonic is selected as the Audio Type.
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Audio Plug-Ins Guide
Threshold The Threshold controls sets the tran-
sient detection threshold from 0.0 dB to
–40.0 dB. Disable transient detection by setting
the Threshold control to Off (turn the knob all
the way to the right). Part of Time Shift’s processing relies upon separating “transient” parts
of the selection from “non-transient” parts.
Transient material tends to change its content
quickly in time, as opposed to parts of the sound
which are more sustained. Adjust the Threshold
control or click the Threshold field and type the
desired value.
The default value for Threshold is –6.0 dB. For
highly percussive material, lower the threshold
for better transient detection, especially with
the Rhythmic audio setting. For less percussive
material, and for shifting with the Polyphonic
audio setting, a higher setting can yield better
results. Experiment with this control, especially
when shifting drums and percussive tracks, to
achieve the best results.
Decay Rate The Decay Rate control determines
how much of the decay from a transient is heard
in the processed audio when time stretching.
When time stretching using the Rhythmic setting, the resulting gaps between the transients
are filled in with audio, and Decay Rate determines how much of this audio is heard by applying a fade out rate. Decay Rate is only available
when Rhythmic is selected as the Audio Type.
Adjust the Decay Rate up to 100% to hear the audio that is filling the gaps created by the time
stretching with only a slight fade, or adjust
down to 1.0% to completely fade out between the
original transients.
Time Shift Pitch Controls
The Pitch section of Time Shift provides controls for pitch shifting the selected audio. Use
the Pitch control to transpose the pitch from
–24.00 semitones (–2 octaves) to +24.00 semitones (+2 octaves), with fine resolution in cents.
Window The Window control sets the analysis
window length for processing audio. You can set
the Window from 6.0 milliseconds to 185.0 milliseconds. Adjust the Window control or click
the Window field and type the desired value.
The Window control is only available when
Polyphonic is selected as the Audio Type.
The default for Window size is 18.0 milliseconds
and works well for many applications, but you
may want to try different Window settings to get
the best results. Try larger window sizes for low
frequency sounds or sounds that do not have
many transients. Try smaller window sizes for
drums and percussion. 37.0 milliseconds tends
to work well for polyphonic instruments such as
piano or guitar. A setting as large as 71.0 milliseconds works well for bass guitar. Settings in
the 12 millisecond range work well on drums or
percussion.
Time Shift plug-in, Pitch section
Transpose Displays the transposition amount in
semitones. You can transpose pitch from –24.00
semitones (–2 octaves) to +24.00 semitones (+2
octaves), with fine resolution in cents. Adjust
the Pitch control or click the Transpose field
and type the desired value.
Shift Displays the pitch shift amount as a per-
centage. You can pitch shift from 25.00% (–2 octaves) to +400.00% (+2 octaves). Adjust the
Pitch control or click the Shift field and type the
desired value. The default value is 100% (no
pitch shift).
Chapter 23: Time Shift
131
AudioSuite Input Modes and
Time Shift
Time Shift as AudioSuite TCE
Plug-In Preference
Time Shift supports the Pro Tools
AudioSuite Input Mode selector for use on
mono or multi-input processing.
The Time Shift plug-in’s high quality time compression and expansion algorithms that can be
used with the Pro Tools TCE Trim tool.
Mono Mode Processes each audio clip as a mono
file with no phase coherency maintained with
any other simultaneously selected clips.
Multi-Input Mode Processes up to 48 input chan-
nels and maintains phase coherency within
those selected channels.
TCE Plug-In option in Processing Preferences page
Time Shift is not available with the TCE
Trim tool in Pro Tools 7.0 and 7.1.
AudioSuite Preview and Time
Shift
Time Shift supports Pro Tools AudioSuite Preview and Bypass. For more information on using
AudioSuite Preview and Bypass, see the
Pro Tools Reference Guide.
AudioSuite Preview and Bypass are not
available with Time Shift in Pro Tools 7.0
and 7.1.
Refer to the Pro Tools Reference Guide for
more information about the TCE Trim tool.
To select Time Shift for use with the TCE Trim tool:
1
Choose Setup > Preferences.
2
Click the Processing tab.
3
From the TC/E Plug-In pop-up menu, select
Time Shift.
Select the desired preset setting from the Default Settings pop-up menu.
4
5
Click OK.
Processing Audio Using Time
Shift
Time Shift lets you change the time and pitch of
selected audio independently or concurrently.
Normalizing a selection before using Time
Shift may produce better results.
132
Audio Plug-Ins Guide
Changing the Time Using Time
Shift
4 If transposing the pitch of the selection up, attenuate the Gain control as necessary.
To change the time of a selected audio clip:
5
1
Select AudioSuite > Pitch Shift > Time Shift.
Select the Audio Mode appropriate to the type
of material you are processing (Monophonic,
Polyphonic, or Rhythmic).
2
In Monophonic or Polyphonic mode, select the
appropriate Range for the selected material
(Low, Mid, High, or Wide).
3
If compressing the duration of the selection,
attenuate the Gain control as necessary.
4
If using Monophonic mode, adjust the Formant Shift control as desired.
5
If using Polyphonic or Rhythmic mode, adjust
the Transient controls as desired.
6
Make sure Pitch Shift is set to 100% (unless
you also want to change the pitch of the selection).
7
Adjust the Time Shift control to the desired
amount of time change. Time change is measured in terms of the target duration using the
selected timebase or as a percentage of the original.
8
9
Click Render.
If using Monophonic mode, adjust the Formant Shift control as desired.
If using Polyphonic or Rhythmic mode, adjust
the Transient controls as desired.
6
Make sure Time Shift is set to 0% (unless you
also want to change the duration of the section).
7
Adjust the Pitch Shift control to the desired
amount of pitch change. Pitch change is measured in semitones (and cents) or as a percentage of the original.
8
9
Changing the Time and Pitch
Using Time Shift
To change the time and pitch of a selected audio
clip:
1
To change the pitch of a selected audio clip:
1
Select AudioSuite > Pitch Shift > Time Shift.
Select AudioSuite > Pitch Shift > Time Shift.
2 Select Varispeed from the Audio Mode pop-up
menu.
Adjust either the Time Shift or Pitch Shift control to match the desired amount of time and
pitch change in terms of a percentage of the original.
3
4
Changing the Pitch Using Time
Shift
Click Render.
Click Render.
Using the Monophonic, Polyphonic, or
Rhythmic modes, you can adjust both the
Time Shift and Pitch Shift controls independently before processing.
Select the Audio Mode appropriate to the type
of material you are processing (Monophonic,
Polyphonic, or Rhythmic).
2
In Monophonic or Polyphonic mode, select the
appropriate Range for the selected material
(Low, Mid, High, or Wide).
3
Chapter 23: Time Shift
133
Post Production Pull Up and Pull Down Tasks with Time Shift
The table below provides information on TCE settings for common post production tasks. Type the
corresponding TCE% (represented to 10 decimal places in the table) in the Time Shift field for the corresponding post production task and the process the selected audio.
134
Desired Pull Up or Pull Down
TCE% (to 10 Decimal Places)
Frames
Pal to Film –4%.tfx
96.0%
25 to 24/30
PAL to NTSC –4.1%.tfx
95.9040959041%
25 to 23.976/29.97
Film to PAL +4.1667%.tfx
+104.1666666667%
24/30 to 25
Film to NTSC –0.1%.tfx
99.9000999001%
24/30 to 23.976/29.97
NTSC to Pal +4.2667%.tfx
+104.2708333333%
23.976/29.97 to 25
NTSC to Film +0.1%.tfx
+100.10%
23.976/29.97 to 24/30
Audio Plug-Ins Guide
Chapter 24: Vari-Fi
Vari-Fi is an AudioSuite plug-in that is part of
the D-Fi suite of plug-ins. Vari-Fi provides a
pitch-change effect similar to a tape deck or record turntable speeding up from or slowing
down to a complete stop. Vari-Fi preserves the
original duration of the audio selection.
Vari-Fi provides a pitch-change effect similar to
a tape deck or record turntable speeding up
from or slowing down to a complete stop. Features include:
• Speed up from a complete stop to normal
speed
• Slow down to a complete stop from normal
speed
Purposely Degrading Audio
Contemporary music styles, especially hip-hop,
make extensive use of retro instruments and
processors such as vintage drum machines, samplers, and analog synthesizers. The low bit-rate
resolutions and analog “grunge” of these devices
are an essential and much-desired part of their
sonic signatures. That is why Avid created D-Fi.
The D-Fi suite of plug-ins combines the best of
these instruments of the past with the flexibility
and reliability of the Pro Tools audio production system. The result is a set of sound design
tools that let you create these retro sounds without the trouble and expense of resampling audio
through 8-bit samplers or processing it through
analog synthesizers.
Vari-Fi Controls
Vari-Fi
Speed Up Speed Up applies a pitch-change effect
to the selected audio, similar to a tape recorder
or record turntable speeding up from a complete
stop. The effect doesn’t change the duration of
the audio selection.
Slow Down Slow Down applies a pitch-change
effect to the selected audio, similar to a tape recorder or record turntable slowing down to a
complete stop. The effect doesn’t change the duration of the audio selection.
Chapter 24: Vari-Fi
135
136
Audio Plug-Ins Guide
Chapter 25: X-Form
The X-Form AudioSuite plug-in that is based on
the Radius ® algorithm from iZotope. X-Form
provides the high quality time compression and
expansion for music production, sound design,
and audio loop applications. Use X-Form to manipulate audio loops for tempo matching or to
change vocal tracks for formant correct pitch
shifting. The X-Form plug-in is useful in audio
post-production for adjusting audio to specific
time or SMPTE durations for synchronization
purposes. X-Form is also ideal for post-production pull up and pull down conversions.
Normalizing a selection before using
X-Form may produce better results.
X-Form Displays and Controls
Overview
The interface for X-Form is organized in four
sections: Audio, Time, Transient, and Pitch.
Audio Use the controls in the Audio section to
select the most appropriate time compression
and expansion algorithm for the type of material
you want to process and to attenuate the gain of
the processed audio to avoid clipping.
Time Use the controls in the Time section to
specify the amount of time compression or expansion you want to apply.
Transient Use the controls in the Transient section to adjust the transient detection parameters
for high quality time compression or expanssion.
Pitch Use the controls in the Pitch section to apply pitch shifting. Pitch shifting can be formant
correct with either the Polyphonic or Monophonic algorithm.
X-Form Audio Section Controls
The Audio section of X-Form provides controls
for specifying the type of audio you want to process and gain attenuation of the processed signal
to avoid clipping.
X-Form plug-in
X-Form plug-in, Audio section
Chapter 25: X-Form
137
X-Form Time Section Controls
Type
The Audio Type determines the type of TCE and
pitch shift algorithm for processing audio: Polyphonic, Monophonic, or Poly (Faster).
Polyphonic Use for processing complex sounds
(such as a multipart musical selection).
The Time section of X-Form provides controls
for specifying the amount of time compression
or expansion as well as the timebase used for
calculating TCE. Adjust the Time control to
change the target duration for the processed audio.
When previewing Polyphonic, Poly (Faster)
is used for faster previewing. However,
when you process the audio selection, the
high-quality Polyphonic setting is used.
Monophonic Use for processing monophonic
sounds (such as a vocal melody).
X-Form plug-in, Time section
Poly (Faster) Use for faster previewing and pro-
cessing, but with slightly reduced audio quality.
Gain
The Gain control attenuates the input level to
avoid clipping. Adjust the Gain control from
0.0 dB to –6.0 dB to avoid clipping in the processed signal.
Clip Indicator The Clip indicator indicates clip-
ping in the processed signal. When using time
compression or pitch shifts above the original
pitch, it is possible for clipping to occur. The
Clip indicator lights when the processed signal
is clipping. If the processed signal clips, undo
the AudioSuite process and attenuate the input
gain using the Gain control. Then, re-process
the selection.
Level Indicator The Level indicator displays the
level of the output signal using a plasma LED,
which uses the full range of plasma level metering colors.
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Audio Plug-Ins Guide
Original
Displays the Start and End times, and Length of
the edit selection. Times are displayed in units
of the timebase selected in the Units pop-up
menu.
Processed
Displays the target End time and Length of the
processed signal. Times are displayed in units of
the timebase selected in the Units pop-up menu.
You can click the Processed End and Length
fields to type the desired values. These values
update automatically when adjusting the Time
control.
Tempo
Displays the Original Tempo and Processed
Tempo in beats per minute (bpm). You can click
the Original Tempo and Processed Tempo fields
to type the desired values. The Processed Tempo
value updates automatically when adjusting the
Time control.
4x Lets you apply Time Shift, Pitch Shift, and
Unit
Select the desired timebase for the Original and
Processed time fields: Bars|Beats, Min:Sec,
Timecode, Feet+Frames, or Samples.
X-Form does not receive Bars|Beat and
Feet+Frame information from Pro Tools 7.0
or 7.1. Consequently, Bars|Beats and
Feet+Frames are displayed as “N/A.”
Shift
Displays the target time compression or expansion as a percentage of the original. Adjust the
Time control or click the Shift field and type the
desired value. Time can be shifted by as much as
12.50% to 800.00% of the original speed (or 8
times to 1/8 of the original duration) depending
on which Range button is enabled (2x, 4x, or 8x).
The default setting is 100%, or no time shift.
The Shift field only displays up to 2 decimal
places, but lets you type in as many decimal
places as you want (up to the IEEE standard).
While the display rounds to 2 decimal places,
the actual time shift is applied based on the
number you typed. This is especially useful for
post-production pull up and pull down factors
(see “Using X-Form for Post Production Pull Up
and Pull Down Tasks” on page 143).
Formant Shift from 25.00% to 400.00% (where
25.00% is 4 times the original duration and
400.00% is 1/4 of the original duration).
8x Lets you apply Time Shift, Pitch Shift, and
Formant Shift from 12.50% to 800.00% (where
12.50% is 8 times the original duration and
800.00% is 1/8 of the original duration).
When changing to a smaller Range setting
(such as switching from 8x to 2x), the Time
Shift and Pitch Shift settings are constrained to the limits of the new, smaller
range. For example, with 8x enabled and
Time Shift set to 500%, switching to 2x
changes the Time Shift value to 200%.
X-Form Transient Section
Controls
The Transient section provides controls for setting the sensitivity for transient detection and
for adjusting the analysis window size.
X-Form plug-in, Transient section
2x, 4x, and 8x Range Buttons
Sensitivity
The 2x, 4x, and 8x Range buttons set the possible
range for the Time Shift, Pitch Shift, and Formant Shift controls.
Controls how X-Form determines and interprets
transients from the original audio. Part of XForm’s processing relies upon separating “transient” parts of the sample from “non-transient”
parts. Transient material tends to change its
content quickly in time, as opposed to parts of
the sound which are more sustained. Sensitivity
is only available when Polyphonic is selected as
the Audio Type.
2x Lets you apply Time Shift, Pitch Shift, and
Formant Shift from 50.00% to 200.00% (where
50.00% is 2 times the original duration and
200.00% is 1/2 of the original duration).
Chapter 25: X-Form
139
For highly percussive material, lower the Sensitivity for better transient detection, especially
with the Rhythmic audio setting. For less percussive material, a higher setting can yield better results. Experiment with this control, especially when shifting drums and percussive
tracks, to achieve the best results.
Window
Sets the analysis window size. You can adjust the
Window from 10.0 milliseconds to 100.0 milliseconds. Adjust the Window control or click the
Window field and type the desired value. Window is only available when Monophonic is selected as the Audio Type.
Try larger window sizes for low frequency
sounds or sounds that do not have many transients. Try smaller window sizes for tuned
drums and percussion. However, the default of
25 milliseconds should work well for most material.
X-Form Pitch Section Controls
The Pitch section provides controls for pitch
shifting the selected audio. Use the Pitch control
to transpose the pitch from as much as –36.00
semitones (–3 octaves) to +36.00 semitones (+3
octaves), with fine resolution in cents, depending on which Range button is enabled (2x, 4x, or
8x). X-Form also lets you transpose the formant
shape independently of the fundamental frequency.
X-Form plug-in, Pitch section
140
Audio Plug-Ins Guide
Transpose
Displays the transposition amount in semitones.
You can transpose pitch by as much as –36.00
semitones (–3 octaves) to +24.00 semitones (+3
octaves), with fine resolution in cents, depending on which Range button is enabled. Adjust
the Pitch control or click the Transpose field
and type the desired value. The default value is
0.00 semitones, or no pitch shift.
Shift
Displays the pitch shift amount as a percentage.
You can shift pitch by as much as 12.50% (–3 octaves) to 800.00% (+3 octaves) depending on
which Range button is enabled (2x, 4x, or 8x).
Adjust the Pitch control or click the Shift field
and type the desired value. The default value is
100%, or no pitch shift.
Formant
Audio with a fundamental pitch has an overtone
series, or set of higher harmonics. The strength
of these higher harmonics creates a formant
shape, which is apparent if viewed using a spectrum analyzer. The overtone series, or harmonics, have the same spacing related to the pitch
and have the same general shape regardless of
what the fundamental pitch is. It is this formant
shape that gives the audio its overall characteristic sound or timbre. When pitch shifting audio, the formant shape is shifted with the rest of
the material, which can result in an unnatural
sound. Keeping this shape constant is critical to
formant correct pitch shifting and achieving a
natural sounding result.
The Pitch section of X-Form lets you pitch shift
the formants of the selected audio independently of the fundamental frequency. This is
useful for achieving formant correct pitch shift-
ing. It can also be used as an effect. For example,
you can formant shift a male vocal up by five
semitones and it will take on the characteristics
of a female voice.
To enable or disable formant shifting:
AudioSuite TCE Plug-In
Preference
The X-Form plug-in’s high quality time compression and expansion algorithms that can be
used with the Pro Tools TCE Trim tool.
 Click the In button. The In button lights when
formant shifting is enabled.
The Formant field displays the amount of formant pitch shifting from –36.00 semitones (–3
octaves) to +36.00 semitones (+3 octaves), with
fine resolution in cents. Adjust the Formant
control or click the Formant field and type the
desired value. The default value is 0.00 semitones, or no formant shift.
TCE Plug-In option in Processing Preferences page
X-Form is not available with the TCE Trim
tool in Pro Tools 7.1.x and lower.
X-Form AudioSuite Input
Modes
When using X-Form for the TCE Trim tool,
the default 2x Range is used for an edit
range of twice to half the duration of the
original audio. If you select a Default Setting that uses either the 4x or 8x Range, the
Time Shift and Pitch Shift setting are constrained to the 2x Range limit of 50% to
200%.
X-Form supports the Pro Tools AudioSuite Input Mode selector for use on mono or multi-input processing.
Mono Mode Processes each audio clip as a mono
file with no phase coherency maintained with
any other simultaneously selected clips.
Multi-Input Mode Processes up to 48 input chan-
Refer to the Pro Tools Reference Guide for
more information about the TCE Trim tool.
nels and maintains phase coherency within
those selected channels.
To select X-Form for use with the TCE Trim tool:
AudioSuite Preview
X-Form supports Pro Tools AudioSuite Preview
and Bypass. For more information on using AudioSuite Preview and Bypass, see the Pro Tools
Reference Guide.
AudioSuite Preview and Bypass are not
available with X-Form in Pro Tools 7.0
and 7.1.
1
Choose Setup > Preferences.
2
Click the Processing tab.
From the TC/E Plug-In pop-up menu, select
Digidesign X-Form.
3
Select the desired preset setting from the Default Settings pop-up menu.
4
5
Click OK.
Chapter 25: X-Form
141
Processing Audio Using
X-Form
X-Form lets you change the time and pitch of selected audio independently or concurrently.
You can adjust both the Time Shift and
Pitch Shift controls independently before
processing.
To change the pitch of a selected audio clip:
1
Select the Audio Type appropriate to the type
of material you are processing (Monophonic or
Polyphonic).
2
3 If transposing the pitch of the selection up, attenuate the Gain control as necessary.
4
To change the time of a selected audio clip:
1
Select AudioSuite > Pitch Shift > X-Form.
Select the Audio Type appropriate to the type
of material you are processing (Monophonic or
Polyphonic).
If compressing the duration of the selection,
attenuate the Gain control as necessary.
7
Adjust the Transient controls as desired.
Enable the desired Range button (2x, 4x, or 8x)
to set the possible range for time change.
5
Adjust the Time Shift control to the desired
amount of time change. Time change is measured in terms of the target duration using the
selected timebase or as a percentage of the original speed.
6
7
142
Enable the desired Range button (2x, 4x, or 8x)
to set the possible range for pitch change.
6
4
Click Render.
Audio Plug-Ins Guide
Adjust the Transient controls as desired.
5
2
3
Select AudioSuite > Pitch Shift > X-Form.
Adjust the Pitch Shift control to the desired
amount of pitch change. Pitch change is measured in semitones (and cents) or as a percentage of the original pitch.
If desired, click the IN button to enable Formant and adjust the Formant control.
8
Click Render.
Using X-Form for Post Production Pull Up and Pull Down
Tasks
The table below provides information on TCE settings for common post-production tasks. Type the
corresponding TCE% (represented to 10 decimal places in the following table) in the X-Form Time
Shift field for the corresponding post-production task and the process the selected audio.
Use the corresponding X-Form Plug-In Setting for the desired post-production task.
Desired Pull up or Pull Down
TCE% (to 10 Decimal Places)
Frames
Pal to Film –4%.tfx
96.0%
25 to 24/30
PAL to NTSC –4.1%.tfx
95.9040959041%
25 to 23.976/29.97
Film to PAL +4.1667%.tfx
+104.1666666667%
24/30 to 25
Film to NTSC –0.1%.tfx
99.9000999001%
24/30 to 23.976/29.97
NTSC to Pal +4.2667%.tfx
+104.2708333333%
23.976/29.97 to 25
NTSC to Film +0.1%.tfx
+100.10%
23.976/29.97 to 24/30
Chapter 25: X-Form
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Audio Plug-Ins Guide
Part V: Reverb Plug-Ins
Chapter 26: AIR Non-Linear Reverb
AIR Non-Linear Reverb is an RTAS plug-in. Use
the Non-Linear Reverb plug-in to apply special
gated or reversed Reverb effects to the audio signal, creating a synthetic, processed ambience.
Dry Delay
The Dry Delay control applies a specified
amount of delay to the dry portion of the signal,
which can create a “reverse reverb” effect, where
the reverb tail is heard before the dry signal.
Reverb Time
Adjust the Reverb Time to change the length of
the reverberation’s decay.
Mix
The Mix control lets you adjust the Mix between
the “wet” (processed) and “dry” (unprocessed)
signal. 0% is all dry, and 100% is all wet, while
50% is an equal mix of both.
Non-Linear Reverb plug-in window
Reverse
The Reverse button turns Reverse mode on and
off. In Reverse mode, the tail of the reverb signal
fades up to full volume, then disappears, rather
than fading out.
Non-Linear Reverb Reverb
Section Controls
The Reverb section provides control over the reverb’s diffusion and stereo width.
Diffusion
Pre-Delay
The Pre-Delay control determines the amount of
time that elapses between the original audio
event and the onset of reverberation.
Adjust the Diffusion control to change the rate
at which the sound density of the reverb tail increases over time. Higher Diffusion settings create a smoother reverberated sound. Lower settings result in more fluttery echo.
Width
The Width control lets you widen or narrow the
effect’s stereo field.
Chapter 26: AIR Non-Linear Reverb
147
Non Linear Reverb EQ Section
Controls
The EQ section provides tonal control over the
reverb signal.
Low Cut
The Low Cut control lets you adjust the frequency for the Low Cut filter. For less bass, raise
the frequency.
High Cut
The High Cut control lets you adjust the frequency for the High Cut filter. For less treble,
lower the frequency
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Audio Plug-Ins Guide
Chapter 27: AIR Reverb
AIR Reverb is an RTAS plug-in. Use the Reverb
effect to apply Reverb to the audio signal, creating a sense of room or space. Typically, you’ll
want to use Reverb on one of the Effect Send inserts or Main Effects inserts. This way you can
process audio from multiple Pro Tools tracks,
giving them all a sense of being in the same
space.
Reverb Controls
The Reverb plug-in provides a variety of controls for adjusting plug-in parameters.
Pre-Delay
The Pre-Delay control determines the amount of
time that elapses between the original audio
event and the onset of reverberation. Under natural conditions, the amount of pre-delay depends on the size and construction of the acoustic space, and the relative position of the sound
source and the listener. Pre-Delay attempts to
duplicate this phenomenon and is used to create
a sense of distance and volume within an acoustic space. Long Pre-Delay settings place the reverberant field behind rather than on top of the
original audio signal.
Room Size
Adjust the Room Size control to change the apparent size of the space.
Reverb Editor
Reverb Time
Adjust the Reverb Time to change the rate at
which the reverberation decays after the original direct signal stops. At its maximum value,
infinite reverberation is produced.
Chapter 27: AIR Reverb
149
Balance
Adjust the Balance control to change the output
level of the early reflections. Setting the Level
control to 0% produces a reverb effect that is
only the reverb tail.
Mix
The Mix control lets you adjust the Mix between
the “wet” (effected) and “dry” (unprocessed)
signal. 0% is all dry, and 100% is all wet, while
50% is an equal mix of both.
Medium Chamber Simulates a bright, medium-
sized room.
Large Chamber Simulates a bright, large-sized
room.
Small Studio Simulates a small, live, empty
room.
Large Studio Simulates a large, live, empty
room.
Scoring Stage Simulates a scoring stage in a medium-sized hall.
Reverb Early Reflections
Section Controls
Philharmonic Simulates the space and ambience
Different physical environments have different
early reflection signatures that our ears and
brain use to localize sound. These reflections affect our perception of the size of a space as well
as where an audio source sits within it. Changing early reflection characteristics changes the
perceived location of the reflecting surfaces surrounding the audio source.
Concert Hall Simulates the space and ambience
of a large concert hall.
of a large, symphonic, concert hall.
Church Simulates a medium-sized space with
natural, clear-sounding reflections.
Opera House Simulates the space and ambience
of an opera house.
Vintage 1 Simulates a vintage digital reverb ef-
Early reflections are simulated in Reverb by using multiple delay taps at different levels that
occur in different positions in the stereo spectrum (through panning). Long reverberation
generally occurs after early reflections dissipate.
Vintage 2 Simulates a vintage digital reverb ef-
Type
Controls the length of the early reflections.
The following Types of Early Reflection models
are provided:
Booth Simulates a vocal recording booth.
Club Simulates a small, clear, natural-sounding
club ambience.
Room Simulates the center of a small room
without many reflections.
Small Chamber Simulates a bright, small-sized
room.
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Audio Plug-Ins Guide
fect.
fect.
Spread
Reverb Plug-In Reverb Section
The Reverb section provides control over the
stereo width of the reverb algorithm.
In Width
Widens or narrows the stereo width of the incoming audio signal before it enters the reverb
algorithm.
Out Width
Widens or narrows the stereo width of the signal
once reverb has been applied.
Delay
Sets the size of the delay lines used to build the
reverb effect. Higher values create longer reverberation.
Reverb Room Section Controls
The Room section offers control over the overall
spatial feel of the simulated room.
Ambience
This control affects the attack of the reverb signal. At low settings, the reverb arrives quickly,
simulating a small room. At higher settings, the
reverb ramps up more slowly, emulating a larger
room.
Reverb High Frequencies
Section Controls
The High Frequencies section provides controls
that let you shape the tonal spectrum of the reverb by adjusting the decay times of higher frequencies.
Time
Adjust the Time control to decrease or increase
the decay time for mid- to high-range frequency
bands. Higher settings provide longer decay
times and lower settings provide shorter decay
time. With lower settings, high frequencies decay more quickly than low frequencies, simulating the effect of air absorption in a hall.
Freq
Adjust the Frequency control to set the frequency boundary between the mid- and highrange frequency bands.
Cut
The High Cut control lets you adjust the frequency for the High Cut filter (1.00–20.0 kHz).
Adjusting the High Cut control to change the decay characteristics of the high frequency components of the Reverb. To cut the high-end of the
processed signal, lower the frequency.
Density
Adjust the Density control to change the rate at
which the sound density of the reverb tail increases over time. Higher Density settings create a smoother reverberated sound. Lower settings result in more fluttery echo.
Chapter 27: AIR Reverb
151
Reverb Low Frequencies
Section Controls
The Low Frequencies section contains controls
that affect the low-frequency-heavy tail of the
reverb signal.
Time
Adjust the Time control to decrease or increase
the decay time for the low-range frequency
band. Higher settings provide longer decay
times and lower settings provide shorter decay
time.
Freq
Adjust the Frequency control to set the frequency boundary between the low and highrange frequency bands.
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Chapter 28: AIR Spring Reverb
AIR Spring Reverb is an RTAS plug-in. Use the
Spring Reverb plug-in for that classic spring reverb sound. Just don’t kick your computer trying to get the springs to rattle!
Spring Reverb Controls
The Spring Reverb plug-in provides a variety of
controls for adjusting plug-in parameters.
Pre-Delay
The Pre-Delay control determines the amount of
time (0–250 ms) that elapses between the original audio event and the onset of reverberation.
Reverb Time
Spring Reverb Plug-In window
The Spring Reverb plug-in models an analog
spring reverb. An analog spring reverb is an
electromechanical device much like a plate reverb. An audio signal is fed to a transducer at
the end of a long suspended metal coil spring.
The transducer causes the spring to vibrate,
which results in the signal reflecting from one
end of the spring to the other. At the other end
of the spring is another transducer that converts
the motion of the spring back into an electrical
signal, thus creating a delayed and reverberated
version of the input signal.
Adjust the Reverb Time to change the reverberation decay time (1.0–10.0 seconds) after the
original direct signal stops. Shorter times result
in a tighter, more ringing and metallic reverb,
such as when walking down a narrow hall with
hard floors and walls. Longer times result in a
larger reverberant space, such as an empty,
large, concrete cistern.
Mix
The Mix control lets you adjust the Mix between
the “wet” (reverbed) and “dry” (non-reverbed)
signal. 0% is all dry, and 100% is all wet, while
50% is an equal mix of both.
Low Cut
The Low Cut control lets you adjust the frequency of the Low Cut Filter
(20.0 Hz–1.00 kHz). Use the Low Cut filter to reduce some of the potential “boomyness” you can
get with longer Reverb Times.
Chapter 28: AIR Spring Reverb
153
Diffusion
Adjust the Diffusion control to change the rate
at which the sound density of the reverb tail increases over time. Higher Diffusion settings create a smoother reverberated sound. Lower settings result in more fluttery echo.
Width
Adjust the Width control to change the spread of
the reverberated signal in the stereo field. A setting of 0% produces a mono reverb, but leaves
the panning of the original source signal unprocessed. A setting of 100% produces a open,
panned stereo image.
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Chapter 29: D-Verb
D-Verb is a studio-quality reverb plug-in that is
available in TDM, RTAS, and AudioSuite formats.
The TDM version of the D-Verb plug-in is
not supported at 192 kHz. Use the RTAS
version instead.
D-Verb Controls
D-Verb provides a variety of controls for adjusting plug-in parameters.
D-Verb Output Meter
The Output meter indicates the output level of
the processed signal. With the stereo version of
D-verb, it represents the summed stereo output.
It is important to note that this meter indicates
the output level of the signal—not the input
level. If this meter clips, it is possible that the
signal clipped on input before it reached
D-Verb. Monitor your send or insert signal levels closely to help prevent this from happening.
D-Verb plug-in
Clip Indicator The Clip indicator shows if clipping has occurred. It is a clip-hold indicator. If
clipping occurs at any time during audio playback, the clip lights remain on. To clear the clip
indicator, click it. With longer reverb times
there is a greater likelihood of clipping occurring as the feedback element of the reverb builds
up and approaches a high output level.
Chapter 29: D-Verb
155
D-Verb Input Level Control
The Input Level slider adjusts the input volume
of the reverb to prevent the possibility of clipping and/or increase the level of the processed
signal.
D-Verb Mix Control
The Mix slider adjusts the balance between the
dry signal and the effected signal, giving you
control over the depth of the effect. This control
is adjustable from 100% to 0%.
D-Verb Algorithm Control
This control selects one of seven reverb algorithms: Hall, Church, Plate, Room 1, Room 2,
Ambience, or Nonlinear. Selecting an algorithm
changes the preset provided for it. Switching the
Size setting changes characteristics of the algorithm that are not altered by adjusting the decay
time and other user-adjustable controls. Each of
the seven algorithms has a distinctly different
character:
Hall A good general purpose concert hall with a
natural character. It is useful over a large range
of size and decay times and with a wide range of
program material. Setting Decay to its maximum value will produce infinite reverberation.
Church A dense, diffuse space simulating a
church or cathedral with a long decay time, high
diffusion, and some pre-delay.
Plate Simulates the acoustic character of a
metal plate-based reverb. This type of reverb
typically has high initial diffusion and a relatively bright sound, making it particularly good
for certain percussive signals and vocal processing. Plate reverb has the general effect of thickening the initial sound itself.
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Room 1 A medium-sized, natural, rich-sounding room that can be effectively varied in size
between very small and large, with good results.
Room 2 A smaller, brighter reverberant characteristic than Room 1, with a useful adjustment
range that extends to “very small.”
Ambient A transparent response that is useful
for adding a sense of space without adding a lot
of depth or density. Extreme settings can create
interesting results.
Nonlinear Produces a reverberation with a natu-
ral buildup and an abrupt cutoff similar to a
gate. This unnatural decay characteristic is particularly useful on percussion, since it can add
an aggressive characteristic to sounds with
strong attacks.
D-Verb Size Control
The Size control, in conjunction with the Algorithm control, adjusts the overall size of the reverberant space. There are three sizes: Small,
Medium, and Large. The character of the reverberation changes with each of these settings (as
does the relative value of the Decay setting). The
Size buttons can be used to vary the range of a
reverb from large to small. Generally, you
should select an algorithm first, and then choose
the size that approximates the size of the acoustic space that you are trying to create.
D-Verb Diffusion Control
Diffusion sets the degree to which initial echo
density increases over time. High settings result
in high initial build-up of echo density. Low settings cause low initial buildup. This control interacts with the Size and Decay controls to affect
the overall reverb density. High settings of diffusion can be used to enhance percussion. Use
low or moderate settings for clearer and more
natural-sounding vocals and mixes.
D-Verb Decay Control
Decay controls the rate at which the reverb decays after the original direct signal stops. The
value of the Decay setting is affected by the Size
and Algorithm controls. This control can be set
to infinity on most algorithms for infinite reverb times.
D-Verb Pre-Delay Control
Pre-Delay determines the amount of time that
elapses between the original audio event and the
onset of reverberation. Under natural conditions, the amount of pre-delay depends on the
size and construction of the acoustic space, and
the relative position of the sound source and the
listener. Pre-Delay attempts to duplicate this
phenomenon and is used to create a sense of distance and volume within an acoustic space. Long
Pre-Delay settings place the reverberant field
behind rather than on top of the original audio
signal.
D-Verb Hi Frequency Cut
Hi Frequency Cut controls the decay characteristic of the high frequency components of the reverb. It acts in conjunction with the Low Pass
Filter control to create the overall high frequency contour of the reverb. When set relatively low, high frequencies decay more quickly
than low frequencies, simulating the effect of air
absorption in a hall. The maximum value of this
control is Off (which effectively means bypass).
D-Verb Low Pass Filter
Low Pass Filter controls the overall high frequency content of the reverb by setting the frequency above which a 6 dB per octave filter attenuates the processed signal. The maximum
value of this control is Off (which effectively
means bypass).
Chapter 29: D-Verb
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Chapter 30: Reverb One
Reverb One is a world-class reverb processing
TDM plug-in. It provides a level of sonic quality
and reverb-shaping control previously found
only on the most advanced hardware reverberation units.
Reverb One features include:
• Editable Reverb EQ graph
• Editable Reverb Color graph
• Reverb Contour graph
• Dynamic control of reverb decay
A set of unique, easy-to-use audio shaping tools
lets you customize reverb character and ambience to create natural-sounding halls, vintage
plates, or virtually any type of reverberant space
you can imagine.
• Chorusing
• Early reflection presets
• Extensive library of reverb presets
• Supports 44.1 khz, 48 kHz, 88.2 kHz, and
96 kHz processing.
For sessions with a sample rate greater than
96 kHz, Reverb One will downsample and
upsample accordingly.
Reverb One
Chapter 30: Reverb One
159
A Reverb Overview
Digital reverberation processing can simulate
the complex natural reflections and echoes that
occur after a sound has been produced, imparting a sense of space and depth—the signature of
an acoustic environment. When you use a reverberation plug-in such as Reverb One, you are artificially creating a sound space with a specific
acoustic character.
This character can be melded with audio material, with the end result being an adjustable mix
of the original dry source and the reverberant
wet signal. Reverberation can take relatively
lifeless mono source material and create a stereo
acoustic environment that gives the source a
perceived weight and depth in a mix.
Creating Unique Sounds
In addition, digital signal processing can be
used creatively to produce reverberation characteristics that do not exist in nature. There are no
rules that need to be followed to produce interesting treatments. Experimentation can often
produce striking new sounds.
Acoustic Environments
When you hear live sound in an acoustic environment, you generally hear much more than
just the direct sound from the source. In fact,
sound in an anechoic chamber, devoid of an
acoustic space’s character, can sound harsh and
unnatural.
Each real-world acoustical environment, from a
closet to a cathedral, has its own unique acoustical character or sonic signature. When the reflections and reverberation produced by a space
combine with the source sound, we say that the
space is excited by the source. Depending on the
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Audio Plug-Ins Guide
acoustic environment, this could produce the
warm sonic characteristics we associate with reverberation, or it could produce echoes or other
unusual sonic characteristics.
Reverb Character
The character of a reverberation depends on a
number of things. These include proximity to
the sound source, the shape of the space, the absorptivity of the construction material, and the
position of the listener.
Reflected Sound
In a typical concert hall, sound reaches the listener shortly after it is produced. The original
direct sound is followed by reflections from the
ceiling or walls. Reflections that arrive within 50
to 80 milliseconds of the direct sound are called
early reflections. Subsequent reflections are
called late reverberation. Early reflections provide a sense of depth and strengthen the perception of loudness and clarity. The delay time between the arrival of the direct sound and the
beginning of early reflections is called the predelay.
The loudness of later reflections combined with
a large pre-delay can contribute to the perception of largeness of an acoustical space. Early reflections are followed by reverberation and repetitive reflections and attenuation of the
original sound reflected from walls, ceilings,
floors, and other objects. This sound provides a
sense of depth or size.
Reverb One provides control over these reverberation elements so that extremely naturalsounding reverb effects can be created and applied in the Pro Tools mix environment.
Reverb One Controls
Reverb One has a variety of controls for producing a wide range of reverb effects. Controls can
be adjusted by dragging their sliders or typing
values directly in their text boxes.
The harmonic spectrum of the reverb can also be
adjusted on the graph displays. See “Reverb One
Graphs” on page 165.
Reverb One Master Mix Controls
The Master Mix section has controls for adjusting the relative levels of the source signal and
the reverb effect, and also the width of the reverb effect in the stereo field.
Master Mix section
Wet/Dry
Adjusts the mix between the dry, unprocessed
signal and the reverb effect.
Stereo Width
Controls the width of the reverb in the stereo
field. A setting of 0% produces a mono reverb. A
setting of 100% produces maximum spread in
the stereo field.
Reverb One Dynamics Controls
The Dynamics section has controls for adjusting
Reverb One’s response to changes in input signal level.
Dynamics can be used to modify a reverb’s decay
character, making it sound more natural, or
conversely, more unnatural, depending on the
desired effect.
Typically, dynamics are used to give a reverb a
shorter decay time when the input signal is
above the threshold, and a longer decay time
when the input level drops below the threshold.
This produces a longer, more lush reverb tail
and greater ambience between pauses in the
source audio, and a shorter, clearer reverb tail in
sections without pauses.
For example, on a vocal track, use Dynamics to
make the reverb effect tight, clear, and intelligible during busy sections of the vocal (where the
signal is above the Threshold setting), and then
“bloom” or lengthen at the end of a phrase
(where the signal falls below the threshold).
Similarly, Dynamics can be used on drum tracks
to mimic classic gated reverb effects by causing
the decay time to cut off quickly when the input
level is below the threshold.
To hear examples of decay dynamics, load
one of the Dynamics presets using the Plug-In
Librarian menu.
100% Wet
Toggles the Wet/Dry control between 100% wet
and the current setting.
Dynamics section
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161
Decay Ratio
Depth
Controls the ratio by which reverb time is increased when a signal is above or below the
Threshold level. Dynamics behavior differs
when the Decay Ratio is set above or below 1. A
ratio setting of greater than 1 increases reverb
time when the signal is above the threshold. A
ratio setting of less than 1 increases a reverb’s
time when the signal is below the threshold.
Controls the amplitude of the sine wave generated by the LFO (low frequency oscillator) and
the intensity of the chorusing. The higher the
setting, the more intense the modulation.
For example, if Decay Ratio is set to 4, the reverb
time is increased by a factor of 4 when the signal
is above the threshold level. If the ratio is 0.25,
reverb time is increased by a factor of 4 when the
signal is below the Threshold level.
Threshold
Sets the input level above or below which reverb
decay time will be modified.
Chorus Controls
The Chorus section has controls for setting the
depth and rate of chorusing applied to a reverb
tail. Chorusing thickens and animates sounds by
adding a delayed, pitch-modulated copy of an
audio signal to itself.
Rate
Controls pitch modulation frequency. The
higher the setting, the more rapid the chorusing.
Setting the Rate above 20 Hz can cause frequency modulation to occur. This will add sideband harmonics and change the reverb’s tone
color, producing some very interesting special
effects.
Reverb One Reverb Section
Controls
The Reverb section has controls for the various
reverb tail elements, including level, time, attack, spread, size, diffusion, and pre-delay.
These determine the overall character of the reverb.
Chorusing produces a more ethereal or spacey
reverb character. It is often used for creative effect rather than to simulate a realistic acoustic
environment.
To hear examples of reverb tail chorusing,
load one of the Chorus presets using the
Plug-In Librarian menu.
Reverb section
Level
Controls the output level of the reverb tail.
When set to 0%, the reverb effect consists entirely of the early reflections (if enabled).
Chorus section
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Audio Plug-Ins Guide
Time
Controls the rate at which the reverberation decays after the original direct signal stops. The
value of the Time setting is affected by the Size
setting. You should adjust the reverb Size setting before adjusting the Time setting. If you set
Time to its maximum value, infinite reverberation is produced. The HF Damping and Reverb
Color controls also affect reverb Time.
Attack
Attack determines the contour of the reverberation envelope. At low Attack settings, reverberation builds explosively, and decays quickly. As
Attack value is increased, reverberation builds
up more slowly and sustains for the length of
time determined by the Spread setting.
When Attack is set to 50%, the reverberation envelope emulates a large concert hall (provided
the Spread and Size controls are set high
enough).
Spread
Controls the rate at which reverberation builds
up. Spread works in conjunctions with the Attack control to determine the initial contour and
overall ambience of the reverberation envelope.
Low Spread settings result in a rapid onset of reverberation at the beginning of the envelope.
Higher settings lengthen both the attack and
buildup stages of the initial reverb contour.
Size Determines the rate of diffusion buildup
and acts as a master control for Time and Spread
within the reverberant space.
Size values are given in meters and can be used
to approximate the size of the acoustic space you
want to simulate. When considering size, keep
in mind that the size of a reverberant space in
meters is roughly equal to its longest dimension.
Diffusion Controls the degree to which initial
echo density increases over time. High Diffusion
settings result in high initial buildup of echo
density. Low Diffusion settings cause low initial
buildup.
After the initial echo buildup, Diffusion continues to change by interacting with the Size control and affecting the overall reverb density. Use
high Diffusion settings to enhance percussion.
Use low or moderate settings for clearer, more
natural-sounding vocals and mixes.
Pre-Delay Determines the amount of time that
elapses between the original audio event and the
onset of reverberation. Under natural conditions, the amount of Pre-delay depends on the
size and construction of the acoustic space, and
the relative position of the sound source and the
listener. Pre-delay attempts to duplicate this
phenomenon and is used to create a sense of distance and volume within an acoustic space. Long
Pre-Delay settings place the reverberant field
behind rather than on top of the original audio
signal.
For an interesting musical effect, set the
Pre-Delay time to a beat interval such as
1/8, 1/16, or 1/32 notes.
Reverb One Early Reflection
Controls
The Early Reflections section has controls for
the various early reflection elements, including
ER setting, level, spread, and delay.
Calculating Early Reflections
A particular reflection within a reverberant field
is usually categorized as an early reflection.
Early reflections are usually calculated by measuring the reflection paths from source to lis-
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163
tener. Early reflections typically reach the listener within 80 milliseconds of the initial audio
event, depending on the proximity of reflecting
surfaces.
Simulating Early Reflections
Different physical environments have different
early reflection signatures that our ears and
brain use to pinpoint location information.
These reflections influence our perception of
the size of a space and where an audio source sits
within it. Changing early reflection characteristics changes the perceived location of the reflecting surfaces surrounding the audio source.
This is commonly accomplished in digital reverberation simulations by using multiple delay
taps at different levels that occur in different
positions in the stereo spectrum (through panning). Long reverberation generally occurs after
early reflections dissipate.
Reverb One provides a variety of early reflections models. These let you quickly choose a basic acoustic environment, then tailor other reverb characteristics to meet your precise needs.
ER Settings
Selects an early reflection preset. These range
from realistic rooms to unusual reflective effects. The last five presets (Plate, Build, Spread,
Slapback and Echo) feature a nonlinear response.
Early reflection presets include:
• Room: Simulates the center of a small room
without many reflections.
• Club: Simulates a small, clear, natural-sounding club ambience.
• Stage: Simulates a stage in a medium-sized
hall.
• Theater: Simulates a bright, medium-sized
hall.
• Garage: Simulates an underground parking
garage.
• Studio: Simulates a large, live, empty room.
• Hall: Places the sound in the middle of a hall
with reflective, hard, bright walls.
• Soft: Simulates the space and ambience of a
large concert hall.
• Church: Simulates a medium-sized space with
natural, clear-sounding reflections.
• Cathedral: Simulates a large space with long,
smooth reflections.
• Arena: Simulates a big, natural-sounding
empty space.
• Plate: Simulates a hard, bright reflection. Use
the Spread control to adjust plate size.
Early Reflections section
• Build: A nonlinear series of reflections
• Spread: Simulates a wide indoor space with
highly reflective walls.
• Slapback: Simulates a large space with a longdelayed reflection.
• Echo: Simulates a large space with hard, unnatural echoes. Good for dense reverb.
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Audio Plug-Ins Guide
Level
Controls the output level of the early reflections.
Turning the Early Reflections Level slider completely off produces a reverb made entirely of reverb tail.
Spread
Reverb One Graphs
The reverb graphs display information about the
tonal spectrum and envelope contour of the reverb. The Reverb EQ and Reverb Color graphs
provide graphic editing tools for shaping the
harmonic spectrum of the reverb.
Globally adjusts the delay characteristics of the
early reflections, moving them closer together
or farther apart. Use Spread to vary the size and
character of an early reflection preset. Setting
the Plate preset to a Spread value of 50%, for example, will change the reverb from a large,
smooth plate to a small, tight plate.
Delay Master
Determines the amount of time that elapses between the original audio event and the onset of
early reflections.
Early Reflect On
Toggles early reflections on or off. When early
reflections are off, the reverb consists entirely of
reverb tail.
Adjusting graph controls
Editing Graph Values
In addition to the standard slider controls, the
Reverb EQ and Reverb Color graph settings can
be adjusted by dragging elements of the graph
display.
To cut or boost a particular band:

Drag a yellow breakpoint up or down.
To adjust frequency or crossover:

Drag a triangular slider right or left.
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165
To adjust high-frequency cut or damp:

Drag the yellow dot right or left.
Band Cut/Boost
HF Cut/HF Damp
The high-frequency slider (64.0 Hz–24.0 kHz)
sets the frequency boundary between the mid
and high cut/boost points in the EQ.
Band Breakpoints Control cut and boost values
for the low, mid, and high frequencies of the EQ.
To cut a frequency band, drag a breakpoint
downward. To boost, drag upward. The adjustable range is from –24.0 dB to 12.0 dB.
HF Cut Breakpoint Sets the frequency above
Frequency/Cross-
Frequency/Cross-
Adjusting graph controls
Reverb EQ Graph
You can use this 3-band equalizer to shape the
tonal spectrum of the reverb. The EQ is post-reverb and affects both the reverb tail and the
early reflections.
Band Out/Boost
High-Frequency
which a 6 dB/octave low pass filter attenuates
the processed signal. It removes both early reflections and reverb tails, affecting the overall
high-frequency content of the reverb. Use the
HF Cut control to roll off high frequencies and
create more natural-sounding reverberation.
The adjustable range is from 120.0 Hz to
24.0 kHz.
Reverb Color Graph
You can use the Reverb Color graph to shape the
tonal spectrum of the reverb by controlling the
decay times of the different frequency bands.
Low and high crossover points define the cut
and boost points of three frequency ranges.
For best results, set crossover points at least two
octaves higher than the frequency you want to
boost or cut. For example, to boost a signal at
100 Hz, set the crossover to 400 Hz.
Low
Frequency
High-Frequency slider
Reverb EQ graph
Frequency Sliders Sets the frequency boundaries between the low, mid, and high band ranges
of the EQ.
The low frequency slider (60.0 Hz–22.5 kHz)
sets the frequency boundary between low and
mid cut/boost points in the EQ.
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Audio Plug-Ins Guide
Set the crossover to 500 Hz to boost low frequencies most effectively. Set it to 1.5 kHz to cut
low frequencies most effectively.
Band Cut/Boost
tings, high frequencies decay more quickly than
low frequencies, simulating the effect of air absorption in a hall. The adjustable range is from
120.0 Hz to 24.0 kHz.
High-Frequency
Reverb Contour Graph
The Reverb Contour graph displays the envelope
of the reverb, as determined by the early reflections and reverb tail.
Low Crossover High Crossover
Reverb Color graph
Crossover Sliders Sets the frequency boundar-
ies between the low, mid, and high frequency
ranges of the reverberation filter.
The low-frequency slider sets the crossover frequency between low and mid frequencies in the
reverberation filter. The adjustable range is
from 60.0 Hz to 22.5 kHz.
The high-frequency slider sets the crossover frequency between mid and high frequencies in the
reverberation filter. The adjustable range is
from 64.0 Hz to 24.0 kHz.
Band Breakpoints Controls cut and boost ratios
for the decay times of the low, mid, and highfrequency bands of the reverberation filter. To
cut a frequency band, drag a breakpoint downward. To boost, drag it upward. The adjustable
range is from 1:8 to 8:1.
HF Damp Breakpoint Sets the frequency above
which sounds decay at a progressively faster
rate. This determines the decay characteristic of
the high-frequency components of the reverb.
HF Damp works in conjunction with HF Cut to
shape the overall high -frequency contour of the
reverb. HF Damp filters the entire reverb with
the exception of the early reflections. At low set-
Reverb Contour graph
ER and RC Buttons Toggles the display mode.
Selecting ER (early reflections) displays early
reflections data in the graph. Selecting RC (reverb contour) displays the initial reverberation
envelope in the graph. Early Reflections and Reverb Contour can be displayed simultaneously.
Other Reverb One Controls
In addition to its reverb-shaping controls, Reverb One also features online help and level metering.
Online Help
To use online help, click the name of any control
or parameter and an explanation will appear.
Clicking the Online Help button itself provides
further details on using this feature.
Online help button
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167
Input Level Meters
Input meters indicate the input levels of the dry
audio source signal. Output meters indicate the
output levels of the processed signal.
An internal clipping LED will light if the reverb
is overloaded. This can occur even when the input levels are relatively low if there is excessive
feedback in the delay portion of the reverb. To
clear the Clip LED, click it.
Reverb One meters
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Audio Plug-Ins Guide
Chapter 31: ReVibe
ReVibe is a studio-quality reverb and acoustic
environment modeling TDM plug-in. ReVibe
works with mono, stereo, and greater-than-stereo multichannel audio. Revibe offers extensive
control over reverb characteristics, and a diverse array of room reflection and coloration
presets.
ReVibe makes it possible to model extremely realistic acoustic spaces and place audio elements
within them in a Pro Tools mix.
Revibe requires one or more HD Accel
cards.
ReVibe plug-in
Chapter 31: ReVibe
169
Reverberation Concepts
Digital reverberation processing can simulate
the complex natural reflections and echoes that
occur after a sound has been produced, imparting a sense of space and depth—the signature of
an acoustic environment. When you use a reverberation plug-in such as ReVibe, you are artificially creating a sound space with a specific
acoustic character.
This character can be melded with audio material, with the end result being an adjustable mix
of the original dry source and the reverberant
wet signal. You can use reverberation to enhance relatively lifeless mono source material
with a stereo acoustic environment that gives
the source audio a perceived weight and depth in
a mix.
Acoustic Environments
When you hear live sound in an acoustic environment, you generally hear much more than
just the direct sound from the source. In fact,
sound in an anechoic chamber, devoid of an
acoustic space’s character, can sound harsh and
unnatural.
Each real-world acoustical environment, from a
closet to a cathedral, has its own unique acoustical character or sonic signature. When the reflections and reverberation produced by a space
combine with the source sound, the space is said
to be excited by the source. Depending on the
acoustic environment, this could produce the
warm sonic characteristics associated with reverberation, or it could produce echoes or other
unusual sonic characteristics.
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Audio Plug-Ins Guide
Reverb Character
Reverb character depends on many factors including the shape of the space, the reflectivity of
the construction material, the proximity of reflective elements to the sound source, and the
position of the listener.
Reflected Sound
In a typical concert hall, sound reaches the listener shortly after it is produced. The original
direct sound is followed by reflections from the
ceiling or walls. These discrete reflections,
which usually arrive within 100 milliseconds of
the direct sound, are called early reflections. The
subsequent, and more diffuse reflections, are
called the reverb tail. The delay time between
the arrival of the direct sound and the beginning
of the reflected sounds is called the pre-delay.
The loudness and panning of early reflections
combined with the length of the pre-delay can
contribute to the perception of size of an acoustical space.
ReVibe also uses Room Coloration to accurately
model acoustic spaces and effects. Room Coloration is a complex filter process, similar to EQ,
that models the frequency shape of each room or
effect.
ReVibe provides control over these reverb elements so that extremely natural-sounding reverb effects can be created and applied in the
Pro Tools mix environment.
ReVibe can also be used to produce reverb characteristics that do not exist in nature. There are
no rules that you need to follow to produce interesting treatments. Experimentation can often
produce striking results.
Using ReVibe
ReVibe supports 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz sessions. ReVibe works with mono and stereo formats, and LCR, LCRS, quad, 5.0, and 5.1 greater-than-stereo multichannel formats.
In general, when working with stereo and greater-than-stereo tracks, use the multichannel version of
ReVibe.
Revibe supports the following combinations of track types and plug-in insert formats:
Track
Type
Mono
Stereo
LCR
LCRS
Quad
5.0
5.1
Plug–in Insert Format
Mono
Stereo
LCR
LCRS
Quad
5.0
5.1
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Chapter 31: ReVibe
171
Adjusting ReVibe Parameters
To adjust EQ frequency crossover:

Drag the control dot right or left.
You can adjust ReVibe parameters by adjusting
the slider controls, dragging dots on the graph
display, or using your computer keyboard.
Editing Slider Controls with a Mouse
You can adjust slider controls with a mouse by
dragging horizontally. Parameter values increase as you drag to the right, and decrease as
you drag to the left.
Some sliders (such as the Diffusion slider) are
bipolar, meaning that their zero position is in
the center of the slider’s range. Dragging to the
right of center creates a positive parameter
value; dragging to the left of center generates a
negative parameter value.
Setting the EQ crossover frequency
To adjust high frequency rear cut:

Drag the control dot right or left.
Editing Graph Display Parameters with a Mouse
You can adjust parameters on the Decay Color &
EQ graph displays with a mouse by dragging the
appropriate dot on the graph.
To cut or boost a particular EQ band:

Drag a control dot up or down.
Cutting or boosting an EQ frequency band
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Audio Plug-Ins Guide
Setting the rear cut frequency
Editing Parameters with a Computer Keyboard
Each control has a corresponding parameter
text field that displays the current value of the
parameter. You can edit the numeric value of a
parameter with your computer keyboard.
To change control values with a computer
keyboard:
ReVibe Controls
Click on the parameter text that you want to
edit.
ReVibe has a variety of controls for producing a
wide range of reverb effects. Controls can be adjusted by dragging their sliders, typing values
directly in their text boxes, and adjusted on the
Decay Color & EQ graph displays.
1
Change the value by doing one of the following.
2
• To increase a value, press the Up Arrow on
your keyboard. To decrease a value, press
the Down Arrow on your keyboard.
– or –
• Type the desired value.
For parameters with values in kilohertz,
typing “k” after a number value will multiply the value by 1000. For example, type
“8k” to enter a value of 8000.
3
ReVibe Master Mix Section
Controls
The Master Mix section has controls for adjusting the relative levels of the source signal and
the reverb effect.
Do one of the following:
• Press Enter on the numeric keyboard to input the value and remain in keyboard editing mode.
– or –
• Press Enter on the alpha keyboard (Windows) or Return (Mac) to enter the value
and leave keyboard editing mode.
To move from a selected parameter to the
next parameter, press the Tab key. To move
backward, press Shift+Tab.
Enabling Switches
To enable a switch, click on the switch (the
round LED indicator next to each switch name).
Switch LEDs illuminate when enabled.
Master Mix controls
Wet/Dry Control
Wet/Dry adjusts the mix between the dry, unprocessed signal and the reverb effect. If you insert the ReVibe plug-in directly onto an audio
track, settings from 30% to 60% are a good starting point for experimenting with this parameter. The range of this control is from 0% to
100%.
You can also achieve a 100% wet mix by
clicking the 100% Wet Mix button.
Early Reflection switch LED (on)
Chapter 31: ReVibe
173
Stereo Width Control
Stereo Width controls the stereo field spread of
the front reverb channels. A setting of 0% produces a mono reverb, but leaves the panning of
the original source signal unaffected. A setting
of 100% produces a hard panned stereo image.
ReVibe Chorus Section Controls
The Chorus section has controls for adjusting
the depth and rate of chorusing applied to the
reverb tail. Chorusing thickens and animates
sounds and produces a more ethereal reverb
character. It is often used for creative effects
rather than to simulate a realistic acoustic environment.
Stereo Width control
Settings above 100% use phase inversion to create an even wider stereo effect. The Stereo
Width slider displays red above the 100% mark
to remind you that a phase effect is being used to
widen the stereo field.
The range of this control is from 0% to 150%.
The default setting is 100%.
The Stereo Width control does not affect the
reverberation effect coming through the rear
channels. If you want to produce a strictly
mono reverb, be sure to set the Rear Reverb
parameter (Levels section) to –INF dB .
174
Chorus controls
Depth Control
Depth controls the amplitude of the sine wave
generated by the LFO (low frequency oscillator)
and the intensity of the chorusing. The higher
the setting, the more intense the modulation.
The range of this control is from 0% to 100%.
100% Wet Mix Button
Rate Control
This button toggles the Wet/Dry control between 100% wet and the current setting. A 100%
wet mix contains only the reverb effect with
none of the direct signal. This setting can be
useful when using pre-fader sends to achieve
send/return bussing. The wet/dry balance in the
mix can be controlled using the track faders for
the dry signal, and the Auxiliary input fader for
the effect return.
Rate controls the frequency of the LFO. The
higher the setting, the more rapid the chorusing.
The range of this control is from 0.1 Hz to
30.0 Hz.
Audio Plug-Ins Guide
Setting the Rate above 20 Hz can cause frequency modulation to occur. This will add sideband harmonics and change the reverb’s tone
color, producing interesting effects. Typical settings are between 0.2 Hz and 1.0 Hz.
Chorus On/Off Button
Level Control
This button toggles the chorus effect on or off.
Level controls the output level of the early reflections. Setting the Level slider to –INF (minus infinity) eliminates the early reflections
from the reverb effect. The range of this control
is from –INF to 6.0 dB.
Chorus on/off button
ReVibe Early Reflection Section
Different physical environments have different
early reflection signatures that our ears and
brain use to pinpoint location information in
physical space. These reflections influence our
perception of the size of a space and where an
audio source sits within it.
Changing early reflection characteristics
changes the perceived location of the reflecting
surfaces surrounding the audio source. In general, the reverb tail continues after early reflections dissipate.
ReVibe room presets use multiple delay taps at
different levels, different times, and in different
positions in the multichannel environment
(through 360° panning) to create extremely realistic sounding environments.
The Early Reflect section has controls for adjusting the various early reflection elements, including level, spread, and pre-delay.
Spread Control
Spread globally adjusts the delay characteristics
of the early reflections, moving the individual
delay taps closer together or farther apart. Use
Spread to vary the size and character of an early
reflection preset. The range of this control is
from –100% to 100%.
At 0%, the early reflections are set to their optimum value for the room preset. Typical spread
values range between –25% and 25%.
Setting Spread to 100% produces very
widely spaced early reflections that may
sound unnatural. At –100% the early reflections have no spread at all, and are heard as
a single reflection.
Pre-Delay Control
The Pre-Delay control in the Early Reflect section determines the amount of time that elapses
between the onset of the dry signal and the first
early reflection delay tap. Some Room Types,
such as those that produce slapback effects, have
additional built-in pre-delay. The range of this
control is from –300.0 ms to 300.0 ms.
Negative Pre-Delay times imply that some early
reflection delay taps should sound before the
original dry signal. Since this is not possible,
any of the delay taps that would sound before
the dry signal are not used and do not sound.
Early reflection section
Chapter 31: ReVibe
175
When Pre-Delay Link is enabled, negative early
reflection Pre-Delay times can be used to make
the early reflections start before the reverb tail,
if desired.
Pre-Delay Link Button
ReVibe Levels Section Controls
The Levels section has controls for adjusting
source input and ReVibe output levels. ReVibe
provides individual output level controls for
front, center, rear reverb, and rear early reflections.
The Pre-Delay Link button toggles linking of the
Early Reflection Pre-Delay control and the Reverb Pre-Delay control. When linked, the Early
Reflection Pre-Delay is offset by the Reverb PreDelay amount, so that the total delay for the
early reflections is the sum of the Early Reflection Pre-Delay and the Reverb Pre-Delay.
Levels controls
Pre-Delay Link button
ER On/Off Button
This button toggles early reflections on or off.
When early reflections are off, the reverb effect
consists entirely of reverb tail.
ER On/Off button
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Audio Plug-Ins Guide
In stereo and greater-than-stereo formats where
there is no center channel or where there are no
rear channels, the center and rear level controls
can be used to augment the reverb sound. Reverb and early reflections that would be heard
either from the center channel or from the rear
channels can be mixed into the front left and
right channels.
Input Control
Input adjusts the level of the source input to
prevent internal clipping. The range of this control is from –24.0 dB to 0.0 dB. Lowering the Input control does not change the levels shown on
the input side of the Input/Output meter, which
shows the level of the signal before the Input
control.
Front Control
Rear Level Link Button
Front controls the output level of the front left
and right outputs. Front is also the main level
control for stereo. The range of this control is
from –INF (minus infinity) to 0.0 dB.
The Rear Level Link button toggles linking of
the Rear Reverb and Rear ER controls on or off.
The Rear Reverb and the Rear ER controls are
linked by default. When linked, the Rear ER and
Rear Reverb controls move in tandem when either is adjusted. When unlinked, the Rear ER
and the Rear Reverb controls can be adjusted independently.
Center Control
Center controls the output level of the center
channel outputs of multichannel formats that
have a center channel (such as LCR or 5.1).
When ReVibe is used in a multichannel format
that has no center channel (such as stereo or
quad), the Center level control adjusts a phantom center channel signal that is center-panned
to the front left and right outputs.
The range of this control is from –INF (minus
infinity) to 0.0 dB.
Rear Reverb Control
Rear Reverb controls the output level of the rear
outputs of multichannel formats that have rear
channels (such as quad or 5.1).
When ReVibe is used in a multichannel format
that has no rear channels (such as a stereo or
LCR) the Rear level control instead adjusts rear
channel signals hard-panned to the front left
and right outputs.
The range of this control is from –INF (minus
infinity) to 0.0 dB.
Rear ER Control
Rear Level Link button
ReVibe Room Type Section
Controls
The controls in the Room Type section let you
select a Room Type, which models early reflection characteristics for specific types of rooms
or effects devices. Each Room Type also incorporates a complex room coloration EQ, which
models the general frequency response of various rooms and effects devices.
Choosing a new Room Type changes the early reflections and room coloration EQ only. All of the
other ReVibe parameters and setting remain unchanged. To create a preset that includes all parameters, use the Plug-In Settings menu.
For more information on saving and importing plug-in presets, see the Pro Tools
Reference Guide.
Rear ER controls the output level of early reflections in the rear outputs. The range of this control is from –INF (minus infinity) to 0.0 dB.
The Rear ER control has no effect when the
early reflections are turned off with the ER
On/Off button.
Chapter 31: ReVibe
177
Room Type Number
Room Type Category pop-up
ReVibe Room Coloration Section
Controls
The Room Coloration controls work in conjunction with the selected Room Type. Coloration
takes the characteristic resonant frequencies or
EQ traits of the room and allows you to apply
this spectral shape to the reverb.
Room Type Name pop-up
Preset Next and
Previous buttons
Room Type display and controls
In addition to letting you adjust the overall
sound of the room, the high-frequency and lowfrequency components are split to allow you to
emphasize or de-emphasize the low and high
frequency response of the room.
The Room Type display shows the Room Type
Category, Room Type Name, Room Type Number and the Next and Previous browse buttons.
Room Type Category Menu
Clicking on the Room Type Category pop-up
menu lets you select one of the 14 Room Type
categories, and selects the first Room Type preset in that category.
Room Type Name Menu
Click the Room Type Name pop-up menu to select from a list of all available Room Type presets.
See “ReVibe Room Types” on page 185 for a
list of room presets.
Room Type Number Field
The Room Type Number field displays the Room
type number for the current Room Type.
Room Coloration controls
Coloration Control
Coloration adjusts how much of the EQ characteristics of the selected Room Type are applied
to the original signal. The range of this control
is from 0% to 200%. A setting of 100% provides
the optimum coloration for the room type. Settings above 100% will tend to produce extreme
and unnatural coloration.
HF Color Control
HF Color adds or subtracts additional high frequency coloration, or relative brightness, to the
acoustic model of the room. The range of this
control is from –50.0% to 50.0%.
Next and Previous Buttons
LF Color Control
Click the Next or Previous buttons to choose the
next or previous Room Type.
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Audio Plug-Ins Guide
LF Color adds or subtracts additional low frequency coloration, or relative darkness, to the
acoustic model of the room. The range of this
control is from –50.0% to 50.0%.
ReVibe Reverb Section Controls
Level Control
The Reverb section has controls for the various
reverb tail elements, including type, level, time,
size, spread, attack time, attack shape, rear
shape, diffusion, and pre-delay. These determine
the overall character of the reverb tail.
Level controls the output level of the reverb tail.
When set to –INF (minus infinity) no reverb tail
is heard, and the reverb effect consists entirely
of the early reflections (if enabled). The range of
this control is from –INF to 6.0 dB.
Time Control
Time controls how long the reverberation continues after the original source signal stops. The
range of this control is from 100.0 ms to Inf (infinity). Setting Time to its maximum value will
produce infinite reverberation.
Pre-Delay Control
Reverb Controls
The Pre-Delay control in the Reverb section sets
the amount of time that elapses between signal
input and the onset of the reverb tail.
Type Menu
Type is a pop-up menu that sets the type of reverb tail. There are nine basic reverb types, plus
the Automatic type. Selecting the Automatic reverb type will select the type of reverb tail that is
stored with the currently selected room type.
The reverb types are:
• Automatic selects the reverb tail type
stored with the room type.
• Natural is an average reverb tail type with
no extreme characteristics.
Under natural conditions, the amount of pre-delay depends on the size and construction of the
acoustic space and the relative position of the
sound source and the listener. Pre-delay attempts to duplicate this phenomenon and is
used to create a sense of distance and volume
within an acoustic space. Extremely long predelay settings produce effects that are unnatural
but sonically interesting.
The range of this control is from 0.0 ms to
300.0 ms.
• Smooth is optimized for large rooms.
• Fast Attack can be useful for plate reverbs.
Diffusion Control
• Dense is similar to smooth, and can also be
good for a plate reverb.
Diffusion controls the rate that the sound density of the reverb tail increases over time. The
control ranges between –50% and 50%. At 0%,
diffusion is set to an optimal preset value. Positive Diffusion settings create a longer initial
buildup of echo density. At negative settings,
the buildup of echo density is slower than at the
optimal preset value.
• Tight is good for small to medium rooms.
• Sparse 1 produces sparse early reflections
with a high diffusion buildup.
• Sparse 2 can be useful for a spring reverb.
• Wide is a generic large reverb.
• Small is optimized for small rooms.
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Attack Time Control
Spread Control
Attack Time adjusts the length of time between
the start of the reverb tail and its peak level. Settings are Short, Medium, or Long.
Spread controls the rate at which reverberation
builds up. Spread works in conjunction with the
Attack Shape control to determine the initial
contour and overall ambience of the reverberation envelope.
Attack Shape Control
Attack Shape determines the contour of the attack portion of the reverberation envelope. At
0%, there is no buildup contour, and the reverb
tail begins at its peak level. At a high Attack
Shape setting the reverb tail begins at a relatively low initial level and ramps up to the peak
reverb level. The range of this control is from 0%
to 100%.
Rear Shape Control
Rear Shape adjusts the envelope of the reverb in
the rear channels to control the length of the attack time. This gives more reverb presence and a
longer reverb bloom in the rear channels. The
range of this control is from 0% to 100%.
Size Control
At low Spread settings there is a rapid onset of
reverb at the beginning of the reverberation envelope. Higher settings lengthen both the attack
and buildup of the initial reverb contour. The
range of this control is from 0% to 100%.
ReVibe Decay Color & EQ
Section Controls
The Decay Color and EQ section provides an editable graphic display of reverb decay color parameters and EQ parameters. Click the EQ button to toggle the display to show EQ parameters.
Click the Color button to toggle the display to
show Color parameters. To edit a parameter on
the graph, drag the appropriate dot.
The Size control adjusts the apparent size of the
reverberant space from small to large. Set the
Size control to approximate the size of the
acoustic space you want to simulate. Size values
are given in meters. The range of this control is
from 2.0 m to 60.0 m (though relative size will
change based on the current Room Type).
Larger settings of the Size parameter increase
both the Time and Spread parameters.
When specifying reverb size, keep in mind
that the size of a reverberant space in meters
is approximately equal to its longest dimension. In general, halls range from 25 m to
50 m; large to medium rooms range from
15 m to 30 m; and small rooms range from
5 m to 20 m. Similarly, a Room Size setting
of 20m corresponds roughly to a 4x8 plate.
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Audio Plug-Ins Guide
Decay Color & EQ display
Each control point (dot) on the graph has corresponding parameter text fields above the display
that show the current parameter values. You can
edit the numeric value of a parameter with your
computer keyboard. (See “Editing Parameters
with a Computer Keyboard” on page 172.)
Low Frequency Ratio Control
Low Frequency Ratio sets cut or boost ratios for
the decay times of the low and mid frequency
bands of the reverberation filter. The range of
this control is between 1:16.0 and 4.0:1.
ReVibe Decay Color Section
You can use the controls in the Decay Color section to shape the tonal spectrum of the reverb by
adjusting the decay times of the low and high
frequency ranges. Low and high crossover
points define the cut and boost points of three
frequency ranges.
For best results, set crossover points at least one
octave higher than the frequency you want to
boost or cut. To boost a signal at 200 Hz, for example, set the crossover to 400 Hz.
Low Frequency Ratio control
High Frequency Crossover Control
High Frequency Crossover sets the crossover
frequency at which transitions from mid frequencies to high frequencies take place in the reverberation filter. The range of this control is
from 1.5 kHz to 20.0 kHz.
Low Frequency Crossover Control
Low Frequency Crossover sets the crossover frequency at which transitions from low frequencies to mid frequencies take place in the reverberation filter. The range of this control is from
50.0 Hz to 1.5 kHz.
High Frequency Crossover control
High Frequency Ratio Control
High Frequency Ratio sets cut or boost ratios for
the decay times of the mid and high frequency
bands of the reverberation filter. The range of
this control is between 1:16.0 and 4.0:1.
Low Frequency Crossover control
High Frequency Ratio control
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181
ReVibe Decay EQ Section
Low Frequency Control
Low Frequency sets the frequency boundary between low and mid cut or boost points in the reverb EQ. The range of this control is from
50.0 Hz to 1.5 kHz.
High Gain Control
High Gain sets cut and boost values for the mid
and high frequencies of the reverb decay EQ.
The range of this control is from –24.0 dB to
12.0 dB.
High Gain control
Low Frequency control
Low Gain Control
Low Gain sets cut and boost values for the low
and mid frequencies of the reverb decay EQ. The
range of this control is from –24.0 dB to 12.0 dB.
High Frequency Rear Cut Control
High Frequency Rear Cut rolls off additional
high frequencies in the rear channels of the early
reflections and reverb tail. The application of
this filter is distinct from the application of Decay Color and Decay EQ. The range of this control is from 250.0 Hz to 20.0 kHz.
Low Gain control
High Frequency Control
High Frequency sets the frequency boundary between mid and high cut or boost points in the reverb EQ. The range of this control is from
1.5 kHz to 20.0 kHz.
High Frequency control
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Audio Plug-Ins Guide
High Frequency Rear Cut control
ReVibe Contour Display
The Contour display shows the current reverb
shape and early reflections as a two-dimensional
graph. Both front and rear reverb tail shapes and
early reflections can be viewed at the same time.
Buttons below the display allow you to select the
type of data being displayed.
Front reverb
Rear reverb
RC Button
The RC (reverb contour) button toggles display
of the reverb contours for both the front and
rear channels on or off within the Contour display. When the RC button is illuminated, the reverberation envelopes are displayed. When the
RC button is not illuminated, the reverberation
envelopes are not displayed. Both early reflections and reverb contour data can be displayed
simultaneously.
Front Button
The Front button toggles display of the front
channel reverb contour and the front channel
early reflections on or off within the Contour
display. When the Front button is illuminated,
the initial reverberation envelope and early reflections for the front channels are displayed.
When the Front button is not illuminated, they
are not displayed.
Early reflections
Contour display
ER Button
The ER (early reflections) button toggles display
of early reflections on or off within the Contour
display. When the ER button is illuminated,
early reflections data is displayed. When the ER
button is not illuminated, early reflections data
is not displayed. Both early reflections and reverb contour data can be displayed simultaneously.
Rear Button
The Rear button toggles display of the rear channel reverb contour and the rear channel early reflections on or off within the Contour display.
When the Rear button is illuminated, the initial
reverberation envelope and early reflections for
the rear channels are displayed. When the Rear
button is not illuminated, they are not displayed.
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183
ReVibe Input/Output Meter
ReVibe Online Help Button
The Input/Output meter indicates the input signal and the ReVibe output. The range of this meter is from 0 dB to –60 dB. The number of input/output meters that operate simultaneously
ranges from a single meter for mono input and
output, up to five input/output meters for 5.0
and 5.1 multichannel processing. The meters
that operate depend on the channel format of
the track on which the plug-in is inserted.
Click the name of any control and information
about that control will appear. Clicking the Online Help button provides additional details on
using this feature.
internal clip indicator
channel clip indicator
Input/Output Meter
Clip Indicators
A red channel clip indicator appears at the top of
each meter, and an internal clip meter appears
above the meter display itself. The clip indicator
lights when the signal level exceeds 0 dB, and
stays lit until the user clears it. Clicking a meter’s clip indicator will clear that meter.
It is possible to clip internally even when input
levels are relatively low. This can occur because
a digital reverb is essentially a series of filters
and delays. Feedback within the signal paths can
cause buildup of the reverb signal, which can
cause the level to increase and overload (similar
to a delay line with a high level of feedback).
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Audio Plug-Ins Guide
Online Help
ReVibe Room Types
Revibe comes with over 200 built-in Room Type
presets in 14 Room Type categories. These Room
Type presets contain complex early reflections
and room coloration characteristics that define
the sound of the space. The Room Type categories and their presets are as follows:
Rooms
Large Bright Room 1
Large Bright Room 2
Large Neutral Room 1
Large Neutral Room 2
Large Dark Room 1
Large Dark Room 2
Large Boomy Room
Studios
Large Natural Studio 1
Large Natural Studio 2
Large Live Room 1
Large Live Room 2
Large Dense Studio 1
Large Dense Studio 2
Medium Natural Studio 1
Medium Natural Studio 2
Medium Natural Studio 3
Medium Natural Studio 4
Medium Live Room 1
Medium Live Room 2
Medium Dense Studio 1
Medium Dense Studio 2
Small Natural Studio 1
Small Natural Studio 2
Small Natural Studio 3
Small Natural Studio 4
Medium Bright Room 1
Medium Bright Room 2
Medium Bright Room 3
Medium Neutral Room 1
Medium Neutral Room 2
Medium Neutral Room 3
Medium Dark Room 1
Medium Dark Room 2
Medium Dark Room 3
Small Bright Room 1
Small Bright Room 2
Small Bright Room 3
Small Neutral Room 1
Small Neutral Room 2
Small Neutral Room 3
Small Dark Room 1
Small Dark Room 2
Small Boomy Room
Small Natural Studio 5
Small Dense Studio 1
Small Dense Studio 2
Vocal Booth 1
Vocal Booth 2
Vocal Booth 3
Vocal Booth 4
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185
Halls
Natural Cathedral 1
Large Natural Hall 2
Natural Cathedral 2
Large Natural Hall 3
Natural Cathedral 3
Large Natural Hall 4
Dense Cathedral 1
Large Natural Hall 5
Dense Cathedral 2
Large Natural Hall 6
Slap Cathedral
Large Dense Hall
Large Sparse Hall
Medium Natural Hall 1
Medium Natural Hall 2
Medium Natural Hall 3
Medium Natural Hall 4
Medium Dense Hall
Small Natural Hall 1
Small Natural Hall 2
Theaters
Large Theater 1
Large Theater 2
Medium Theater 1
Medium Theater 2
Small Theater 1
Small Theater 2
Churches
186
Cathedrals
Large Natural Hall 1
Plates
Large Natural Plate
Large Bright Plate
Large Synthetic Plate
Medium Natural Plate
Medium Bright Plate
Small Natural Plate
Small Bright Plate
Springs
Guitar Amp Spring 1
Guitar Amp Spring 2
Guitar Amp Spring 3
Guitar Amp Spring 4
Guitar Amp Spring 5
Guitar Amp Spring 6
Studio Spring 1
Studio Spring 2
Large Natural Church 1
Studio Spring 3
Large Natural Church 2
Studio Spring 4
Large Dense Church
Dense Spring 1
Large Slap Church
Dense Spring 2
Medium Natural Church 1
Resonant Spring
Medium Natural Church 2
Funky Spring 1
Medium Dense Church
Funky Spring 2
Small Natural Church 1
Funky Spring 3
Small Natural Church 2
Funky Spring 4
Audio Plug-Ins Guide
Chambers
Film and Post
Large Chamber 1
Medium Kitchen
Large Chamber 2
Small Kitchen
Large Chamber 3
Bathroom 1
Large Chamber 4
Bathroom 2
Large Chamber 5
Bathroom 3
Large Chamber 6
Bathroom 4
Medium Chamber 1
Bathroom 5
Medium Chamber 2
Shower Stall
Medium Chamber 3
Hallway
Medium Chamber 4
Closet
Medium Chamber 5
Classroom 1
Small Chamber 1
Classroom 2
Small Chamber 2
Large Concrete Room
Small Chamber 3
Medium Concrete Room
Small Chamber 4
Locker Room
Ambience
Large Ambience 1
Large Ambience 2
Large Ambience 3
Large Ambience 4
Medium Ambience 1
Medium Ambience 2
Medium Ambience 3
Medium Ambience 4
Medium Ambience 5
Small Ambience 1
Small Ambience 2
Small Ambience 3
Muffled Room
Very Small Room 1
Very Small Room 2
Very Small Room 3
Car 1
Car 2
Car 3
Car 4
Car 5
Phone Booth
Metal Garbage Can
Drain Pipe
Tin Can
Very Small Ambience
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187
Large Spaces
Effects
Parking Garage 1
Mono Slapback 1
Parking Garage 2
Mono Slapback 2
Parking Garage 3
Mono Slapback 3
Warehouse 1
Wide Slapback 1
Warehouse 2
Wide Slapback 2
Stairwell 1
Wide Slapback 3
Stairwell 2
Multi Slapback 1
Stairwell 3
Multi Slapback 2
Stairwell 4
Multi Slapback 3
Stairwell 5
Multi Slapback 4
Gymnasium
Spread Slapback 1
Auditorium
Spread Slapback 2
Indoor Arena
Mono Echo 1
Stadium 1
Mono Echo 2
Stadium 2
Mono Echo 3
Tunnel
Wide Echo 1
Vintage Digital
Large Hall Digital
Medium Hall Digital
Large Room Digital
Medium Room Digital
Small Room Digital
Wide Echo 2
Multi Echo 1
Multi Echo 2
Prism
Prism Reverse
Inverse Long
Inverse Medium
Inverse Short
Stereo Enhance 1
Stereo Enhance 2
Stereo Enhance 3
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Chapter 32: TL Space TDM and TL Space
Native
TL Space is a convolution reverb plug-in that is
available in TDM, RTAS, and AudioSuite formats. There are two versions of TL Space:
TL Space TDM and TL Space Native. TL Space
TDM includes TDM, RTAS, and AudioSuite
plug-in formats. TL Space Native includes RTAS
and AudioSuite plug-in formats only.
TL Space was designed to be the ultimate reverb
for music and post-production applications. By
combining the sampled acoustics of real reverb
spaces with advanced DSP algorithms, TL Space
offers stunning realism with full control of reverb parameters in mono, stereo, and surround
formats.
TL Space plug-in
Chapter 32: TL Space TDM and TL Space Native
189
TL Space Feature Highlights
• Automatically recognizes common IR formats for one click loading
TL Space has an extensive feature set designed
to assist users in creating the best reverb effect
in the shortest possible time.
• IR browser hides to save screen real estate
Listed below are some of the key innovations
that TL Space offers over traditional software
reverbs.
Reverb Features
• Mono, Stereo, and Quad and 5.0–channel
output support
• Multiband EQ
• Independent wet/dry and decay levels
• Separate reverb early and late levels and
length
• Control of early size, low-cut, and balance
• Pre delay and late delay controls
• Precise control of low, mid, and high decay
crossover
• Quick browser buttons allow rapid IR loading and preview
Automation and Ease of Use Features
• Snapshot mode supports rapid changes between ten predefined reverb scenes
• Picture preview mode allows user to view
image files stored with impulse responses
• Impulse responses stored directly in
Pro Tools presets and sessions for easy session sharing
• New impulse responses can be copied to
system and loaded without closing
TL Space
• iLok support for quick and easy relocation
to other Pro Tools systems
Surround and Post-Production Features
• Adjustable waveform reverse, displayed in
beats per minute
• Full input and output surround metering on
screen at all times
• Waveform processing bypass
• Separate front, center, and rear levels
• Independent front and rear decay
Interface Features
• Full waveform view, zoom, and channel
highlight functions
• Onscreen input and output metering with
clip indicators
• Impulse response information display
Impulse Response (IR) Loading and
Organization Features
• Scrollable IR browser makes finding impulse responses easy
• Browser supports user defined IR groups on
any local drives
• Browser keyboard shortcuts
• IR favorites function
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Audio Plug-Ins Guide
• Snapshot mode ideal for post automation
requirements
• Seamless snapshot switching (RTAS)
• Automatic phantom channel creation
IR Library
• A wide variety of both real and synthetic reverb spaces and effects
• Mono, stereo, and surround formats
• All reverb impulse responses stored in WAV
file format
TL Space Overview
The following sections provide information on
the concepts of reverb and convolution reverb.
Reverb Basics
Reverberation is an essential aspect of the sound
character of any space in the real world. Every
room has a unique reverb sound, and the qualities of a reverb can make the difference between
an ordinary and an outstanding recording. The
same reverb principles responsible for the
sound of a majestic, soaring symphony in a concert hall also produce the booming, unintelligible PA system at a train station. Recordings of
audio in the studio context have traditionally
been captured with a minimum of real reverb,
and engineers have sought to create artificial reverbs to give dry recorded material additional
dimension and realism.
The first analog reverbs were created using the
‘echo chamber’ method, which is comprised of a
speaker and microphone pair in a quiet, closed
space with hard surfaces, often a tiled or concrete room built in the basement of a recording
studio. Chamber reverbs offered a realistic,
complex reverb sound but provided very little
control over the reverb, as well as requiring a
large dedicated room.
Plate reverbs were introduced by EMT in the
1950s. Plate reverbs provide a dense reverb
sound with more control over the reverb characteristics. Although bulky by modern standards,
plate reverb units did not require the space
needed by a chamber reverb. Plate reverbs function by attaching an electrical transducer to the
center of a thin plate of sheet metal suspended
by springs inside a soundproof enclosure. An
adjustable damping plate allows control of the
reverb decay time and piezoelectric pickups attached to the plate provide the return reverb sig-
nal to the console. An alternative and less expensive analog reverb system is the spring
reverb, most commonly seen in guitar amplifiers beginning in the 1960s. Similar to the plate
reverb in operation, the spring reverb uses a
transducer to feed the signal into a coiled steel
spring and create vibrations. These are then
captured via a pickup and fed back into an amplifier.
Since the advent of digital audio technology in
the 1980s, artificial reverberation has been created primarily by digital algorithms that crudely
mimic the physics of natural reverb spaces by
using multiple delay lines with feedBack. Digital
“synthetic” reverb units offer a new level of realism and control unavailable with older analog
reverb systems, but still fall short of the actual
reverb created by a real space.
Components of Reverb
Reverberation sound in a normal space usually
has several components. For example, the sound
of a single hand clap in a large cathedral will
have the following distinct parts.Initially, the
direct sound of the hand clap is heard first, as it
travels from the hand directly to the ear which is
the shortest path. After the direct sound, the
first component of reverb heard by a listener is
reflected sound from the walls, floor and ceiling
of the cathedral. The timing of each reflection
will vary on the size of the room, but they will always arrive after the direct sound. For example,
the reflection from the floor will typically occur
first, followed typically by the ceiling and the
walls. The initial reflections are known as early
reflections, and are a function of the reflective
surfaces, the position of the audio source and
the relative location of the listener.
Chapter 32: TL Space TDM and TL Space Native
191
A small room may have only a fraction of a second before the first reflections, whereas large
spaces may take much longer. The elapsed time
of the early reflections defines the perceived size
of the room from the point of view of a listener.
TL Space offers various controls over early reflection parameters.
The time delay between the direct sound and the
first reflection is usually known as Pre Delay. TL
Space lets you adjust Pre Delay. Increasing the
Pre Delay will often change the perceived clarity
of audio such as vocals.
Reflections continue as the audio reaches other
surfaces in a space, and they create more reflections as the sound waves intermingle with one
another, becoming denser and changing in character depending on the properties of the room.
As the room absorbs the energy of the sound
waves, the reverb gradually dies away. This is
known as the reverb tail and may last anywhere
up to a minute in the very largest of spaces.
The reverb tail will often vary at different frequencies depending on the space. Cavernous
spaces often produce a booming, bassy reverb
whereas other spaces may have reverb tails
which taper off to primarily high frequencies.
TL Space allows for equalization of the frequencies of the reverb tail in order to adjust the tonal
characteristics of the reverb sound.
A reverb tail is often described by the time it
takes for the sound pressure level of the reverb
to decay 60 decibels below the direct sound and
is known as RT60. Overall, TL Space allows decay to be adjusted as required. For surround
processing, decay can be adjusted for individual
channel groups.
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TL Space Convolution Reverb
Convolution reverb goes beyond traditional analog and synthetic digital reverb techniques to
directly model the reverb response of an actual
reverb space. First, an impulse response (IR) is
taken of an actual physical space or a traditional
reverb unit. An IR can be captured in mono, stereo, surround, or any combination. The IR, as
displayed by TL Space, clearly shows the early
reflections and the long decay of the reverb tail.
Impulse Response sample
TL Space uses a set of mathematical functions to
convolve an audio signal with the IR, creating a
reverb effect directly modeled on the sampled
reverb space. By using non-reverb impulse responses, TL Space expands from reverb applications to a general sound design tool useful for
many types of audio processing.
The downside of traditional software based convolution reverbs is the heavy CPU processing requirement. This has often resulted in earlier
convolution reverbs with unacceptable latency.
Many early software convolution reverbs did not
offer adequate control over traditional reverb
parameters such as Pre Delay, EQ, or decay time.
TL Space redefines reverb processing in
Pro Tools by offering zero and low latency convolution with the full set of controls provided by
traditional synthetic reverbs.
TL Space System Design
TL Space uses advanced DSP algorithms to deliver convolution processing on both TDM and
native host processing.
The following figure shows the internal system
design of TL Space and demonstrates how TL
Space processes the audio signal.
The impulse computer is an internal module of
TL Space that provides extensive user control
over the currently loaded impulse response
waveform. When the user adjusts the parameters shown below, the IR is automatically recalculated by the impulse computer and reloaded
into the convolution processor.
The following figure shows the internal functions of the impulse computer as it processes the
waveform and loads it into the convolution processor.
TL Space internal system design
TL Space internal functions of the impulse computer
Chapter 32: TL Space TDM and TL Space Native
193
TL Space and System Performance
This section describes TL Space and system performance.
TL Space Supported Plug-In Formats
TL Space is available as TDM, RTAS, and AudioSuite plug-in formats depending on your Pro Tools
system and version of TL Space.
HTDM plug-ins are not supported in Pro Tools 7.0 or higher. Use the corresponding TDM or RTAS
plug-in instead.
TL Space TDM Edition includes all plug-in formats. TL Space Native Edition includes RTAS and AudioSuite plug-in formats only. The characteristics of each plug-in format, including maximum reverb
time, sample rate support, and latency are shown in the following table:
Plug-In
Format
DSP
Maximum
Reverb
Time (sec)
Maximum
Sample
Rate (khz)
Dry
Latency
(Samples)
Wet
Latency
(Samples)
TL Space
Short
TDM
HD
1.1
48 kHz
3
1029
TL Space
Medium
TDM
HD Accel
2.3
96 kHz
3
5
TL Space
Long
TDM
HD Accel
3.4
96 kHz
3
5
TL Space
RTAS
—
10.0
96 kHz
0
480
TL Space
AudioSuite
—
10.0
96 kHz
—
—
Latency and TL Space
Latency is a function of how Pro Tools processes audio and is typically measured in samples. The latency of each different mode of TL Space is shown in the table below. Latency is displayed in the Mix
window for each track in Pro Tools TDM using Delay Compensation view.
Near zero latency on HD Accel is ideal for recording live, as TL Space latency is kept to five samples
or less. RTAS plug-ins have more inherent latency. However, for some users latency is not critical and
RTAS plug-ins may lend themselves to post production environments with a requirement to switch
seamlessly in real time between reverb snapshots.
Regardless of the plug-in format, Pro Tools TDM 6.4 or higher can compensate for any latency automatically on playback using Pro Tools Delay Compensation.
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Audio Plug-Ins Guide
TL Space Channel Format Support
TL Space supports a variety of channel formats depending on your Pro Tools system, including mono,
stereo, quad, and 5.0 channels. The following table outlines channel support in specific modes.
True Stereo at 96 kHz is only available in TL Space Long.
Stereo processing is available in both summed stereo and true stereo. Summed stereo processing uses
the traditional reverb technique of summing the two input channels into a single channel that is processed by the reverb. The stereo image of the input is not reproduced in the reverb. Instead, the reverb
processes the input as if it is from a single audio source positioned in the center. An IR used for
summed stereo processing would have a single sound input source and multiple sound outputs.
True stereo processing processes two separate input signals. This stereo image of the two inputs is reproduced in the reverb. An IR used for true stereo requires two sound sources, and hence the total
number of channels in the IR will be equal to double the number of outputs. True stereo is more CPU
and DSP intensive than summed stereo, consuming twice the resources.
To use true stereo with TL Space on TDM, insert TL Space in true stereo. Stereo RTAS TL Space automatically switches between summed and true stereo modes depending on the IR loaded.
The following table shows TL Space channel formats.
Mono Input
Plug-In
Forma
t
True Stereo
Input
Stereo Input
Mono
Mono
to
Stere
o
Mono
to
Quad
Mono
to 5.0
Stere
o to
Stere
o
Stere
o to
Quad
Stere
o to
5.0
True
Stere
o to
Stere
o
True
Stere
o to
Quad
TL
Space
Short
TDM
Y
Y
Y
Y
Y
Y
Y
Y
Y
TL
Space
Medium
TDM
Y
Y
Y
—
Y
Y
—
Y
Y
TL
Space
Long
TDM
Y
Y
Y
Y
Y
Y
Y
Y
Y
TL
Space
RTAS
Y
Y
Y
Y
Y
Y
Y
—
—
TL
Space
AS
Y
—
—
—
Y
—
—
Y
—
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195
TL Space DSP Usage
TDM Systems
On Pro Tools HD and HD Accel systems,
TL Space can be instantiated as TL Space Short,
Medium and Long. The plug-in name displayed
in the menu refers to the maximum reverb time
as shown in the table below.
The different versions of TL Space have different DSP usage requirements. A Pro Tools HD
card contains nine identical DSP chips. A Pro
Tools HD Accel card contains nine DSP chips,
four of which offer external SRAM. In some
modes, TL Space requires Accel chips with external SRAM. The following table shows the TL
Space DSP requirements by reverb time.
The number of DSP chips required is a function
of the number of inputs and outputs, and the
type of processing in use. The maximum chip
usage is 8 DSP chips across two HD Accel cards.
The following table shows the TL Space DSP requirements by channel.
Input
Output
Maximum
Number of
DSP Chips
Mono
Mono
1
Stereo
2
Quad
4
5.0
5
Stereo
2
Stereo
Plug-In
Format
DSP
Quad
4
TL Space Short
TDM
Any HD DSP chip
5.0
5
TL Space Medium
TDM
Any HD Accel
chip with external
SRAM
Stereo
4
Quad
8
TL Space Long
TDM
Any HD Accel
chip with external
SRAM
True Stereo
These numbers represent the maximum possible
DSP usage of TL Space Long. For example, TL
Space Medium has only 50% of the DSP requirement in supported stereo and quad channel formats.
CPU Usage
On all Pro Tools systems, TL Space can be instantiated as an RTAS plug-in. This impacts the
performance of the CPU. CPU usage can be monitored in the System Usage window.
To optimize performance of TL Space for
RTAS processing, set the Hardware Buffer
Size in the Playback Engine to 512 samples.
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Impulse Response (IR) and
TL Space
This section covers aspects of impulse response
(IR) and TL Space.
IR Processing Control Lag
Adjusting some controls in TL Space requires
the impulse computer to recalculate the waveform and reload it into the convolution processor. This operation uses DSP and host processing capacity. When this occurs, some control lag
may be experienced. This should be kept in
mind if controls are being automated in real
time during a session.
How Impulse Responses Are Captured
An IR of an actual physical space is captured using a combination of an impulse sound source
and capture microphones. The sound source is
used to excite the physical space to create a reverb, and can be a starter pistol or a frequency
tone played through a speaker. The microphones can be placed in various configurations.
The resulting IR is then processed to create a
digital representation of both the physical
space, potentially colored by the sound source
and the type of microphone used. Likewise, an
IR can be captured of effects hardware, such as
analog reverbs, by sending a test pulse through
the unit and capturing the result digitally. In addition to reflecting reverb or delay characteristics, an IR also reflects tonal character and can
be used for a variety of effects beyond pure reverb applications.
Multiple IRs may be taken of a physical space
where the sound source has been moved to physical locations. Each resulting IR may be used to
create individual reverbs for separate instruments. This effectively allows an engineer to
place each instrument in the reverb sound field
as if the instruments were physically arranged in
the space.
TL Space IR Library Installation
If you purchased the boxed version of TL Space,
it includes an installer disc of the standard
TL Space IR Library. If you purchased TL Space
online, you will need to download IR Libraries
from Avid’s TL Space Online IR Library. For
more information on downloading and installing IR Libraries from the TL Space Online IR Library, see “Installing TL Space IR Packages” on
page 207.
To install the TL Space IR Library from disc:
1 Insert the correct TL Space IR library installer
disc for your operating system (Windows or
Macintosh) in your computer’s CD/DVD drive.
Double-click the TL Space IR library installer
application to launch it. Read the license agreement. If you agree to the terms, click Accept.
2
3 Click Install to perform an easy install of the
entire IR library on the system drive.
If you want to install only part of the library,
select Custom Install and select the parts of the library you want to install.
4
When the installation is completed, click Quit
to finish the installation.
5
Depending on the capture technique used, the
IR may be suitable for use with mono, stereo,
surround or a combination of those formats. For
example, a capture setup with a single sound
source and two microphones is ideal for a mono
to stereo IR.
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197
Using Third-Party IRs in
TL Space
TL Space Multichannel IR
Formats
TL Space reads a wide range of IR formats automatically, including WAV, SDII, and AIFF file
formats, allowing you to import a variety of IRs.
TL Space supports IR sample rates from 22 kHz
up to 96 kHz in bit depths from 16 to 32 bits. In
addition, TL Space supports the display of JPEG
format picture files stored with IRs.
TL Space supports IRs in multichannel or multiple mono audio files. IRs with a single input are
used for mono or summed stereo processing and
can be stored as a single interleaved multichannel file, or as multi-mono files. IRs with stereo
inputs used for true stereo processing must be
stored as multi-mono files.
To use third-party IR libraries with TL Space:
The following table shows TL Space IR channel
formats.
1
In the IR Browser, select Edit > Import Other IR
Folder.
Locate and select the library on your hard
drive.
Input
Output
Channel
Order
File Format
Mono
Mono
—
Mono file
Mono
Stereo
LR
One 2-channel
file or two
mono files
Mono
Quad
L R Ls Rs
One 4-channel
file or four
mono files
Mono
5.0
L C R Ls
Rs
One 5-channel
file or five
mono files
Stereo
Stereo
LR
Four mono files
Stereo
Quad
L R Ls Rs
Eight mono
files
Stereo
5.0
L C R Ls
Rs
Ten mono files
2
3
Click Choose.
TL Space will add the new library to the IR
browser.
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Audio Plug-Ins Guide
For multi-mono files, TL Space understands the
following filename conventions, based on those
used by Pro Tools. The filename format is based
on the impulse name plus two suffixes which indicate input and output channels as follows:
Impulsename.inputchannel.outputchannel.type
• Impulsename is the name of the impulse.
Mixing multiple IR files with the same Impulsename in the same folder is not supported.
The following examples show how various
multi-mono IR files could be named.
Stereo to Stereo IR
Cathedral.1.L.wav
Cathedral.1.R.wav
Cathedral.2.L.wav
Cathedral.2.R.wav
Stereo to 5.0 IR
Cathedral.1.L.wav
• Inputchannel refers to the number of
sources used for the impulse, starting at the
number 1. An IR captured in true stereo
will usually have two input channels numbered 1 and 2. If there is only one input
channel, then inputchannel is optional and
can be omitted. Also, instead of using numbers 1 and 2, the inputchannel can be designated as L and R.
Cathedral.1.C.wav
• Outputchannel refers to the microphones
used to capture the impulse, and corresponds to your studio monitors. outputchannel is designated using the
standard L, C, R, Ls and Rs extensions.
Cathedral.2.Rs.wav
• Type is optionally .WAV, .AIFF or .SD2. For
best performance, filenames should always
be suffixed with type to avoid TL Space having to open the file to determine audio format.
Cathedral.1.R.wav
Cathedral.1.Ls.wav
Cathedral.1.Rs.wav
Cathedral.2.L.wav
Cathedral.2.C.wav
Cathedral.2.R.wav
Cathedral.2.Ls.wav
Mono to Quad IR
Cathedral.L.wav
Cathedral.R.wav
Cathedral.Ls.wav
Cathedral.Rs.wav
Stereo to quad IR
Cathedral.1.L.wav
Cathedral.1.R.wav
Cathedral.1.Ls.wav
Cathedral.1.Rs.wav
Cathedral.2.L.wav
Cathedral.2.R.wav
Cathedral.2.Ls.wav
Cathedral.2.Rs.wav
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199
Channel Compatibility and TL
Space
TL Space works best with IRs that match your
current channel configuration. For example, if
TL Space is instantiated in a mono to stereo configuration, stereo IRs will be highlighted in the
IR browser. The IR information displayed in the
display area shows how many inputs and outputs an IR has. For example, an IR listed as 2 input 4 output is a stereo to quad IR.
If an IR is loaded that doesn’t match the current
configuration, TL Space will try to create the
best possible match with the IR provided. For
example, if a stereo IR is loaded into a mono instantiation of TL Space, TL Space will sum the
left and right channels in order to mimic a stereo reverb with both channels panned to mono.
If an IR is loaded that is missing a required
channel, TL Space will automatically create a
phantom channel for the IR if needed. For example, if a stereo IR is loaded into a quad instantiation, TL Space will compute left and right surround channels automatically based on the
existing channels. If a quad IR is loaded into a
5.0 channel instantiation, TL Space will compute a phantom center from the front left and
right channels. Phantom channels are indicated
by comparing the IR information displayed in
the display area to the number of channels in
use. For example, a 2 input 4 output IR used with
a 5.0 output instantiation of TL Space will automatically have a phantom center channel created.
200
Audio Plug-Ins Guide
TL Space Presets
TL Space supports the Pro Tools Plug-In Librarian. When an IR file is loaded, all controls remain at their current positions as the IR file only
contains the audio waveform. By default, presets
contain both the IR waveform and control settings and can be saved as required so that specific control settings can be retained for future
sessions. If you save presets without embedding
the IR waveform, be sure that you include the IR
waveform with the session when transferring
the session between different Pro Tools systems.
There are two important items to note about using presets in TL Space:
• TL Space presets do not store information
for the Wet and Dry level controls. This is
to enable you to change presets without losing level information. Likewise, the
Pro Tools Compare function is not enabled
for these controls.
• A TL Space preset only includes the currently selected snapshot.
IR files are audio files only and do not contain information about TL Space control
settings. If you wish to save specific control
settings for an IR, you should save them using the Pro Tools Plug-In Librarian or using
the snapshot facility of TL Space.
TL Space Snapshots
In addition to presets, TL Space lets you manage
a group of settings, called snapshots, that can be
switched quickly using a single, automatable
control. Each snapshot contains a separate IR
and settings for all TL Space controls.
IRs in a snapshot have been pre-processed by
the impulse computer and can be loaded instantly into the convolution processor. With
RTAS, switching between snapshots does not
cause audio to drop out. Snapshots are useful,
for example, in post production mixes when the
reverb is changed for different scenes via automation as the picture moves from one scene to
another.
Embedding IRs in Sessions, Presets, and
Snapshots
By default, all IR and snapshot info used by TL
Space (including up to ten IRs) is saved in the
Pro Tools session file. Likewise, plug-in presets
contain a saved copy of the IR and settings in the
currently selected snapshot. Session and preset
file sizes will increase as TL Space stores each IR
waveform inside the file. This provides maximum compatibility between different Pro Tools
systems without the need for them to have identical IR libraries.
IR embedding can be disabled in TL Space’s
Preferences. If IR embedding is disabled, TL
Space stores only a reference to the name of the
IR file. When the session is transferred to a different system, TL Space attempts to load the
matching IR file from the TL Space IR library.
For maximum compatibility, ensure that all of
the appropriate IR files are available on the new
system.
When working with an IR that only exists in a
session file, ensure it is saved to a separate snapshot or preset. If the IR is overwritten by loading
a new IR and the session is saved, the original IR
cannot be recovered without access to the original IR file.
By default, Pro Tools presets or session files
created using TL Space automatically include copies of all relevant IR waveforms.
This provides maximum compatibility of
session files between different Pro Tools systems.
It is your responsibility to ensure that you
observe the copyright on any IR transferred
to a third party in this fashion.
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201
TL Space Controls and Displays
The TL Space interface is divided into the following sections:
• Display area (See “TL Space Display Area” on page 203.)
• IR Browser (See “TL Space IR Browser” on page 206.)
• Primary controls (See “TL Space Primary Controls” on page 208.)
• Group Selectors and Controls (See “TL Space Group Selectors and Controls” on page 209.)
The TL Space interface
202
Audio Plug-Ins Guide
TL Space Display Area
The display area of TL Space operates in the following four modes, indicated by the Display
Mode selectors at the top right hand corner of
the TL Space window:
• Waveform mode
• Picture Preview mode
• Snapshot mode
• Preferences mode
Display Mode selectors
The Display area changes based on the selected
mode.
Info Bar
At all times, the Info bar at the bottom of the
display area window shows the following controls and information.
IR is loaded (for example, the IR in use has been
loaded from a preset or session but does not exist in the IR browser), the Quick browser controls are inoperative.
TL Space Waveform Mode
Waveform mode is selected using the Waveform
icon at the top of the TL Space window. In Waveform mode, the display area shows the IR waveform with the following controls.
Waveform mode displays the IR waveform along
a horizontal axis marked in seconds and the vertical axis marked in amplitude. The early section
of the waveform is highlighted in a lighter color.
In addition, the channel selector highlights the
current channel in the waveform.
IR information such as sample rate and number
of input and output channels is displayed at the
bottom right of the waveform.
Info bar
Snapshot Menu A pop-up menu allowing quick
selection or automation of a snapshot.
Display area, Waveform mode
IR Name Displays the folder and file name of the
currently loaded IR.
The controls in Waveform mode function as follows:
Quick Browser Controls The Quick browser con-
Original Bypasses all waveform processing, al-
trols allow the IR to be quickly changed even
when the IR browser is closed, automatically
loading each IR sequentially. The Waveform
icons step backwards and forwards through IRs
and automatically load the IR file. The Folder
icons step backwards and forwards through
folders. The Quick browser requires an IR to be
currently loaded from the IR browser. If no such
lowing the original IR to be auditioned. This
control effectively bypasses the processing in
the IR computer as shown in the system diagram.
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203
Channel Selectors Displays from one to five
channels (in the order Left, Center, Right, Left
Surround, Right Surround). Click the desired
channel to display the IR waveform for that
channel. In Mono mode, no channel selector is
displayed.
Zoom Zooms in and out on the time axis for the
waveform display.
TL Space Picture Preview Mode
Picture Preview mode is selected using the Picture Preview icon at the top of the TL Space window. When selected, Picture Preview mode
shows pictures associated with the IR. For an IR
provided with TL Space, this will usually include
a photograph of the location, and an image with
technical details such as microphones used or an
overview of the microphone setup. Thumbnails
of images are displayed in the right hand column. In this mode, the IR browser can be used to
view the associated pictures without loading the
IR itself.
considerably faster than loading a new IR. Snapshot mode allows all ten snapshots to be viewed
as well as the option to select, rename, copy,
paste, and clear snapshots.
The name of the currently selected snapshot is
always displayed in the Info bar at the bottom of
the display area, and can be automated. This lets
you switch reverb settings during playback and
is useful for post production sessions where the
reverb setting may change as the scene changes.
Display area, Snapshot mode
The active snapshot can be selected in one of two
ways. At any time, a snapshot can be selected by
using the snapshot menu in the Info bar. Alternatively, when the display area is in Snapshot
mode, a snapshot can be selected by clicking the
selection area next to the snapshot name.
Select Lets you select which snapshot is currently loaded.
Display area, Picture Preview mode
TL Space Snapshot Mode
Snapshot mode is selected using the Snapshot
icon at the top of the TL Space window. TL Space
provides ten snapshots available at all times.
Each snapshot stores a separate IR waveform
and all control settings. Snapshots are optimized for quick loading into the convolution
processor, and switching between snapshots is
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Audio Plug-Ins Guide
Name Displays the name of each snapshot. By
default, snapshots are named “Snapshot 1”
through “Snapshot 10.” Snapshots can be renamed by clicking on the snapshot name and entering a new name followed by the Enter key
(Windows) or the Return key (Macintosh).
Sample Path Displays the name of the IR selected for each snapshot.
Copy Copies the currently selected snapshot settings into a clipboard.
Paste Pastes the clipboard into the currently se-
lected snapshot. Note that the name of the existing snapshot is not changed by pasting a new
snapshot, in order to avoid duplicate snapshot
names.
Clear Clears the IR from the currently selected
snapshot.
TL Space Preferences Mode
Preferences mode is selected using the Preferences icon at the top of the TL Space window.
This displays a number of preferences settings
for TL Space.
TL Space Meters
The Meters display the amplitude of the incoming and outgoing audio signals by channel. The
number of meters shown will depend on the
number of input and output channels. Input meters may be mono or stereo, and output meters
may be mono, stereo, quad, or 5.0 channels.
Each meter is marked as either mono, left, right,
center, left surround, or right surround. A logarithmic scale marked in decibels and momentary peaks are also displayed on the meter.
Display area, Preferences mode
Embed IRs in Preset & Session Files Enables or
disables the embedding of IR waveforms in presets and session file. By default, this is enabled.
PCI Throttle Increasing the PCI throttle control
reduces PCI contention for Pro Tools systems
when using PCI video capture hardware. For
more information, see “PCI Bus Contention and
TL Space” on page 213.
For most users, this control should not be adjusted. This control is only displayed for TDM
instantiations of TL Space on Pro Tools|24 Mix
and Pro Tools|HD systems.
Meters, stereo input to 5.0 output shown
The red Clip indicator indicates that audio for
that channel has exceeded 0 dB in amplitude.
When a channel has clipped once, the clip indicator remains lit and additional clips will be
shown by a variation in the color of the indicator. The clip indicator for all channels can be
cleared by clicking on any clip indicator, or selecting the Pro Tools Clear All Clip Indicators
command.
The meters do not function when TL Space is
used as an AudioSuite plug-in.
Installed IR Packages Displays a list of installed
TL Space IR packages and their versions.
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205
TL Space IR Browser
The TR Browser icon at the top right hand corner of the TL Space window opens the IR
browser. By default, TL Space will display a single IR group for the TL Space library.
The IR browser can be operated using the following shortcuts. When the IR browser has keyboard focus, a blue highlight is displayed around
the edge of the browser window.
The following table shows IR browser keyboard
shortcuts.
The IR browser lets you quickly and easily install, locate, and organize IRs on local hard
drives. The Load and Edit buttons in the IR
browser let you install and import IRs, create
Favorites, and change the IR groups displayed.
TL Space automatically highlights each IR that
matches the current channel configuration. For
example, when using a TL Space Stereo to Quad
inset, each IR with that configuration is highlighted. Impulses that are not highlighted can
still be loaded, and TL Space tries to adapt the IR
to the current channel format (see “Channel
Compatibility and TL Space” on page 200).
Browser
Navigation
Arrow Keys
Load IR
Enter (Windows)
Return (Macintosh)
Open/close
all folders
Alt-click (Windows)
Option-click (Macintosh)
Edit menu
Right-click (Windows or Macintosh)
Control-click (Macintosh)
Return keyboard focus
to Pro Tools
Escape key
The IR browser lets you install and import new
IRs. Each IR folder reflects a folder on the hard
drive. When importing a new IR folder, a standard file dialog will be displayed to enable the
user to choose the folder that contains the desired IR.
The IR browser also provides a Favorites folder,
which is a user defined group of links to IRs in
the IR browser. Favorites can be sorted in any
desired order by dragging and dropping them as
required. In addition, folders can be created in
Favorites using the ‘New Folder in Favorites’
function in the Edit menu.
IR Browser
An IR can be loaded by double clicking with the
mouse, or using the Load button displayed at the
top of the IR browser drawer. The currently
loaded IR is highlighted with a small dot next to
the file name in the browser.
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Audio Plug-Ins Guide
To add an IR file or folder to the Favorites folder:
1 In the IR browser, select the desired IR file or
folder.
2
From the IR browser’s Edit menu, select Add to
Favorites.
TL Space IR Browser Edit Menu
The IR browser’s Edit menu contains the following commands:
Download TL Space IR Package Opens a Web
browser to the TL Space online IR library.
Rescan for Files Forces TL Space to check the
hard drive for new IRs. This is typically required
if new IR files have been copied to the hard
drive. Using the Rescan for Files command loads
new IRs into TL Space without needing to close
TL Space or the Pro Tools session.
TL Space may pause briefly while it scans
the hard drives to locate IRs or if all folders
are opened at once. The amount of time
taken is proportional to the number of folders and IRs scanned.
Install TL Space IR Package Installs a new IR
package downloaded from the TL Space online
library (see “Installing TL Space IR Packages”
on page 207).
Import Other IR Folder Lets you import a new IR
folder in common file formats. By default, the
new IR is given the same name as the selected
folder.
Remove Imported IR Folder Lets you remove the
currently selected IR folder.
Rename Imported IR Folder Lets you rename the
currently selected IR folder.
Add to Favorites Adds the currently selected IR
to the Favorites group at the top of the browser
window.
New Folder in Favorites Creates a folder in the
Favorites group. Favorite IRs can be dragged
and dropped into the folder.
Rename Favorites Folder Lets you rename the
Installing TL Space IR Packages
Additional IR packages for TL Space are available for registered users to download from the
TL Space Online IR Library at:
www.avid.com/tlspace/impulselibrary/
These package files are supplied in a lossless
compressed format.
To install a TL Space IR package:
In the TL Space IR browser, select Download IR
Package from the Edit menu. Your default Web
1
browser launches and loads the Avid TL Space
Online IR Library website (www.avid.com/tlspace/impulselibrary/).
2
currently selected Favorites folder.
Click Download
Login using your email address and password.
You may need to create a new account if you have
not yet registered TL Space.
3
Remove from Favorites Removes the currently
selected IR from the Favorites group. This function only removes the link in the Favorites
group and does not remove the original IR file
from the system.
Reset to Default IR Library Resets TL Space to
the default library. This also removes any user
imported IR folder, but does not affect the Favorites folder, or IR packages installed from the
TL Space online IR library.
To download IR packages from the TL Space
Online IR Library, you must first register
with Avid and create an online profile.
4
Click Continue.
5
Click Download for the IR package you want.
6 In TL Space, select Install TL Space IR Package
from the Edit menu.
Chapter 32: TL Space TDM and TL Space Native
207
In the resulting dialog, locate and select the
file you downloaded.
7
8
Click Choose.
TL Space will display a summary of the IR package with a short description, copyright statement, and a list of the contents.
TL Space Primary Controls
The primary control group is visible at all times
and allows control of key reverb parameters.
This includes the wet and dry levels of the audio
passing through TL Space.
9 Click Install to install the IR package. A window is displayed with the results of the installation.
The IR browser in TL Space updates to include
the new IR.
If a problem occurs with the IR installation, TL
Space displays an error message. Review the log
file stored in the TL Space IR library for further
details. Each IR package has a version number,
and TL Space warns you if an IR package has already been installed.
The details of all installed IR packages can be reviewed using the Show Packages option in Preferences mode.
TL Space primary controls
Reset Resets all TL Space parameters except
Wet, Dry, and Input and Output Level.
Wet Controls the level of wet or effected reverb
signal, from –inf dB to +12 dB.
Dry Controls the level of dry or unaffected re-
verb signal, from –inf dB to +12 dB.
Decay Controls the overall decay of the IR waveform and is displayed as a percentage of the
original. When Decay is adjusted, the waveform
is recalculated in real time.
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Audio Plug-Ins Guide
TL Space Group Selectors
and Controls
TL Space presents reverb controls in five different groups. Each group is activated by selecting
the corresponding selector.
TL Space Delay Controls
The Delays group allows control of delay timings for the reverb. When changes are made to
any control in the Delays group, the IR waveform is recalculated and displayed in the Waveform display.
Pre Delay Adjusts length of the Pre Delay from
Group Selectors
TL Space Level Controls
The Levels group provides control of the overall
input and output of the reverb, including individual controls for early and late reflections,
and independent front, rear, and center levels
for surround outputs.
Input Cuts or boosts the input signal level from
–inf dB to +12 dB.
Output Cuts or boosts the output signal level
from –inf dB to +12 dB.
Early Cuts or boosts the levels of the early reflections from –inf dB to +12 dB.
Late Cuts or boosts the levels of the late reflections from –inf dB to +12 dB.
Front/Rear/Center In quad and 5.0 channel out-
put modes, Cuts or boosts the front, rear, and
center signal levels from –inf dB to +12 dB. In
5.0 output mode, the level of the Center channel
is affected by both the Front and Center controls.
–200 to +200 ms. The Pre Delay is the time between the direct sound and the first reflection.
Increasing the Pre Delay often changes the perceived clarity of audio such as vocals. Pre Delay
adjusts the delay of the overall impulse and affects both the Early and Late portions of the IR
equally.
Pre Delay can be set to negative values to allow
for subtle or radical changes to the reverb. For
example, a small negative Pre Delay setting can
be used to eliminate the early portion of an IR. A
large negative Pre Delay setting lets you use the
very end of a reverb tail for creative sounds not
possible with standard reverbs.
Late Delay Adjusts length of the Late Delay from
zero to +200 ms. The Late Delay is the time between the Early Reflections and the Late Reflections or tail of the reverb.
Increasing the Late Delay control from zero allows the reverb tail to be delayed so that it does
not start immediately after the early portion of
the IR. As Late Delay is increased, the reverb tail
starts later in time and makes the reverb space
sound larger. Large amounts of late delay can be
used to achieve creative effects not possible with
standard reverbs.
Front/Rear/Center Delay In quad and 5.0 channel
output modes, adjusts length of the Front, Rear,
and Center Delays independently from zero to
+200 ms.
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209
TL Space Early Section Controls
The Early group controls the character of the
early portion of the IR and the early reflections.
The primary control is Early Length which defines the size of the early portion of the IR waveform. When loading an IR from an audio file,
TL Space relies on the user to define which part
of the IR is the early portion of the waveform. By
default, the Early length is set to 20 ms.
The early portion of the IR waveform is highlighted in the Waveform display. If Early length
is set to zero, then the Early setting have no effect on the audio. Otherwise, when changes are
made to any control in the Early group, the IR
waveform is recalculated and displayed in the
Waveform display.
Length Adjusts the length of the Early reflec-
tions from zero to 500 ms. When set to zero,
other controls in the Early group have no effect
on the audio. The Early Length control adjusts
the point in the impulse where the early portion
ends and the late portion or tail begins.
For the most realistic reverb results, Early
Length should be adjusted while viewing the
waveform display. The early portion of a reverb
IR is typically seen as a series of discrete spikes
at the beginning of the waveform. Early Length
can however be adjusted to any value to explore
other creative possibilities.
Size Changes the size of the Early reflections,
from 50% to 200%. Early Size expands or contracts the reflections in the early portion of the
IR (as specified by the Early Length control). Reduce the Early Size to give the space a smaller,
tighter sound. Increase the Early Size to give the
space a larger, roomier sound.
210
Audio Plug-Ins Guide
Lo Cut Early Lo Cut controls the frequency of a
highpass filter applied to the early portion of the
IR (as specified by the Early Length control).
The default setting of zero disables the highpass
filter. As the control is set to a higher value, the
corner frequency of the highpass filter is increased. Use this control to reduce boom and
low frequency cancellations that can happen
when mixing the reverb output with a dry signal.
Balance Early Balance controls the left/right
gain balance of the early portion of the IR (as
specified by the Early Length control). Adjust
the Balance to control the apparent position of
the reverb input in the stereo image. A negative
value reduces the right channel gain. A positive
value reduces the left channel gain.
When loading an IR from an audio file, TL
Space relies on the user to define which part
of the IR is the early portion of the waveform. If the Early Length is set to zero, controls in the Early group will not affect the
IR.
TL Space Reverb Section
Controls
The Reverb group offers a low and high shelf EQ
in addition to width and balance controls. The
EQ operates prior to convolution processing.
Lo Freq Adjusts the frequency of a low frequency filter from 20 to 500 Hz.
Lo Gain Cuts or boosts the frequency set in Lo
Freq from –15 dB to +15 dB.
Hi Freq Adjusts the frequency of a high frequency filter from 500 Hz to 20 kHz.
Hi Gain Cuts or boosts the frequency set in Hi
Freq from –15 dB to +15 dB.
Width Increase or reduces the stereo spacious-
High Xover Adjusts the frequency point that di-
ness of the reverb. Use this control to tailor the
reverb’s character in a mix. Keep in mind that an
IR that has little stereo separation to begin with
may have limited results.
vides the IR into mid and high frequency portions.
Balance Controls the balance of the reverb out-
put. Use this control to balance a reverb from an
IR that has been captured without a centered
stereo image, or for creatively controlling the
character of the reverb in a mix.
Reverse Reverses the IR waveform and controls
the total length. As the IR waveform is recalculated, it is re-displayed in the Waveform display.
The value shown is measured in Beats Per Minute to let you easily match the tempo of the music.
High Decreases or increases the rate at which
high frequencies decay.
Front/Rear In quad and 5.0 channel output
modes, Front and Rear independently control
the decay for front and rear channels.
TL Space Info Screen
Click the Trillium Lane Labs logo to view the
Info screen. The Info screen displays copyright
and version information.
If the waveform is reversed using the Reverse control, effected audio may continue
to play for several seconds after the transport is stopped or audio input finishes.
TL Space Decay Section
Controls
The Decay group controls allow the user to control the decay of the low, mid, and high frequency portions of the IR. Use the controls to
tailor the reverb’s character for a mix or for creative possibilities not found in traditional reverb processors.
Low Decreases or increases the rate at which low
frequencies decay.
Low Xover Adjusts the frequency point that di-
vides the IR into low and mid frequency portions.
Mid Decreases or increases the rate at which mid
frequencies decay.
Chapter 32: TL Space TDM and TL Space Native
211
Using TL Space
This section addresses some common scenarios in which TL Space can be used during a Pro Tools session.
TL Space Plug-In Formats
TL Space is available in TDM, RTAS, and AudioSuite plug-in formats. The following table provides
some general recommendations for use of TL Space based on the advantages and disadvantages of
each plug-in format. The following table shows the pros and cons for different plug-in formats:
Plug-In
Format
Pros
Cons
Typical Use
TDM
Zero latency on HD Accel
Minimal CPU load
Very fast waveform manipulation
DSP Usage
Max 3.4 second reverb tail
Audio pause during snapshot
switching
Mixing, live recording,
post production
RTAS
Seamless Snapshot switching
Very long reverb tails
CPU load
RTAS latency
Pro Tools host-based
systems
AudioSuite
Low CPU load
Non-real-time
No surround support
Using TL Space Presets
TL Space ships with a selection of factory presets
for different reverb sounds. The presets are designed to give a sample of the various IRs available from the Plug-In Presets selector in conjunction with various reverb settings. However,
the presets do not cover the entire IR library.
1
Insert TL Space on a track.
2
Select Snapshot mode.
4
Name each Snapshot as desired.
Using TL Space on an Effect
Send
5
Click Auto.
6
Add Snapshot to the list of automated controls.
When TL Space is used on an Aux Input track as
an effects send, the Dry control should be set to
–inf dB.
7
Automating TL Space Snapshots
9
Snapshot automation is a powerful method of
changing the reverb parameters without having
to individually automate each parameter.
212
To automate TL Space Snapshots:
Audio Plug-Ins Guide
Load an IR into each Snapshot and make any
desired changes to specific TL Space controls.
3
Select TL Space > Snapshot from the automation menu for the track.
8
Select the Pencil tool.
Draw the desired automation. The names displayed in the automation track will match the
names entered for each Snapshot.
PCI Bus Contention and TL
Space
Large Pro Tools TDM systems running TL Space
TDM in conjunction with video capture and
playback or other PCI cards may encounter
–6042 errors. These errors are caused when the
Pro Tools DAE engine cannot transfer audio
track data from the computer to the Pro Tools
card over the PCI bus quickly enough. The error
typically occurs when TL Space attempts to use
the PCI bus to load impulses. PCI bus contention can be addressed with the following steps.
First, you may wish to locate more demanding
PCI cards on the main PCI bus rather than in an
expansion chassis. By locating the PCI cards
away from Pro Tools DSP cards, PCI contention
is typically reduced.
Secondly, assign more DSPs to the Pro Tools
Playback Engine. Open the Playback Engine dialog and increase the number of DSPs per the
Number of Voices.
The PCI Throttle control can be adjusted in
Preferences mode. Settings take effect immediately across all instances of TL Space. This control offers the settings shown in the following
table.
Setting
Effect
Off
No PCI throttle control—
maximum PCI contention
33%
Default setting for Macintosh G5
systems
66%
Default setting for Windows XP
systems
100%
Maximum PCI throttle—minimum
PCI contention
Increasing the PCI throttle control will reduce
TL Space performance as PCI activity is reduced.
For most users, the PCI throttle control provides optimum performance at the default
setting and should not be adjusted.
If this does not resolve bus contention issues,
the PCI Throttle control can be adjusted upwards one step at a time until the –6042 errors
stop. For example, the default setting for a Macintosh G5 system is 33% and it can be increased
in two steps to 100% until the bus contention is
resolved. As more PCI throttling is used,
TL Space will take longer to update the data on
the DSP chip(s) running TL Space.
Chapter 32: TL Space TDM and TL Space Native
213
TL Space IR Library
TL Space includes an extensive impulse response library, divided into the following categories.
214
Category
Description
Halls
Halls and auditoriums
Churches
Churches and chapels
Rooms
Large and small rooms
Chambers
Traditional studio reverb chambers
Plates
Classic electromechanical reverb plates
Springs
Classic electromechanical reverb springs
Digital Reverbs
Classic and contemporary digital reverb units
Post Production
Post production impulses
Tiny Spaces
Small reverbs from everyday objects
Pure Spaces
A selection of Pure Space impulses in multiple categories
Effects
Non-reverb effects for sound design in multiple categories
• Colors
Sound coloring and positioning
• Cosmic
Spacey smears and washes
• Impressions
Smears and washes that evoke an image
• Industrial
Heavy machinery
• Periodic table
Better living through chemistry
Audio Plug-Ins Guide
Part VI: Delay Plug-Ins
Chapter 33: AIR Dynamic Delay
AIR Dynamic Delay is an RTAS plug-in. Use the
Dynamic Delay Plug-In for a delay line that can
synchronize to the Pro Tools session tempo and
be modulated by an Envelope follower.
Dynamic Delay Controls
The Dynamic Delay plug-in provides a variety of
controls for adjusting plug-in parameters.
Sync
When Sync is enabled, the delay time synchronizes to the Pro Tools session tempo. When Sync
is disabled, you can set the delay time in milliseconds independently of the Pro Tools
session tempo. The Sync button is lit when it is
enabled.
Delay
Dynamic Delay plug-in window
When Sync is enabled, the Delay control lets you
select a rhythmic subdivision or multiple of the
beat (based on the Pro Tools session tempo) for
the delay time.
Chapter 33: AIR Dynamic Delay
217
Select from the following rhythmic values:
• 16 (sixteenth note)
• 8T (eighth-note triplet)
• 16D (dotted sixteenth-note)
• 8 (eighth note)
• 4T (quarter-note triplet)
• 8D (dotted eighth-note)
• 4 (quarter note)
• 2T (half-note triplet)
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet (delayed) signal. At 50%, there are equal amounts of dry and
wet signal. At 0%, the output is all dry and at
100% it is all wet.
Dynamic Delay Delay Section
The delay section of the Dynamic Delay plug-in
provides L/R Ratio and Stereo Width controls.
• 4D (dotted quarter-note)
• 2 (half note)
L/R Ratio
• 1T (whole-note triplet)
• 7/4 (seven tied quarter notes)
The Left/Right Ratio control lets you set the ratio of left to right delay times. Move the control
all the way to the left (50:100) and the left channel delay time is half the right channel delay
time. Move the control all the way to the right
(100:50) the right channel delay time is half the
left channel delay time.
• 8/4 (double whole note)
Stereo Width
• 3/4 (dotted half note)
• 4/4 (whole note)
• 5/4 (five tied quarter notes)
• 6/4 (dotted whole note)
When Sync is disabled, the Time control lets you
set the delay time in milliseconds and seconds
(1 ms to 4.00 seconds).
Feedback
The Feedback control lets you adjust the amount
of delay feedback. At 0% the delayed signal repeats only once. As you increase the feedback,
the number of times the delay repeats increases.
At 100%, the delay doesn’t repeat indefinitely,
but it does last a very long time!
Note that each Delay mode produces a different
feedback pattern, especially when the L/R Ratio
control is not centred.
The Stereo Width control lets you adjust the
width of the delay effect in the stereo field.
Dynamic Delay EQ Section
The EQ section of the Dynamic Delay plug-in
provides low and high cut filters.
Low Cut
The Low Cut control lets you adjust the frequency for the Low Cut filter. For less bass, raise
the frequency.
High Cut
The High Cut control lets you adjust the frequency for the High Cut filter. For less treble,
lower the frequency.
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Audio Plug-Ins Guide
Dynamic Delay Env Mod
(Envelope Modulation) Section
The Dynamic Delay plug-in provides an Envelope follower that can control various parameters in real time.
Rate
Adjust the Rate control to determine how
quickly the Feedback and Mix parameters respond to input from the Envelope follower.
Fbk
Adjust the Feedback control to determine how
much the Envelope follower affects the Feedback
amount.
Dynamic Delay Feedback Modes
Select one of the following options for the Feedback Mode:
Mono Sums the incoming stereo signal to mono,
then offers separate left and right delay output
taps from that signal.
Stereo Processes the left and right channels of
the incoming stereo signal independently and
outputs the processed signal on the corresponding left and right channels.
Cross Processes the left and right channels of
the incoming stereo signal independently, and
feeds the each side’s delayed signal back to the
opposite channel.
Mix
Adjust the Mix control to determine how much
the Envelope follower affects the wet/dry mix.
 At 0%, the Envelope follower has no effect on
the given parameter.
 At +100%, the parameter’s value is increased
in direct proportion to the incoming signal’s
amplitude envelope.
 At –100%, the parameter’s value is decreased
in direct proportion to the incoming signal’s amplitude envelope.
Chapter 33: AIR Dynamic Delay
219
220
Audio Plug-Ins Guide
Chapter 34: AIR Multi-Delay
AIR Multi-Delay is an RTAS plug-in. Use the
Multi-Delay plug-in to apply up to six delay
lines to the audio signal.
Multi-Delay Controls
The Multi-Delay plug-in provides a variety of
controls for adjusting plug-in parameters.
Sync
When Sync is enabled, the Delay time synchronizes to the Pro Tools session tempo. When Sync
is disabled, you can set the delay time in milliseconds independently of the Pro Tools session
tempo. The Sync button is lit when it is enabled.
Delay
When Sync is enabled, the Delay control lets you
set the main delay length in 16th note lengths
(based on the Pro Tools session tempo).
Multi-Delay plug-in window
When Sync is disabled, the Time control lets you
the main delay time in milliseconds and seconds.
Feedback
The Feedback control lets you adjust the amount
of delay feedback. At 0% the delayed signal repeats only once. As you increase the feedback,
the number of times the delay repeats increases.
At 100%, the delay doesn’t repeat indefinitely,
but it does last a very long time!
Chapter 34: AIR Multi-Delay
221
From and To
The From and To controls let you feed signal
from one delay Tap to another, or back to the
main input, to create complex delay/feedback
effects.
Multi-Delay Delay Taps Controls
The Multi-Delay provides five Taps (delay
lines). Each Tap provides the same set of controls. Controls for each Tap can be edited independently of the other Taps. Each Tap provides
the following controls:
From
The From control sets the tap from which signal
will be cross-routed.
On
The On button turns the selected tap’s signal on
or off.
To
The To control sets the tap (or the main input)
that the cross-routed signal will be routed to.
If the delay time of the “To” tap is greater
than the delay time of the “From” tap, then
the result is “feed-forward” rather than
feedback, so only one delay repeat will be
heard.
Delay
Adjust the Delay control to set the length of delay for the tap, relative to the main Delay setting.
Level
Adjust the Level control to change the output
level of the Tap.
High Cut
Pan
The High Cut control lets you adjust the frequency for the High Cut filter. For less treble,
lower the frequency.
Low Cut
The Low Cut control lets you adjust the frequency for the Low Cut filter. For less bass, raise
the frequency.
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet (delayed) signal. At 50%, there are equal amounts of dry and
wet signal. At 0%, the output is all dry and at
100% it is all wet.
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Audio Plug-Ins Guide
Adjust the Pan control to pan the audio signal
from the Tap left or right in the stereo field.
Chapter 35: Mod Delay II
Mod Delay II is a set of modulating delay
plug-ins that are available in TDM, RTAS, and
AudioSuite formats.
There are six different Mod Delay II plug-ins,
capable of different maximum delay times:
• The AudioSuite only version of the Delay plugin provides up to 10.9 seconds of delay at all
sample rates.
The TDM versions of the Extra Long Delay
mono-to-stereo and stereo plug-in are not
supported at 96 kHz. All TDM versions of
the Extra Long Delay plug-in are not supported at 192 kHz. RTAS versions of the Extra Long Delay plug-in are fully supported
at all sample rates.
• The Short Delay provides 43 ms of delay at all
sample rates.
• The Slap Delay provides 171 ms of delay at all
sample rates.
Short Delay and Slap Delay do not have
Tempo, Meter, Duration, and Groove controls.
• The Medium Delay provides 341 ms of delay at
all sample rates.
• The Long Delay provides 683 ms of delay at all
sample rates.
Mod Delay II plug-in (Long Delay shown)
• The Extra Long Delay provides 2.73 seconds of
delay at all sample rates.
Chapter 35: Mod Delay II
223
Mod Delay II Controls
Mod-Delay II provides a variety of controls for
adjusting plug-in parameters.
Mod Delay II Gain Control
This control controls the input level to the delay
to prevent clipping.
Mod Delay II Mix Control
This control controls the balance between the
delayed signal (wet) and the original signal
(dry). If you are using a delay for flanging or
chorusing, you can control the depth of the effect somewhat with the Mix setting.
Mod Delay II LPF (Low Pass
Filter)
Controls the cutoff frequency of the Low Pass
Filter. Use the LPF setting to attenuate the high
frequency content of the feedback signal. The
lower the setting, the more high frequencies are
attenuated. The maximum value for LPF is Off.
This lets the signal pass through without limiting the bandwidth of the plug-in.
Mod Delay II Rate Control
This control controls the rate of modulation of
the delayed signal.
Mod Delay II Feedback Control
This control controls the amount of feedback
applied from the output of the delay back into
its input. It also controls the number of repetitions of the delayed signal. Negative feedback
settings give a more intense “tunnel-like” sound
to flanging effects.
Mod Delay II Tempo Sync
Control
Tempo sync provides a direct connection between the Pro Tools session tempo and plug-in
controls that support MIDI Beat Clock (such as
Delay). This direct connection lets plug-in parameters such as delay, automatically synchronize to, and follow changes in, session tempo.
When Tempo Sync is enabled, the Tempo and
Meter controls are uneditable and follow the
session tempo and meter changes. The Duration
and Groove controls apply when Tempo Sync is
enabled.
Mod Delay II Delay Control
To enable Tempo Sync:
This control sets the delay time between the
original signal and the delayed signal.
 Click the Tempo Sync icon. The tempo shown
changes to match the current session tempo and
the meter changes to match the current meter.
Mod Delay II Depth Control
This controls the depth of the modulation applied to the delayed signal.
Tempo Sync icon
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Audio Plug-Ins Guide
Mod Delay II Tempo Control
This control sets the desired tempo in beats per
minute (bpm). This setting is independent of
Pro Tools’ tempo. When a specific Duration is
selected (see “Duration” below), moving this
control affects the Delay setting. Likewise, the
range of both controls will be limited to the
maximum available delay with the currently selected Duration. To enter very short or long delays it may be necessary to deselect all Duration
buttons.
Note value buttons
Dot modifier button
When Tempo Sync is enabled, the Tempo control is unavailable.
Mod Delay II Meter Control
Use this control to enter either simple or compound time signatures. The Meter control defaults to a 4/4 time signature.
When Tempo Sync is enabled, the Meter control
is unavailable.
Mod Delay II Duration Controls
Specifies a desired delay from a musical perspective. Enter the desired delay by selecting appropriate note value (whole note, half note,
quarter note, eight note, or sixteenth note). Select the Dot or Triplet modifier buttons to dot
the selected note value or make it a triplet. For
example, selecting a quarter note and then selecting the dot indicates a dotted quarter note,
and selecting an eighth note and then selecting
the triplet indicates a triplet eight note.
Triplet modifier button
Mod Delay II Groove Control
This control provides fine adjustment of the delay in percentages of a 1:4 subdivision of the
beat. It can be used to add “swing” by slightly
offsetting the delay from the precise beat of the
track.
It is not possible to exceed the maximum delay length for a particular version of
Mod Delay II. Consequently, when adjusting any of the tempo controls (Tempo, Meter, Duration, and Groove) you may not be
able to adjust the control across its full
range. If you encounter this behavior,
switch to a version of Mod Delay II that has
a longer delay time (for example, switch
from Medium Delay to Long Delay).
Mod Delay II Duration controls
Chapter 35: Mod Delay II
225
Multichannel Mod Delay II
The Tempo and Meter controls are linked on
multichannel versions of Mod Delay II. Each
channel has its own Duration and Groove controls, but the Tempo and Meter controls are
global.
Tempo, Meter, Duration, and Groove controls for a
stereo instance of Mod Delay II
Selecting Audio for
ModDelay II AudioSuite
Processing
Because AudioSuite Delay adds additional material (the delayed audio) to the end of selected
audio, make a selection that is longer than the
original source material to allow the additional
delayed audio to be written into the end of the
audio file.
Selecting only the original material, without
leaving additional space at the end, will cause
delayed audio that occurs after the end of the
rendered clip to be cut off.
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Audio Plug-Ins Guide
Chapter 36: Mod Delay III
Mod Delay III provides mono, multi-mono,
mono-to-stereo, and stereo modulating delay effects. Mod Delay III is available in AAX and AudioSuite formats, and supports 44.1 kHz,
48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz and
192 kHz sample rates.
Input
Input Meters
The Input meters show peak signal levels before
processing:
Dark Blue Indicates nominal levels from –INF to
Mod Delay III Controls
Mod Delay III provides separate sections in the
plug-in window for Input and Output metering,
Delay and Modulation controls, and for the
Wet/Dry Mix control. Stereo and mono-to-stereo versions provide meters and controls for
each channel. Delay, Modulation, and Mix controls for stereo and mono-to-stereo instances of
Mod Delay III can be linked, or can be operated
independently.
–12 dB.
Light Blue Indicates pre-clipping levels, from
–12 dB to 0 dB.
Red Indicates clipping.
Phase Invert
The Phase Invert button at the top of the Input
section inverts the phase (polarity) of the input
signal, to help compensate for phase anomalies
that can occur either in multi-microphone environments or because of mis-wired balanced connections.
To enable (or disable) phase inversion on input:
 Click the Phase Invert button so that it is highlighted. Click it again so that it is not highlighted to disable it.
Mod Delay III plug-in (Mono shown)
Chapter 36: Mod Delay III
227
Delay
Link
For stereo and mono-to-stereo tracks, enable
the Link button to link the Delay, Modulation,
and Mix controls between the Left and Right
channels. This option is highlighted when it is
enabled.
For mono tracks, this option reads Mono and is
display only.
Delay Time
The Delay Time control sets the delay time between the original signal and the delayed signal
(from 0.0 ms to 5,000.0 ms).
Feedback (FBK)
The Feedback setting controls the amount of
feedback applied from the output of the delay
back into its input (from –100% to 100%). It also
controls the number of repetitions of the delayed signal. Negative feedback settings give a
more intense “tunnel-like” sound to flanging effects.
Low Pass Filter (LPF)
The Low Pass Filter setting controls the cutoff
frequency of the Low Pass Filter (from 10 Hz to
22 kHz). Use the LPF setting to attenuate the
high frequency content of the feedback signal.
The lower the setting, the more high frequencies
are attenuated. The maximum value for LPF is
Off. This lets the signal pass through without
limiting the bandwidth of the plug-in.
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Pro Tools Reference Guide
Sync
When Sync is enabled, and a Duration (a rhythmic note value) is selected, the Delay Time setting is affected by the session tempo. When Sync
is disabled, and a Duration is selected, the Delay
Time setting is affected by changes to the Tempo
setting.
When Tempo Sync is enabled, the Tempo and
Meter controls are uneditable and follow the
session tempo and meter changes in the
Pro Tools timeline. The Duration and Groove
controls apply regardless of whether Sync is enabled.
Meter
The Meter setting lets you enter either simple or
compound time signatures. The Meter control
defaults to a 4/4 time signature.
When Sync is enabled, the Meter control is unavailable.
Tempo
The Tempo control sets the tempo in beats per
minute (from 5.00 to 500.00 bpm). This setting
is independent of the Pro Tools session tempo.
When a specific Duration is selected, moving
this control affects the Delay Time setting.
When Sync is enabled, the Tempo control is unavailable.
Duration
The Duration setting lets you set the Delay Time
based on a rhythmic value. Select the desired
note value (whole note, half note, quarter note,
eight note, or sixteenth note). Additionally, you
can select the Dot or Triplet modifier buttons to
dot the selected note value or make it a triplet.
For example, selecting a quarter note and then
selecting the dot indicates a dotted quarter note,
and selecting an eighth note and then selecting
the triplet indicates a triplet eighth note.
Output
The Output section provides output metering
and controls for adjusting the level of the output
signal.
Output Meters
The Output meters show peak signal levels after
processing:
Duration buttons
Dark Blue Indicates nominal levels from –INF to
Groove
–12 dB.
The Groove control provides fine adjustment of
the delay in percentages of a 1:4 subdivision of
the beat (from –100% to 100%). It can be used to
add “swing” by slightly offsetting the delay from
the precise beat of the track.
Light Blue Indicates pre-clipping levels, from
–12 dB to 0 dB.
Red Indicates full scale levels (clipping)
Output Gain
Modulation Section
Rate
The Rate control sets the rate of modulation of
the delayed signal (from 0.00 Hz to 20.0 Hz).
The Output Gain control sets the output level after processing. For mono instances of Mod Delay III, there is a single Gain control. For stereo
and mono-to-stereo instances of Mod Delay III,
there are independent Gain controls for each
channel (left and right).
Depth
The Depth control sets the depth of the modulation applied to the delayed signal (from 0% to
100%).
Mix
The Mix control sets the balance between the delayed signal (wet) and the original signal (dry).
If you are using a delay for flanging or chorusing, you can control the depth of the effect
somewhat with the Mix setting. Click the Dry
button to set the Mix to 100% dry. Click the Wet
button to set the Mix to 100% wet.
Selections for Mod Delay III
AudioSuite Processing
Because AudioSuite Delay adds additional material (the delayed audio) to the end of selected
audio, make a selection that is longer than the
original source material to allow the additional
delayed audio to be written to the end of the audio file.
If you select only the original material without
leaving additional space at the end, delayed audio that occurs after the end of the selection to
be cut off.
Chapter 36: Mod Delay III
229
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Pro Tools Reference Guide
Chapter 37: Moogerfooger Analog Delay
The Moogerfooger Analog Delay is a delay
plug-in that is available in TDM, RTAS, and AudioSuite formats. It provides a warm sounding
delay in the digital domain.
Moogerfooger Analog Delay
The Moogerfooger Analog Delay uses Bucket
Brigade Analog Delay Chips to achieve its delay.
These analog integrated circuits function by
passing the audio waveform down a chain of
thousands of circuit cells, just like water being
passed by a bucket brigade to put out a fire. Each
cell in the chip introduces a tiny time delay. The
total time delay depends on the number of cells
and on how fast the waveform is “clocked,” or
moved from one cell to the next.
With the advent of digital technology, these and
similar analog delay chips have gradually been
phased out of production. In fact, Bob Moog secured a supply of the last analog delay chips ever
made, and used them to build a Limited Edition
of 1,000 “real-world” Moogerfooger Analog Delay units.
How the Moogerfooger Analog Delay Works
So Why Analog?
A delay circuit produces a replica of an audio
signal a short time after the original signal.
Mixed together, the delayed signal sounds like
an echo of the original. And if this mixture is fed
back to the input of the delay circuit, the delayed
output provides a string of echoes that repeat
and die out gradually. It’s a classic musical effect.
Compared to digital delays, the frequency and
overload contours of well-designed analog delay
devices generally provide smoother, more natural series of echoes than digital delay units. Another difference is that the echoes of a digital
delay are static because they are the same digital
sound repeated over and over, whereas a bucket
brigade device itself imparts a warm, organically evolving timbre to the echoes.
Of course, Bomb Factory’s digital replica re-creates all the warm, natural sounds of its analog
counterpart.
Chapter 37: Moogerfooger Analog Delay
231
Not Better—Different
Working directly with Bob Moog, Bomb Factory
enhanced the Moogerfooger Analog Delay to be
even more useful for digital recording. An integrated Highpass Filter allows you to remove unwanted bass buildup from the feedback loop, allowing you to have warmer, more-controllable
echo swarms while minimizing the potential for
digital clipping.
Moogerfooger Analog Delay
Controls
The Moogerfooger Analog Delay provides the
following controls:
Mix The Mix control blends the original input
signal with the delayed signal.
LED Indicators
Three LEDs down the center of the unit provide
visual feedback.
Input Level The Input Level LED glows green
when signal is present.
HPF The HPF LED turns green when the highpass filter is enabled.
Bypass The Bypass LED glows either red (bypassed) or green (not bypassed) to show
whether or not the effect is in the signal path.
Moogerfooger Analog Delay Tips and Tricks
Delay Time Delay Time allows you to select the
length of delay between the original and the delayed signal. Used with Feedback, it also affects
how long apart the echoes are.
Short/Long The Short/Long switch sets the
range of the Delay Time control. Set to Short,
the Delay Time ranges from 0.04 to 0.4 seconds.
Set to Long, it ranges from 0.08 to 0.8 seconds.
Feedback Feedback determines how much sig-
nal is fed back to the delay input, affecting how
fast the echoes die out.
Highpass The Highpass knob removes low fre-
quencies from the feedback loop. It removes undesirable low frequency “mud” common when
mixing with delays and also allows the creation
of amazing echo swarms that won’t clip the output. Dial in a highpass frequency from 50 Hz to
500 Hz. Frequencies below the setting are filtered from the feedback loop.
HPF On/Off The HPF Off/HPF On enables or disables the highpass filter (HPF).
Drive The Drive control sets the input gain.
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Audio Plug-Ins Guide
Infidelity
Because analog delay chips offer only a fixed
number of cells, the extended delay times store a
lower-fidelity version of the input signal. Try
the Long delay setting when going for cool “lofi” sounds and textures.
Echo Swarms
By carefully adjusting the Feedback, Drive, and
Highpass controls, you can use the Moogerfooger Analog Delay as a sound generator. Simply pulse the delay unit with a short piece of audio (even a second will do), and adjust the Delay
Time knob. Set correctly, the unit will generate
cool timbres for hours all by itself.
Chapter 38: Multi-Tap Delay
Multi-Tap Delay is an AudioSuite plug-in that
adds up to four independently-controllable delays or taps to the original audio signal.
Use the Multi-tap delay to add spatialization or
complex rhythmic echo effects to audio material. You can individually control the delay time
and number of repetitions of each of the four
taps.
The Multi-Tap Delay plug-in was formerly
called D-fx Multi-Tap Delay. It is fully compatible with all settings and presets created
for D-fx Multi-Tap Delay.
Multi-Tap Delay Controls
The Multi-Tap Delay plug-in provides the following controls:
Gain Provides individual control of the input
level for each of the four delay lines (or “taps”).
Individually adjust the Gain for each of the four
taps, either to prevent clipping or to increase the
level of the processed signal.
Selecting the Sum Inputs button sums the dry
input signals (mono or stereo) before processing
them. The dry signal then appears in the center
of the stereo field and the wet, effected signal
will be output in stereo.
Feedback Provides individual control over the
amount of feedback applied from the output of
the delay into its input for each tap. It also controls the number of repetitions of the delayed
signal. For the feedback feature to function, the
Gain slider for that tap must be raised above its
lowest setting.
Multi-Tap Delay plug-in
Pan Provides individual control over the appar-
ent location of each of the four taps in the stereo
field.
Chapter 38: Multi-Tap Delay
233
Delay Sets the delay time between the original
signal and the delayed signal. The higher the setting, the longer the delay. This control is adjustable from 0–1500 milliseconds (1.5 seconds).
Mix Adjusts the balance between the effected
signal and the original signal and controls the
depth of the effect. Mix is adjustable from 0% to
100%.
Selecting Audio for AudioSuite
Delay Processing
Because delays add additional material to the
end of selected audio (a delay tap), make a selection that is longer than the original source material so AudioSuite can write the additional delayed audio to the audio file.
Selecting only the original material, without
leaving additional space at the end results in the
delayed audio being cutoff at the end of the selection. To accommodate delayed audio that
comes after the source audio, place the clip in a
track, and select the desired audio plus an
amount of blank space at the end of the clip
equal to the amount of delay that you have
added in the plug-in. The plug-in will then have
space at the end of the clip in which to write the
final delay.
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Chapter 39: Ping-Pong Delay
Ping-Pong Delay is an AudioSuite plug-in that
adds a controllable delay to the original audio
signal. Use the Ping-Pong delay to add spatialization, and panned echo to audio material. This
plug-in feeds back delayed signals to their opposite channels, creating a characteristic pingpong echo effect.
Delay Sets the delay time between the original
signal and the delayed signal. The higher the setting, the longer the delay. This control is adjustable from 0–1500 milliseconds (1.5 seconds).
Low Pass Filter Controls the cutoff frequency of
the low pass filter. Use this to attenuate the high
frequency content of the feedback signal. The
lower the setting, the more high frequencies are
removed from the feedback signal.
The range of the Low Pass Filter is 20 Hz to
19.86 kHz, with a maximum value of Off (which
effectively means bypass).
Feedback Controls the amount of feedback ap-
Ping-Pong Delay plug-in
The Ping-Pong Delay plug-in was formerly
called D-fx Ping-Pong Delay. It is fully compatible with all settings and presets created
for D-fx Ping-Pong Delay.
Ping-Pong Delay Controls
plied from the output of the delay into its input.
It also controls the number of repetitions of the
delayed signal.
Cross-Feedback Cross-Feedback feeds the de-
layed signals to their opposite channel: The left
channel delay is fed to the right channel input
and vice-versa. The result is a stereo echo that
ping-pongs back and forth between the right
and left channels.
The Ping-Pong Delay plug-in provides the following controls:
Gain Adjusts the input volume of the Ping-Pong
Delay to prevent clipping or to increase the level
of the processed signal.
Mix Adjusts the balance between the effected
signal and the original signal and controls the
depth of the effect. Mix is adjustable from 0% to
100%.
Chapter 39: Ping-Pong Delay
235
Selecting Audio for AudioSuite
Delay Processing
Because delays add additional material to the
end of selected audio (a delay tap), make a selection that is longer than the original source material so AudioSuite can write the additional delayed audio to the audio file.
Selecting only the original material, without
leaving additional space at the end results in the
delayed audio being cutoff at the end of the selection. To accommodate delayed audio that
comes after the source audio, place the clip in a
track, and select the desired audio plus an
amount of blank space at the end of the clip
equal to the amount of delay that you have
added in the plug-in. The plug-in will then have
space at the end of the clip in which to write the
final delay.
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Chapter 40: Reel Tape Delay
Reel Tape Delay is part of the Reel Tape suite of
tape-simulation effects plug-ins that are available in TDM, RTAS, and AudioSuite formats.
Reel Tape Delay simulates an analog tape echo
effect, modeling the frequency response, noise,
wow and flutter, and distortion characteristics
of analog tape. It also reproduces the varispeed
effect you get when the tape speed control is adjusted.
Reel Tape Delay automatically applies tape saturation effects that correspond to the following
control settings in Reel Tape Saturation:
• Speed: 15 ips
• Bias: 0.0 dB
• Cal Adjust: +9 dB
You can use the BPM Sync feature to synchronize the Reel Tape Delay effect to the current
tempo of the Pro Tools session.
Reel Tape Delay can be placed on mono, stereo,
or multichannel tracks.
Reel Tape Delay
How Reel Tape Delay Works
For years, engineers have relied on analog tape
to add a smooth, warm sound to their recordings. When driven hard, tape responds with gentle distortion rather than abrupt clipping as in
the digital domain. Magnetic tape also has a frequency-dependent saturation characteristic that
can lend punch to the low end, and sweetness to
the highs.
Reel Tape Delay models a studio tape machine in
record/playback mode, with a fixed distance between the record head and the play head, and a
continuously variable tape speed.
Chapter 40: Reel Tape Delay
237
Reel Tape Common Controls
All Reel Tape plug-ins share the following
controls:
Drive
Drive controls the amount of saturation effect
by increasing the input signal to the modeled
tape machine while automatically compensating
by reducing the overall output. Drive is adjustable from –12 dB to +12 dB, with a default value
of 0 dB.
Output
Output controls the output signal level of the
plug-in after processing. Output is adjustable
from –12 dB to +12 dB, with a default value of
0 dB.
Tape Machine
The Tape Machine control lets you select one of
three tape machine types emulated by the plugin, each with its own sonic characteristics:
US Emulates the audio characteristics of a
3M M79 multitrack tape recorder.
Swiss Emulates the audio characteristics of a
Studer A800 multitrack tape recorder.
Lo-Fi Simulates the effect of a limited-bandwidth analog tape device, such as an outboard
tape-based echo effect.
Tape Formula
The Tape Formula control lets you select either
of two magnetic tape formulations emulated by
the plug-in, each with its own saturation characteristics:
Classic Emulates the characteristics of
Ampex 456, exhibiting a more pronounced saturation effect.
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Audio Plug-Ins Guide
Hi Output Emulates the characteristics of
Quantegy GP9, exhibiting a more subtle saturation effect.
Reel Tape Delay Controls
In addition to the Drive, Output, Tape Machine,
and Tape Formula controls, Reel Tape Delay has
the following controls:
Speed
The Speed control adjusts the delay time, calibrated to tape speed. A slower tape speed results
in a longer delay. A faster tape speed results in a
shorter delay.
The displayed tape Speed value corresponds to
the delay time resulting from the distance between the record and play heads on an Ampex
440-series tape transport.
Tape speed is adjustable from approximately
1 7/8 ips (1486 ms delay) to approximately
30 ips (93 ms delay), with a default value of approximately 15 ips (172 ms delay).
You can synchronize the delay time to the current tempo of the Pro Tools session. See “Synchronizing Reel Tape Delay to Session Tempo”
on page 240.
Feedback
The Feedback control adjusts the amount of delayed output fed back into the input, allowing
generation of multiple echoes. A higher feedback amount results in more echo regeneration.
A lower feedback amount results in less echo regeneration. Feedback amount is adjustable from
0 to 100 percent, with a default value of
30 percent.
Wow/Flutter
Treble
The Wow/Flutter control adjusts the amplitude
of the tape machine’s wow and flutter, or the
amount of fluctuation in tape speed. A higher
setting results in wider fluctuations in speed. A
lower setting results in narrower fluctuations in
speed. Wow/Flutter is adjustable from 0 to
1 percent, with a default value of 0.20 percent.
The Treble control boosts or cuts the amount of
high-mid frequencies fed to the echo feedback
loop. Treble amount is adjustable from –10 dB
to +10 dB, with a default value of 0 dB.
Wow Speed
(Plug-In Automation Playlist or
Control Surface Access Only)
The Wow Speed parameter adjusts the frequency of the tape machine’s wow effect, or the
rate of fluctuation in tape speed. A higher value
results in faster fluctuations in speed. A lower
value results in slower fluctuations in speed.
Wow Speed is adjustable from 0 to 100 percent,
with a default value of 50 percent.
This parameter is accessible only from the plugin automation playlist or from a supported control surface.
Settings for this parameter are saved with
plug-in presets. If you use a preset for the
TDM, RTAS or AudioSuite version of this
plug-in, any settings for this parameter
will be active.
Bass
The Bass control boosts or cuts the amount of
low frequencies fed to the echo feedback loop.
Bass amount is adjustable from –10 dB to
+10 dB, with a default value of 0 dB.
Note that this control does not affect the
first delayed signal, only the repeated delays caused by the Feedback control.
Mix
The Mix control adjusts the amount of processed signal mixed with the input signal in the
final output of the plug-in. The default Mix
value is 25 percent.
Noise
(Plug-In Automation Playlist or Control
Surface Access Only)
The Noise parameter controls the level of simulated tape hiss that is added to the processed signal. Noise is adjustable from Off (–INF) to
–24 dB, with a default value of –80 dB.
This parameter is accessible only from the plugin automation playlist or from a supported control surface.
Settings for this parameter are saved with
plug-in presets. If you use a preset for the
TDM, RTAS or AudioSuite version of this
plug-in, any settings for this parameter
will be active.
Note that this control does not affect the
first delayed signal, only the repeated delays caused by the Feedback control.
Chapter 40: Reel Tape Delay
239
Synchronizing Reel Tape Delay
to Session Tempo
You can set the delay time (Speed control) in the
Reel Tape Delay to synchronize to the session
tempo (in beats per minute).
To synchronize the delay time to the session
tempo:
In the BPM Sync section, click the On button.
The Tempo/Rate display changes to match the
current session tempo.
1
Reel Tape Delay Presets
The Reel Tape Delay presets coordinate Speed,
Wow/Flutter, Feedback and the Bass and Treble
controls for different tape speeds.
3.75 ips Sets the delay time to correspond to a
Speed Control setting of 3.75 inches per second.
3.75 ips Flutter Includes the 3.75 ips setting plus
Wow/Flutter.
7.5 ips Sets the delay time to correspond to a
Speed Control setting of 7.5 inches per second.
7.5 ips Flutter Includes the 7.5 ips setting plus
Wow/Flutter.
Tempo/Rate
display
On
button
Note Value
display
Dot
button
Triplet
button
BPM Sync controls
2 To set a rhythmic delay, click the Note Value to
choose from the available note values (whole,
half, quarter, eighth, sixteenth, or thirty-second
note)
To adjust the rhythm further, do any of the following:
3
• To enable triplet rhythm delay timing, click
the Triplet (“3”) button so that it is lit.
• To set a dotted rhythm delay value, click the
Dot (“.”) button so that it is lit.
You can override the settings derived from
BPM Sync at any time by manually adjusting the plug-in Speed control.
To set the delay time to a specific time value,
turn off BPM Sync and enter the delay time
(in msec) in the Tempo/Rate display.
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Audio Plug-Ins Guide
30 ips Flutter Adds Wow/Flutter to the highest
Speed Control setting.
Rockabilly A common tape slap effect, useful on
vocals or electric guitar. Sets the delay time to
130 ms, which corresponds to the delay time resulting from the distance between the record
and play heads on an Ampex 300-series or Ampex 350-series tape transport.
Rockabilly Plus Includes the Rockabilly setting
plus Feedback, Wow/Flutter, Bass and Treble
adjustments on feedback.
Chapter 41: Tel-Ray Variable Delay
Tel-Ray Variable Delay is a delay/echo plug-in
that is available in TDM, RTAS, and AudioSuite
formats.
Add delay or echo to any voice or instrument using the Tel-Ray Variable Delay. It provides lush
delay, amazing echo, and warms up your tracks
and mixes.
How the Tel-Ray Works
In the early 1960s, a small company experimented with electronics and technology. When
they came up with something great, they would
Tell Ray (the boss).
Tel-Ray Variable Delay
Space-age technology in a can
One invention involved a tuna can, a motor, and
a few tablespoons of cancer-causing oil. The creation: an Electronic Memory Unit. A technology, they were sure, that would be of great interest to companies like IBM and NASA.
Though it never made it to the moon, most every
major guitar amp manufacturer licensed the
killer technology that gives Tel-Ray its unique
sound.
Chapter 41: Tel-Ray Variable Delay
241
Tel-Ray Controls
Tel-Ray Tips and Tricks
Input/Output Section
Variation? Do They Ever!
Input Input sets the signal level to the tuna can
echo unit.
Each and every Tel-Ray we tested (and Bomb
Factory owns more than a dozen) varied drastically in motor and flywheel stability, resulting
in different pitch and variation effects. The
same unit even sounded different day to day, depending on temperature, warm-up time and
other factors.
Tone Tone is a standard tone control like those
commonly found on guitar effects.
Mix Mix adjusts the amount of dry (unpro-
cessed) signal relative to the amount of wet
(processed) signal. Full clockwise is 100% wet.
(On original units, this control is located deep
inside the box, typically soaked in carcinogenic
PCB oil.)
Output Output is a simple digital output trim
control.
Echo/Delay Section
Variable Delay Variable Delay selects the delay
time. Delay times vary from 0.06 to 0.3 seconds.
Full clockwise is slowest.
Variation Variation adjusts how much variation
occurs in the delay. The more variation you use,
the more warbled and wobbly the sound becomes.
Sustain Sustain determines how long the delay
takes to die out. It is actually a feedback control
similar to the one found on the Moogerfooger
Analog Delay.
Echo/Doubler Echo/Doubler determines
whether or not a second record head is engaged,
resulting in a double echo.
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Since the original units are basically thirty yearold tuna cans bolted to plywood with springs
and motors flopping around inside, Bomb Factory added the Variation knob so you can dial in
a Tel-Ray in whatever state of disrepair you desire.
Chapter 42: TimeAdjuster
TimeAdjuster is a time-processing plug-in that
is available in TDM and RTAS formats.
The TimeAdjuster plug-in is an efficient way to
compensate for DSP or host-based processing
delays in your Pro Tools system.
Long Supports a maximum delay of 8192 samples at all sample rates.
For more information on Delay Compensation and Time Adjuster, see the Pro Tools
Reference Guide.
TimeAdjuster Controls
The TimeAdjuster plug-in provides the following controls:
TimeAdjuster plug-in
Use the TimeAdjuster plug-in for any of the following:
• Delay compensation
• Gain compensation (+/– 24 dB)
• Phase inversion for correcting out-of-phase
signals
There are three versions of the TimeAdjuster
plug-in, each of which supports different sample
delay ranges:
Short Supports a maximum delay of 256 samples
at all sample rates.
Medium Supports a maximum delay of 2048
samples at all sample rates.
Phase Invert This controls inverts the phase (polarity) of the input signal. While most Avid
plug-ins supply a phase invert button of their
own, some third-party plug-ins may not. Phase
inversion is also useful for performing delay
compensation by tuning unknown delay factors
by ear (see “Using TimeAdjuster for Manual Delay Compensation” on page 244).
Gain Provides up to 24 dB of positive or negative
gain adjustment. This control is useful for altering the gain of a signal by a large amount in real
time. For example, when you are working with
audio signals that are extremely low level, you
may want to adjust the channel gain to a reasonable working range so that a fader is positioned
at its optimum travel position. Use the Gain
control to make a wide range of gain adjustment
in real time without having to permanently process the audio files, as you would with an AudioSuite plug-in.
Chapter 42: TimeAdjuster
243
Delay Provides up to 8192 samples of delay com-
pensation adjustment, or general adjustment of
phase relationships of audio recorded with multiple microphones, depending on which version
of TimeAdjuster is used. It defaults to a minimum delay of four samples, which is the delay
created by use of the TimeAdjuster plug-in itself.
While phase inversion controls have been used
for many years by engineers as creative tools for
adjustment of frequency response between multiple microphones, sample-level delay adjustments provide far more control. Creative use of
this control can provide a powerful tool for adjusting frequency response and timing relationships between audio signals recorded with multiple microphones.
Using TimeAdjuster for
Manual Delay Compensation
DSP and host-based processing in all digital systems incurs delay of varying amounts. You can
use the TimeAdjuster TDM plug-in to apply an
exact number of samples of delay to the signal
path of a Pro Tools track to compensate for delay incurred by specific plug-ins. TimeAdjuster
provides presets for common delay-compensation scenarios.
To compensate for several plug-ins in-line, use
the delay times from each settings file as references, and add them together to derive the total
delay time.
Some plug-ins (such as Avid’s Maxim and
DINR BNR) have different delays at different sample rates. See for more information
about these plug-ins.
Alternatively, look up the delay in samples for
the plug-ins you want to compensate for, then
apply the appropriate amount of delay.
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Audio Plug-Ins Guide
To manually compensate for DSP-induced delays, try one of the following methods:
• Phase inversion
– or –
• Comb-filter effect cancellation
Phase Inversion
If you are working with phase-coherent track
pairs, or tracks recorded with multiple microphones, you can invert the phase to negate the
delay. If you don’t hear any audio when you invert a signal’s phase, you have precisely adjusted
and compensated for the delay. This is because
when you monitor duplicate signals and invert
the polarity (phase) of one of them, the signals
will be of opposite polarity and cancel each
other out. This technique is convenient for finding the exact delay setting for any plug-in.
To determine the delay of a plug-in by inverting its
signal phase:
1 Place duplicate audio clips on two different audio tracks and pan them to the center (mono).
Apply the plug-in whose delay you want to calculate to the first track, and a Time Adjuster
plug-in to the second track.
2
3
With TimeAdjuster, invert the phase.
Control-drag (Windows) or Command-drag
(Mac) to fine-tune delay in one sample increments, or use the up/down arrow keys to change
the delay one sample at a time until the audio
signal disappears.
4
5
Change the polarity back to normal.
6
Save the TimeAdjuster setting for later use.
Comb-Filter Effect Cancellation
Adjust the delay with the signal in phase until
any comb-filter effects cancel out.
Viewing Channel Delay and
TimeAdjuster
Because plug-ins display their delay values in
the channel delay indicators, this can be used as
another method for determining delay compensation.
To view time delay values and use TimeAdjuster to
compensate for the delay:
Control-click (Windows) or Command-click
(Mac) the Track Level Indicator to toggle between level (that appears on the display as
“vol”), headroom (“pk”), and channel delay
(“dly”) indications. Delay values are shown in
samples.
1
When to Compensate for
Delays
If you want to compensate for delays across your
entire system with Time Adjuster, you will want
to calculate the maximum delay incurred on any
channel, and apply the delays necessary to each
channel to match this channel.
However, this may not always be necessary. You
may only really need to compensate for delays
between tracks where phase coherency must be
maintained (as with instruments recorded with
multiple microphones or stereo pairs). If you
are working with mono signals, and the accumulated delays are small (just a few samples, for example), you probably needn’t worry about delay
compensation.
For more information about delays and
mixing with Pro Tools HD, see the Pro Tools
Reference Guide.
Determining the DSP delay of track inserts (Mix
window shown)
Apply the TimeAdjuster plug-in to the track
whose delay you want to increase, and Controlclick (Windows) or Command-click (Mac) its
Track Level indicator until the channel delay
value is displayed for that track.
2
Change the delay time in TimeAdjuster by
moving the Delay slider or entering a value in
the Delay field, until the channel delay value
matches that of the first track.
3
Test the delay values by duplicating an audio
track and reversing its phase while compensating for delay.
4
Chapter 42: TimeAdjuster
245
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Audio Plug-Ins Guide
Part VII: Modulation Plug-Ins
Chapter 43: AIR Chorus
AIR Chorus is an RTAS plug-in that lets you apply a short modulated delay to give depth and
space to an audio signal.
Feedback Sets the Feedback amount.
Pre-Delay Delays the chorused signal, in milli-
seconds.
LFO Section
The Chorus plug-in’s LFO section’s controls let
you select the waveform, phase, rate, and depth
of modulation.
Waveform Selects either a Sine wave or a Triangle wave for the LFO.
L/R Phase Sets the relative phase of the LFO’s
Chorus plug-in window
AIR Chorus Controls
The Chorus plug-in provides a variety of controls for adjusting plug-in parameters.
modulation in the left and right channels.
Mix
This control adjusts the Mix between the “wet”
(processed) and “dry” (unprocessed) signal. 0%
is all dry, and 100% is all wet, while 50% is an
equal mix of both.
Rate
This controls sets the rate for the oscillation of
the LFO in Hertz.
Depth
This control sets the depth of LFO modulation of
the audio signal.
Chorus Section
The Chorus plug-in’s chorus section’s controls
let you select the amount of feedback and the
length of pre-delay.
Chapter 43: AIR Chorus
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Audio Plug-Ins Guide
Chapter 44: AIR Ensemble
AIR Ensemble is an RTAS plug-in that lets you
apply fluid, shimmering modulation effects to
the audio signal.
Shimmer The Shimmer control lets you random-
ize the Delay time, adding texture to the effect.
Stereo Width The Stereo Width control lets you
widen or narrow the effect’s stereo field.
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet signal. At
50%, there are equal amounts of dry and wet signal. At 0%, the output is all dry and at 100% it is
all wet.
Ensemble plug-in window
Ensemble Controls
The Ensemble plug-in provides the following
controls:
Rate The Rate control changes the frequency of
the modulating LFO (0.01–10.0 Hz).
Depth The Depth control lets you adjust the
amount of modulation applied to the Delay
time.
Modulation Section
The Modulation controls let you adjust and/or
randomize the delay time.
Delay The Delay control lets you adjust the De-
lay time.
Chapter 44: AIR Ensemble
251
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Audio Plug-Ins Guide
Chapter 45: AIR Filter Gate
AIR Filter Gate is an RTAS plug-in that you can
use to chop up an audio signal into staccato
rhythmic patterns with variable filtering, amplitude, and panning.
Rate
The Rate selector lets you select the duration, or
frequency of the Low Frequency Oscillator
(LFO). The duration of one cycle of the LFO is
measured in Steps.
Swing
The Swing control sets the amount of rhythmic
swing applied to the chosen gating pattern.
Mix
The Mix control lets you adjust the Mix between
the “wet” (filtered) and “dry” (unfiltered) signal. 0% is all dry, and 100% is all wet, while 50%
is an equal mix of both.
Filter Gate plug-in window
Filter Gate Gate Section
Filter Gate Controls
The Filter Gate plug-in provides a variety of
controls for adjusting plug-in parameters.
Pattern
The Pattern control let you select from a number
of preset rhythmic patterns that the gate will
follow.
The Gate controls let you adjust the Attack,
Hold, and Release amounts for the Gater step sequencer pattern. At the maximum settings, the
gating provides a smooth morphing effect.
Attack
The Attack control lets you adjust the duration
of the attack as a percentage of the step duration.
Hold
The hold control lets you adjust the duration of
the hold (or sustain) as a percentage of the step
duration.
Chapter 45: AIR Filter Gate
253
Release
The Release control lets you adjust the duration
of the release as a percentage of the step duration.
Filter Gate Filter Section
The Filter controls provide controls for the selected filter type.
Filter Gate Modulation Section
Env
The Env control lets you adjust how much an
Envelope Follower affects the Cutoff frequency.
Note that the Cutoff is fixed for the duration of
each step, so it will not respond to a peak in the
envelope until the start of the next step.
LFO Mod
Mode
The Filter Mode selector lets you select the type
of Filter.
Off Provides no filtering.
LP Provides a Low Pass filter.
BP Provides a Band Pass filter.
HP Provides a High Pass filter.
Phaser Provides a Phaser.
Cutoff
The Cutoff control lets you adjust the Filter Cutoff frequency.
Res
The Res control lets you adjust the Resonance at
the Cutoff frequency.
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Audio Plug-Ins Guide
The LFO Mod control lets you adjust the amount
of LFO modulation of the Cutoff frequency.
LFO Steps Sets the duration of one cycle of the
LFO to the selected number of steps. Changes to
the Step Rate consequently affect the durations
of cycles of the LFO. When set to Random mode,
the level of the LFO changes randomly every
step, for a “sample and hold” waveform.
Chapter 46: AIR Flanger
AIR Flanger is an RTAS plug-in that lets you apply a short modulating delay to the audio signal.
Rate
When Sync is enabled, the Rate control lets you
select a rhythmic subdivision or multiple of the
beat for the Flanger Modulation Rate. Select
from the following rhythmic values:
• 16 (sixteenth note)
• 8T (eighth-note triplet)
• 16D (dotted sixteenth-note)
• 8 (eighth note)
• 4T (quarter-note triplet)
• 8D (dotted eighth-note)
Flanger plug-in window
• 4 (quarter note)
• 2T (half-note triplet)
AIR Flanger Controls
• 4D (dotted quarter-note)
The Flanger plug-in provides a variety of controls for adjusting plug-in parameters.
• 1T (whole-note triplet)
• 2 (half note)
• 3/4 (dotted half note)
Sync
• 4/4 (whole note)
When Sync is enabled, the Flanger Rate control
synchronizes to the Pro Tools session tempo.
When Sync is disabled, you can set the delay
time in milliseconds independently of the
Pro Tools session tempo. The Sync button is lit
when it is enabled.
• 5/4 (five tied quarter notes)
• 6/4 (dotted whole note)
• 8/4 (double whole note)
When Sync is disabled, the Rate control lets you
the modulation rate in independently of the
Pro Tools session tempo.
Chapter 46: AIR Flanger
255
Depth
L/R Offset
The Depth control lets you adjust the amount of
modulation applied to the Delay time.
The L/R Offset control lets you adjust the phase
offset for the LFO waveform applied to the left
and right channels.
Feedback
The Feedback control lets you adjust the amount
of delay feedback for the Flanger. At 0%, the delay repeats only once. At +/–100%, the Flanger
feeds back on itself.
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet (flanged) signal. At 50%, there are equal amounts of dry and
wet signal. At 0%, the output is all dry and at
100% it is all wet.
The Mix control can be used to create an “infinite phaser” effect between the dry and shifted
signals, which is always rising or always falling
(depending on the direction of shift)
Pre-Delay
The Pre-Delay control sets the minimum delay
time in milliseconds.
AIR Flanger LFO Section
Controls
The LFO section provides controls for the Low
Frequency Oscillator (LFO) used to modulate
the Delay time.
Wave
The Wave control lets you interpolate between a
triangle wave and a sine wave for the modulating LFO.
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Audio Plug-Ins Guide
Retrigger
Click the Retrigger button to reset the LFO
phase. This lets you manually start the filter
sweep from that specific point in time (or using
automation, at a specific point in your arrangement). Clicking the Trig button also forces the
Mix control up if it is too low while the button is
held; this ensures that the sweep is audible.
AIR Flanger EQ Section Controls
The EQ section provides controls for cutting
lows from the Flanger signal, and inverting
phase.
Low Cut
The Low Cut control lets you adjust the Low Cut
frequency for the Flanger, to limit the Flanger
effects to higher frequencies.
Phase Invert
When Phase Invert is enabled, the wet signal’s
polarity is flipped, which changes the harmonic
structure of the effect.
Chapter 47: AIR Fuzz-Wah
AIR Fuzz-Wah is an RTAS plug-in that lets you
add color to an audio signal with various types
and varying amounts of transistor-like distortion.
Fuzz-Wah Controls
The Fuzz-Wah plug-in provides a variety of controls for adjusting plug-in parameters.
Fuzz
Click the Fuzz button to turn the distortion effect on and off.
Drive
The Drive control sets the level of gain in the
Fuzz algorithm.
Mix
Fuzz-Wah plug-in window
The Mix control lets you balance the amount of
dry signal with the amount of wet (distorted)
signal. At 50%, there are equal amounts of dry
and wet signal. At 0%, the output is all dry and at
100% it is all wet.
Post Wah
The Post Wah control lets you place the Fuzz
section before the Wah section, or vice versa.
Wah
Click the Wah button to turn the wah filter on
and off.
Pedal
The Pedal control sweeps the wah center frequency up and down.
Chapter 47: AIR Fuzz-Wah
257
Filter
The Filter control switches the wah filter between LP (lowpass), BP (bandpass), and HP
(highpass) modes.
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet (wah-processed) signal. At 50%, there are equal amounts
of dry and wet signal. At 0%, the output is all dry
and at 100% it is all wet
Mix (Overall)
The overall Mix control lets you balance the
amount of fuzz-processed signal with the
amount of wah-processed signal. At 50%, there
are equal amounts of fuzz and wah signal. At 0%,
the output is all fuzz, and at 100% it is all wah.
Fuzz-Wah Pedal Min and Pedal
Max Section Controls
Freq
Sets the low (Pedal Min) and high (Pedal Max)
limits of the wah filter’s frequency sweep.
Res
Sets the low (Pedal Min) and high (Pedal Max)
limits of the wah filter’s resonance.
Fuzz-Wah Modulation Section
Controls
The Modulation section provides controls for
the Low Frequency Oscillator (LFO) and Envelope Follower (ENV) that can be used to modulate the wah filter’s sweep.
Rate
Fuzz-Wah Fuzz Section Controls
The Fuzz section provides tonal and volume
control over the plug-in.
Tone
The Tone control lets you change the brightness
of the Fuzz algorithm.
Output
The Output control sets the overall output volume of the Fuzz section.
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The Rate control sets either the LFO frequency,
or the response time of the envelope follower,
depending on the setting of the Mode control.
Type
The Type control lets you select either the LFO
or the Envelope follower as the modulation
source for the wah filter.
Depth
The Depth control sets the amount of modulation sent by the LFO or envelope follower.
Chapter 48: AIR Multi-Chorus
AIR Multi-Chorus is an RTAS plug-in that lets
you apply a thick, complex Chorus effect to an
audio signal.
Voices
The Voices control sets the number of layered
chorus effects that are applied to the audio signal. The more Voices that are used, the thicker
the effect.
Mix
The Mix control lets you adjust the Mix between
the “wet” (processed) and “dry” (unprocessed)
signal. 0% is all dry, and 100% is all wet, while
50% is an equal mix of both.
Multi-Chorus Plug-In window
Multi-Chorus Controls
The Multi-Chorus plug-in provides a variety of
controls for adjusting plug-in parameters.
Multi-Chorus Chorus Section
Controls
The Chorus section provides control over the
low-frequency content and stereo width of the
MultiChorus effect.
Low Cut
Rate
The Rate control sets the rate for the oscillation
of the LFO in Hertz.
The Low Cut control lets you adjust the Low Cut
frequency for the Flanger, to limit the Flanger
effects to higher frequencies.
Depth
Width
The Depth control sets the depth of LFO modulation of the audio signal in milliseconds.
The Width control lets you widen or narrow the
effect’s stereo field
Chapter 48: AIR Multi-Chorus
259
Multi-Chorus Mod Section
Controls
The Mod section controls let you set the Pre-Delay amount, and the waveform of the LFO.
Pre-Delay
Sets the Pre-Delay in milliseconds.
Waveform
Selects either a Sine wave or a Triangle wave for
the LFO.
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Chapter 49: AIR Phaser
AIR Phaser is an RTAS plug-in that applies a
phaser to an audio signal for that wonderful
“wooshy,” “squishy” sound.
Rate
When Sync is enabled, the Rate control lets you
select a rhythmic subdivision or multiple of the
beat for the Phaser Modulation Rate. Select from
the following rhythmic values:
• 16 (sixteenth note)
• 8T (eighth-note triplet)
• 16D (dotted sixteenth-note)
• 8 (eighth note)
• 4T (quarter-note triplet)
• 8D (dotted eighth-note)
• 4 (quarter note)
Phaser plug-in window
• 2T (half-note triplet)
• 4D (dotted quarter-note)
Phaser Controls
• 2 (half note)
The Phaser plug-in provides a variety of controls for adjusting plug-in parameters.
• 3/4 (dotted half note)
Sync
• 5/4 (five tied quarter notes)
When Sync is enabled, the Phaser Rate control
synchronizes to the Pro Tools session tempo.
When Sync is disabled, you can set the Rate in
milliseconds independently of the Pro Tools
session tempo. The Sync button is lit when it is
enabled.
• 1T (whole-note triplet)
• 4/4 (whole note)
• 6/4 (dotted whole note)
• 8/4 (double whole note)
When Sync is disabled, the Rate control lets you
the rate of the Phaser in independently of the
Pro Tools session tempo.
Chapter 49: AIR Phaser
261
Depth
The Depth control lets you adjust the depth of
modulation, which in turn affects the amount of
phasing applied to the audio signal.
Phaser LFO Section Controls
The LFO section provides control over the waveform and stereo offset of the LFO.
Wave
Feedback
The Feedback control feeds the output signal of
Phaser back into the input, creating a resonant
or singing tone in the phaser when set to its
maximum.
Mix
The Mix control lets you adjust the Mix between
the “wet” (effected) and “dry” (unprocessed)
signal. 0% is all dry, and 100% is all wet, while
50% is an equal mix of both.
Low Cut
The Low Cut control lets you adjust the frequency of the Low Cut Filter in the phaser’s
feedback loop. This can be useful for taming low
frequency “thumping” at high feedback settings.
Phaser Section Controls
The Phaser section provides control over the effect’s center frequency and number of phaser
stages (or Poles).
Center
The Center control lets you change the frequency center (100 Hz to 10.0 kHz) for the
phaser poles.
Poles
Select the number of phaser poles (stages): 2, 4,
6, or 8. The number of poles changes the character of the sound. The greater the number of
poles, the thicker and squishier the sound.
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Audio Plug-Ins Guide
The Wave control lets you interpolate between a
triangle wave and a sine wave for modulating
the Phaser.
L/R Phase
The L/R Phase control lets you adjust the relative phase of the LFO modulation applied to the
left and right channels.
Chapter 50: AIR Talkbox
AIR Talkbox is an RTAS plug-in that lets you
add a voice-like resonances to audio signals.
Talkbox Plug-In window
Talkbox Controls
The Talkbox plug-in provides a variety of controls for adjusting plug-in parameters.
Env Depth
The Env Depth knob creates a positive or negative offset in the setting of the Vowel control, effected by the Envelope follower. At its center,
the knob has no effect. Turned to the right or left
of center, the Env Depth knob shifts the value of
the Vowel control up or down.
When the Envelope follower is triggered, the
Vowel parameter moves to its normal setting (in
time with the envelope’s attack), then back to
the offset value (in time with the envelope’s release).
Formant
The Formant control lets you shift the formant
center of the processed audio up or down 12
semitones, changing the harmonic structure
dramatically.
Vowel
The Vowel control lets you choose the shape of
the formant filter, by the vowel sound that is
simulated (OO/OU/AU/AH/AA/AE/EA/EH
/EE/ER/UH/OH/OO).
Mix
The Mix control lets you adjust the Mix between
the “wet” (processed) and “dry” (unprocessed)
signal. 0% is all dry, and 100% is all wet, while
50% is an equal mix of both.
Chapter 50: AIR Talkbox
263
Talkbox LFO Section Controls
Saw Provides a saw-tooth wave.
The LFO section provides controls that let you
apply a Low Frequency Oscillator to modulate
the Formant setting.
Square Provides a square wave.
Rate
Random Provides random modulation.
When Sync is enabled, the Rate control lets you
select a rhythmic subdivision or multiple of the
beat for the LFO Rate. Select from the following
rhythmic values:
S&H Provides Sample and Hold (S&H) modula-
tion.
Depth
The Depth control lets you adjust the amount of
modulation applied to the Formant setting.
• 16 (sixteenth note)
• 8T (eighth-note triplet)
• 16D (dotted sixteenth-note)
• 8 (eighth note)
• 4T (quarter-note triplet)
• 8D (dotted eighth-note)
Sync
Enable Sync to synchronize the LFO Rate to the
Pro Tools session tempo. When Sync is disabled,
you can set the Rate time in milliseconds independently of the Pro Tools session tempo. The
Sync button is lit when it is enabled.
• 4 (quarter note)
• 2T (half-note triplet)
• 4D (dotted quarter-note)
• 2 (half note)
• 1T (whole-note triplet)
• 3/4 (dotted half note)
Talkbox Envelope Section
Controls
The Talkbox plug-in provides an Envelope follower for modulating the Formant setting. This
is useful for accentuating and enhancing signal
peaks in rhythmic material.
• 4/4 (whole note)
• 5/4 (five tied quarter notes)
Thresh
• 6/4 (dotted whole note)
Adjust the Thresh control to set the amplitude
threshold at which the Formant setting begins to
be modulated by the Envelope follower.
• 8/4 (double whole note)
When Sync is disabled, the Rate control lets you
change the modulation rate independently of
the Pro Tools session tempo (0.01–10.0 Hz).
Wave
Select from the following waveforms for the
LFO:
Sine Provides a sine wave.
Tri Provides a triangle wave.
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Audio Plug-Ins Guide
Attack
Adjust the Atk (attack) control to set the time
(10.0 ms to 10 seconds) it takes to respond to increases in the audio signal level.
Release
Adjust the Rel (release) control to set the time
(10.0 ms to 10 seconds) it takes to recover after
the signal level falls.
Chapter 51: AIR Vintage Filter
AIR Vintage Filter is an RTAS plug-in that applies a modulating, resonant filter to an audio
signal. Have fun with filter sweeps or give your
sounds that extra-resonant aura.
Vintage Filter Controls
The Vintage Filter plug-in provides a variety of
controls for adjusting plug-in parameters.
Cutoff
The Cutoff control lets you adjust the Cutoff frequency (20.0 Hz to 20.0 kHz) of the filter.
Resonance
The Resonance control lets you adjust the
amount filter Resonance (0–100%). The filter
can go into self-oscillation at high values creating a sine wave-like overtone at the Cutoff frequency.
Filter plug-in window
Fat
The Fat control lets you adjust the amount of
overdrive in the resonant peak. At lower settings
the signal gets quieter at high Resonance settings for clean distortion. At higher settings the
signal is over-driven at high resonance settings.
Chapter 51: AIR Vintage Filter
265
Mode
Select one of the following options for the type
of filter:
LP24 Provides a low pass filter with a 24 dB cut-
off.
LP18 Provides a low pass filter with a 18 dB cut-
off.
LP12 Provides a low pass filter with a 12 dB cut-
off.
BP Provides a band pass filter.
HP Provides a high pass filter.
Output
The Output control lets you lower the Output
level from 0.0 dB to –INF dB.
Vintage Filer Envelope Section
Controls
The Filter effect provides an Envelope follower
for controlling the Cutoff frequency. The Envelope section offers control over the envelope’s
shape and depth of modulation.
Attack
Adjust the Attack control to set the time
(10.0 ms to 10 seconds) it takes to respond to increases in the audio signal level.
Release
Adjust the Release control to set the time
(10.0 ms to 10 seconds) it takes to recover after
the signal level falls.
Depth
Adjust the Depth control to determine how
much the Envelope follower affects the Cutoff
frequency.
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Audio Plug-Ins Guide
 At 0%, the Envelope follower has no effect on
the Cutoff frequency.
 At +100%, the Attack ramps up to the Cutoff
frequency setting; and the Release starts from
the Cutoff frequency setting and ramps down.
 At –100%, the Attack starts from the Cutoff
frequency setting and ramps down; and the Release ramps up to the Cutoff frequency setting.
Vintage Filer LFO Section
Controls
The Filter effect provides a sinusoidal Low Frequency Oscillator (LFO) for modulating the filter cutoff frequency. The LFO section offers control over the rate, depth and synchronization of
the modulation.
Rate
Adjust the Rate control to increase or decrease
the frequency (0.01–100.0 Hz) of the LFO.
Lower settings are slower and higher settings
are faster. When Sync is on, the Rate knob
switches from counting in milliseconds, to
rhythmic values.
Depth
Adjust the Depth control to increase (or decrease) the amount of modulation (0–100%) of
the Cutoff frequency by the LFO. Lower settings
create a slight vibrato (with the rate set high)
and higher settings create a wide sweep of the
Cutoff frequency range.
Sync
Click the Sync button to synchronize the LFO
with the session tempo.
Chapter 52: Cosmonaut Voice
Cosmonaut Voice is a plug-in effect that is available in RTAS and AudioSuite formats. The Cosmonaut Voice plug-in is a radio and shortwave
simulator. Use it to add squelch or noise to
tracks.
Cosmonaut Voice Controls
Cosmonaut Voice is, in simple terms, an amplitude-driven noise generator with adjustable
sensitivity, selectable noise type (beep or
squelch), and an additional RFI/static noise
generator.
Cosmonaut Voice provides the following controls:
Threshold Sets the point at which the selected
Cosmonaut Voice
voice (beep or squelch) is triggered. Turning
Threshold clockwise raises the threshold and increases sensitivity (resulting in more triggering); turning Threshold counter-clockwise decreases sensitivity.
Noise Raises or lowers the amount of RFI/static
noise mixed in with the signal (independent of
the Beep/Squelch or Threshold controls). Turning Noise to the right adds a more constant noise
“bed” behind the Beep/Squelch effect; turning
Noise to the left decreases the ambient noise, resulting in sharper Beep/Squelch cut-in.
Beep/Squelch Sets the voice mode between Beep
(NASA-style radio beep) and Squelch (noise
burst).
Chapter 52: Cosmonaut Voice
267
Accessing Additional
Cosmonaut Voice Controls
On-Screen
Cosmonaut Voice also provides a
Beep/Squelch Level control to set the balance of
the generated noise and dry signal.
Beep/Squelch level can be adjusted on-screen by
editing Pro Tools breakpoint automation data.
To access Beep/Squelch level on-screen:
Click the Plug-In Automation button in the
Plug-In window to open the Plug-In Automation
window.
1
In the list of controls at the left, select B/S
Level and click Add (or, just double-click the desired control in the list). Repeat to access and
enable additional controls.
2
Click OK to close the Plug-In Automation window.
3
4
In the Edit window, do one of the following:
• Click the Track View selector and select B/S
Level from the Cosmonaut Voice sub-menu.
– or –
• Reveal an Automation lane for the track,
click the Automation Type selector and select B/S Level from the Cosmonaut Voice
sub-menu.
Edit the breakpoint automation for the enabled control.
5
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Audio Plug-Ins Guide
Accessing Cosmonaut Voice
Controls from a Control Surface
When using a control surface, all plug-in parameters are available whenever the plug-in is focused. You only need to enable plug-in automation (as described previously) if you want to
record your adjustments as breakpoint automation.
To access the Beep/Squelch level from a control
surface:
Focus the Cosmonaut Voice plug-in on your
control surface. All available parameters are
mapped to encoders, faders, and switches.
1
Adjust the control currently targeting the desired parameter.
2
To automate your adjustments, be sure to
enable automation for that parameter as described above. See the Pro Tools Reference
Guide for complete track automation instructions.
Chapter 53: Chorus
Chorus is an AudioSuite plug-in that adds a
shimmering quality to audio material by combining a time-delayed, pitch-shifted copy of an
audio signal with itself.
Chorus Controls
The Chorus plug-in provides the following controls:
Gain Adjusts the input volume of the chorus to
prevent clipping or increase the level of the processed signal. This slider is set to a default of
+3 dB. If your source audio has been recorded
very close to peak level, this +3 dB default setting could cause clipping. Use this control to reduce the input level.
Chorus plug-in
The Chorus plug-in was formerly called
D-fx Chorus. It is fully compatible with all
settings and presets created for D-fx Chorus.
Selecting the Sum Inputs button sums the dry
input signals (mono or stereo) before processing
them. The dry signal then appears in the center
of the stereo field and the wet, effected signal
will be output in stereo.
When the Sum Inputs button is selected, the
LFO waveform on the right channel is automatically phase inverted to enhance the mono-stereo effect.
Sum Inputs button
Chapter 53: Chorus
269
Mix Adjusts the balance between the effected
signal and the original signal and controls the
depth of the effect. Mix is adjustable from 0% to
100%.
Low Pass Filter Controls the cutoff frequency of
the Low Pass Filter. Use this to attenuate the
high frequency content of the feedback signal.
The lower the setting, the more high frequencies
are removed from the feedback signal.
The range of the Low Pass Filter is 20 Hz to
19.86 kHz, with a maximum value of Off (which
effectively means bypass).
Delay Sets the delay time between the original
signal and the chorused signal. The higher the
setting, the longer the delay and the wider the
chorusing effect. Delay is adjustable from 0–20
milliseconds.
LFO Rate Adjusts the rate of the LFO (low frequency oscillator) applied to the delayed signal
as modulation. The higher the setting, the more
rapid the modulation. You can select either a
sine wave or a triangle wave as a modulation
source, using the LFO Waveform selector.
LFO Width Adjusts the intensity of the LFO applied to the delayed signal as modulation. The
higher the setting, the more intense the modulation. Use the LFO Waveform selector to select a
sine or a triangle wave as a modulation source.
Feedback Controls the amount of feedback ap-
plied from the output of the delayed signal back
into its input. Negative settings provide a more
intense effect.
LFO Waveform Selects a sine wave or triangle
wave for the LFO. This affects the character of
the modulation. The sine wave has a gentler
ramp and peak than the triangle wave.
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Audio Plug-Ins Guide
Chapter 54: Flanger
Flanger is an AudioSuite plug-in that animates
and adds a swirling, moving quality to audio
material by combing a time-delayed copy of an
audio signal with itself.
Flanger Controls
The Flanger uses a through-zero flanging algorithm that results in a tape-like flanging effect.
This technique delays the original dry signal by
256 samples, then modulates the delayed signal
back and forth in time in relation to the dry signal, passing through its zero point on the way.
Gain Adjusts the input volume of the flanger to
prevent clipping or increase the level of the processed signal. This slider is set to a default of
+3 dB. If your source audio has been recorded
very close to peak level, this +3 dB default setting could cause clipping. Use this control to reduce the input level.
The Flanger plug-in provides the following controls:
Selecting the Sum Inputs button sums the dry
input signals (mono and stereo) before processing them. The dry signal then appears in the
center of the stereo field and the wet, effected
signal will be output in stereo.
Flanger plug-in
The Flanger plug-in was formerly called
D-fx Flanger. It is fully compatible with all
settings and presets created for D-fx
Flanger.
When the Sum Inputs button is selected, the
LFO waveform on the right channel is phase inverted to enhance the mono-stereo effect.
Mix Adjusts the balance between the effected
signal and the original signal and controls the
depth of the effect. Mix is adjustable from 0% to
100%.
High Pass Filter Controls the cutoff frequency of
the high pass filter. Use this to attenuate the frequency content of the feedback signal and the
frequency response of the flanging. The higher
the setting, the more low frequencies are removed from the feedback signal.
Chapter 54: Flanger
271
LFO Rate Adjusts the rate of the LFO (low frequency oscillator) applied to the delayed signal
as modulation. The higher the setting, the more
rapid the modulation. You can select either a
sine wave or a triangle wave as a modulation
source, using the LFO Waveform selector.
LFO Width Adjusts the intensity of the LFO applied to the delayed signal as modulation. The
higher the setting, the more intense the modulation.
Feedback Controls the amount of feedback ap-
plied from the output of the delayed signal back
into its input. Negative settings provide a more
intense effect.
LFO Waveform Selects a sine wave or triangle
wave for the LFO. This affects the character of
the modulation. The sine wave has a gentler
ramp and peak than the triangle wave.
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Chapter 55: Moogerfooger Lowpass Filter
The Moogerfooger Lowpass Filter features a
2-pole/4-pole variable resonance filter with envelope follower and is available in TDM, RTAS,
and AudioSuite formats. Use it to achieve classic
60s and 70s sounds on bass and electric guitar,
or just dial in some warm, fat analog resonance
when you need it.
Moogerfooger Low Pass Filter
How the Moogerfooger Lowpass Filter Works
With the invention of the MOOG ® synthesizer in
the 1960s, Bob Moog started the electronic music revolution. A direct descendent of the original MOOG Modular synthesizers, the Moogerfooger Lowpass Filter provides two classic
MOOG modules: a Lowpass Filter and an Envelope Follower.
A low pass Filter allows all frequencies up to a
certain frequency to pass, and cuts frequencies
above the cutoff frequency. It removes the high
frequencies from a tone, making it sound more
mellow or muted. The Moogerfooger Lowpass
Filter contains a genuine four-pole lowpass filter. We say “genuine” because the four-pole filter—a major part of the “MOOG Sound” of the
60s and 70s—was first patented by Bob Moog in
1968! Bob worked directly with Bomb Factory to
ensure that the digital version preserved all the
character, nuances, and personality of his original classic analog design.
Chapter 55: Moogerfooger Lowpass Filter
273
“Envelope”
of the sound
Time
Audio waveform
of the sound
Audio waveform of a musical sound
Time
Envelope signal of the same sound
Envelope
Moogerfooger Lowpass Filter
Controls
Envelope Section
Amount The Amount knob determines how
much the envelope varies the filter. When the
knob is counterclockwise, the envelope signal
has no effect on the filter. When the knob is fully
clockwise, the envelope signal opens and closes
the filter over a range of five octaves.
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Audio Plug-Ins Guide
Smooth/Fast The Smooth/Fast switch deter-
mines how closely the envelope tracks the loudness of the input signal. Some sounds (like guitar chords) have long, rough envelopes, and
often sound better with less dramatic changes in
the filter. Other sounds (like bass or snare
drum) are quick and sharp, and sound great
when the filter closely tracks their attack.
Mix The Mix control blends the original input
signal with the filtered signal. Use it to get any
mixture of filtered and unfiltered sound.
Filter Section
Control the filter using the Cutoff and Resonance knobs and the 2-Pole/4-Pole switch.
Cutoff Cutoff opens and closes the filter. Turned
counterclockwise, fewer high frequencies pass
through the filter. Turned clockwise, more high
frequencies pass.
Gain
An Envelope Follower tracks the loudness contour, or envelope, of a sound. Think of it like
this: each time you play a note, the envelope
goes up and then down. The louder and harder
you play, the higher the envelope goes. In the
Moogerfooger Lowpass Filter, the Envelope Follower drives the cutoff frequency of the Lowpass
Filter. Since the envelope follows the dynamics
of the input, it “plays” the filter by sweeping it
up and down in response to the loudness of the
input signal.
2-Pole
4-Pole
Frequency
The 2/4 pole switch selects the filter slope
Resonance Resonance changes the way the filter
sounds. At low resonance, low frequencies come
through evenly. At high resonance, frequencies
near the cutoff frequency are boosted, creating a
whistling or vowel-type quality. When resonance is maxed out, the filter oscillates and produces its own tone at the cutoff frequency. This
oscillation interacts with other tones as they go
through the filter, producing the signature
Moog sound.
2-Pole/4-Pole The 2-Pole/4-Pole switch selects
whether the signal goes through half the filter
(2-pole) or the entire filter (4-pole). 2-pole is
brighter, while 4-pole has a deeper, mellow
quality.
Drive The Drive control sets the input gain. Use
it to adjust the input to the filter and envelope
follower for desired impact.
LED Indicators
Three LEDs down the center of the unit provide
visual feedback.
Level Level glows green when signal is present
to the envelope circuit.
Env Env (envelope) glows redder in response to
the envelope tracking of the input.
Bypass Bypass glows either red (bypassed) or
green (not bypassed) to show whether or not the
effect is in the signal path.
Moogerfooger Lowpass Filter
Tips and Tricks
Auto Wah Using an External LFO
Try inserting an LFO ahead of the Moogerfooger
Lowpass Filter to produce a cool “auto wah” effect. Or use Voce Spin’s rotating speaker for
even trippier sounds!
Chapter 55: Moogerfooger Lowpass Filter
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Audio Plug-Ins Guide
Chapter 56: Moogerfooger 12-Stage Phaser
The Moogerfooger 12-Stage Phaser combines a
6- or 12-stage phaser with a wide-ranging variable LFO and is available in TDM, RTAS, and
AudioSuite formats. Start with subtle tremolo or
radical modulation effects, then crank the distortion and resonant filters for unbelievable
new tones—all featuring classic MOOG ® sound.
How the Moogerfooger 12-Stage Phaser Works
The Moogerfooger 12-Stage Phaser offers 6 or 12
stages of MOOG resonant analog filters. Unlike
the Lowpass Filter, however, the filters are arranged in an allpass configuration.
Time
Low Pass
Filter
1
Time
Cutoff
Frequency
Resonant
Filter
1
Center
Frequency
Time
1
Moogerfooger 12-stage Phaser
5-Stage
Phaser
Mid-Shift
Frequency
Different types of filters
A phaser works by sweeping the mid-shift frequency of the filters back and forth. As this happens, the entire frequency response of the output moves back and forth as well. The result is
the classic phaser “whooshing” sound as different frequency bands of the signal are alternately
emphasized and then attenuated.
Chapter 56: Moogerfooger 12-Stage Phaser
277
A sweep control allows you to adjust the range of
the frequency shift. And, keeping in the spirit of
the MOOG modular synthesizers, an integrated
LFO allows you to modulate the sweep control,
allowing for extreme effects.
Gain
1
Moogerfooger 12-Stage Phaser
Controls
Frequency
Mid-Shift Frequency
LFO Section
Responses of a phaser with high resonance
Control the LFO using the Amount and Rate
knobs and the Lo/Hi selector switch.
Amount Amount varies the depth of phaser
modulation, from barely perceptible at the full
counterclockwise position, to the full sweep
range of the phaser (full clockwise or “Kill” setting).
Sweep Sweep adjusts the center frequency point
of the filters. Use it in conjunction with Amount
to control the frequencies affected by the
phaser.
Gain
Mid-shift frequency
moves
1
Rate Rate determines how fast the LFO oscil-
lates. The LFO light blinks to give a visual indication of the LFO rate.
Lo/Hi The Lo/Hi switch selects the range of the
Rate control. When the switch is Lo, the Rate
control varies from 0.01 Hz (one cycle every
hundred seconds) to 2.5 Hz (2.5 cycles every
second). When the switch is Hi, the Rate control
varies from 2.5 Hz (2.5 cycles every second) to
250 Hz (two hundred fifty cycles per second).
With such a wide range of rates available, obviously you’ll need to adjust Rate after you flick
the Lo/Hi switch to get the sound you desire.
Sweep adjusts the center frequency point
Drive
The Drive control sets the input gain.
Phaser Section
LED Indicators
Control the Phaser with the Sweep and Resonance knobs and the 6-Stage/12-Stage switch.
Three LEDs provide visual feedback.
Resonance Resonance adjusts the feedback of
the analog filters. As you add more resonance,
the peaks caused by the filters get sharper and
more noticeable.
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Audio Plug-Ins Guide
Level Level glows green when signal is present.
LFO LFO blinks to show the LFO rate.
Bypass Bypass glows either red (bypassed) or
green (not bypassed) to show whether or not the
effect is in the signal path.
Moogerfooger 12-Stage Phaser
Tips and Tricks
More Harmonics = More Fun
The richer the harmonic content of the sound,
the more there is to filter and sweep. Try adding
distortion using the SansAmp PSA-1 before the
phaser–it’s a cool variation on the common signal path used when putting a phaser in front of a
guitar amp.
Aggressive. Extreme.
Dr. Moog apparently took these mantras of early
21st Century recording science to heart when he
designed the Rate knob on his phaser. Flick the
Rate switch to Hi and let the party begin. Try
muting a track and mixing in bits of extremely
phase-swept material.
It’s an Effect—Play with It
All the controls on the Moogerfooger 12-Stage
Phaser are fully independent of one another.
This means you can set them in any combination
that you wish. There is no such thing as a
“wrong” combination of settings, so you can experiment all you like to find new, exciting effects for your music.
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Chapter 57: Moogerfooger Ring Modulator
The Moogerfooger Ring Modulator that provides a wide-range carrier oscillator and dual
sine/square waveform LFO and is available in
TDM, RTAS, and AudioSuite formats. Add motion to rhythm tracks and achieve radical lo-fidelity textures—you set the limits!
The Carrier Oscillator is a wide-range sinusoidal
oscillator. It’s called the Carrier Oscillator because, like the carrier of an AM radio signal, it’s
always there, ready to be modulated by the input.
A Ring Modulator takes two inputs, and outputs
the sum and difference frequencies of the two
inputs. For example, if the first input contains a
500 Hz sine wave, and the second input contains
a 100 Hz sine wave, then the output contains a
600 Hz sine wave (500 plus 100) and a 400 Hz
(500 minus 100) sine wave.
Moogerfooger Ring Modulator
Controls
LFO Section
Moogerfooger Ring Modulator
How the Moogerfooger Ring Modulator Works
Like the Lowpass Filter, the Moogerfooger Ring
Modulator has its roots in the original MOOG
Modular synthesizers. It provides three classic
MOOG modules: a Low Frequency Oscillator, a
Carrier Oscillator, and a Ring Modulator.
Low Frequency Oscillators (or LFOs) create slow
modulations like vibrato and tremolo. The LFO
in the Moogerfooger Ring Modulator is a widerange, dual-waveform (sine/square) oscillator.
Control the LFO using the Amount and Rate
knobs and the Square/Sine waveform selector
switch.
Amount Amount determines the amount of LFO
waveform that modulates the frequency of the
carrier oscillator. When the knob is full counterclockwise, the carrier is unmodulated. Fully
clockwise, the carrier oscillator is modulated
over a range of three octaves.
Rate Rate determines how fast the LFO oscil-
lates, from 0.1 Hz (one cycle every ten seconds)
to 25 Hz (twenty-five cycles per second). The
LFO light blinks to give a visual indication of the
LFO rate.
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281
Sine/Square The Square/Sine switch selects ei-
ther a square or sine waveform. The square wave
produces trill effects, whereas the sine waveform produces vibrato and siren effects.
Modulator Section
The Carrier Oscillator is controlled by the Frequency knob and the Low/High switch.
Frequency Knob Operating at the selected frequency, the carrier oscillator provides one input
to the ring modulator, with the other coming
from the input signal.
Lo In the Lo position, the Frequency knob
ranges from 0.5 Hz to 80 Hz.
Hi In the High position, the Frequency knob
ranges from 30 Hz to 4 kHz.
Mix The Mix control blends the input signal and
the Ring Modulator output. You hear only the
input signal when the knob is counterclockwise,
and only the ring modulated signal with the
knob fully clockwise .
Drive
The Drive control sets the input gain.
LED Indicators
Three LEDs provide visual feedback.
Level Level glows green when signal is present.
LFO LFO blinks to show the LFO rate.
Bypass Bypass glows either red (bypassed) or
green (not bypassed) to show whether or not the
effect is in the signal path.
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Moogerfooger Ring Modulator
Tips and Tricks
A Little Goes a Long Way
You’ll discover tons of great uses for the
Moogerfooger Ring Modulator through experimentation. But don’t forget to try using it in
subtle ways, adding “just a hint” to harshen up
or add a metallic quality to individual tracks
buried in the mix. Almost all the great MOOG
sounds feature subtle, clever uses of Ring Modulation.
Chapter 58: Reel Tape Flanger
Reel Tape Flanger is part of the Reel Tape suite
of tape-simulation effects plug-ins and is available in TDM, RTAS, and AudioSuite formats.
Reel Tape Flanger simulates a tape machine
flanging effect, modeling the frequency sweep
and “crossover” comb-filtering effects that can
result when the flanger variable delay is adjusted. It also reproduces the frequency response, noise, wow and flutter, and distortion
characteristics of analog tape recording.
The two machines are fed an input signal in parallel, and the output of the machines is then
mixed. When the variable delay on the second
machine is changed at a constant rate (using an
LFO), the resulting frequency cancellations
cause a periodic phasing of the original signal.
The use of a fixed delay on the first machine
makes it possible to adjust the variable delay on
the second machine to pass the “zero” point (to
a delay value less than the fixed delay), resulting
in phase cancellation (or the “crossover” flanging effect).
Reel Tape Flanger automatically applies tape
saturation effects that correspond to the following control settings in Reel Tape Saturation:
• Speed: 15 ips
• Bias: 0.0 dB
Reel Tape Flanger
How Reel Tape Flanger Works
For years, engineers have relied on analog tape
to add a smooth, warm sound to their recordings. When driven hard, tape responds with gentle distortion rather than abrupt clipping as in
the digital domain. Magnetic tape also has a frequency-dependent saturation characteristic that
can lend punch to the low end, and sweetness to
the highs.
• Cal Adjust: +9 dB
You can use the BPM Sync feature to synchronize the Reel Tape Flanger effect to the current
tempo of the Pro Tools session.
Reel Tape Flanger can be placed on mono, stereo, or multichannel tracks.
Reel Tape Flanger models a classic tape flanging
setup with two analog tape machines and a
mixer, where one tape machine has a fixed delay
and the other has a continuously variable delay.
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283
Reel Tape Common Controls
All Reel Tape plug-ins share the following
controls:
Drive
Drive controls the amount of saturation effect
by increasing the input signal to the modeled
tape machine while automatically compensating
by reducing the overall output. Drive is adjustable from –12 dB to +12 dB, with a default value
of 0 dB.
Tape Formula
The Tape Formula control lets you select either
of two magnetic tape formulations emulated by
the plug-in, each with its own saturation characteristics:
Classic Emulates the characteristics of
Ampex 456, exhibiting a more pronounced saturation effect.
Hi Output Emulates the characteristics of
Quantegy GP9, exhibiting a more subtle saturation effect.
Output
Output controls the output signal level of the
plug-in after processing. Output is adjustable
from –12 dB to +12 dB, with a default value of
0 dB.
Tape Machine
The Tape Machine control lets you select one of
three tape machine types emulated by the plugin, each with its own sonic characteristics:
US Emulates the audio characteristics of a
3M M79 multitrack tape recorder.
Swiss Emulates the audio characteristics of a
Studer A800 multitrack tape recorder.
Lo-Fi Simulates the effect of a limited-bandwidth analog tape device, such as an outboard
tape-based echo effect.
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Reel Tape Flanger Controls
In addition to the Drive, Output, Tape Machine,
and Tape Formula controls, Reel Tape Flanger
has the following controls:
Range
The Range control adjusts the overall magnitude
of the variable delay, which determines the offset between the two modeled tape machines. A
center or “zero” setting results in no offset.
Range is continuously adjustable from –20 ms
to +20 ms, and is divided into two types of effects: flanging and automatic double tracking.
Flange Range settings within the narrow center
band around “zero” simulate tape flanging, with
a phase cancellation effect as the variable delay
crosses the “zero” point.
LFO Rate
Feedback
The Feedback control adds a short delay to the
flanged signal. Feedback amount is adjustable
from 0 to 100 percent, with a default value of 0
percent. (This is not the same feedback effect as
on an electronic flanger or delay.
LFO Depth
Wow/Flutter
zero point
Operation with “Flange” Range setting (no offset)
ADT (Artificial Double Tracking) Range settings
outside the narrow center band simulate artificial double tracking, in which the variable delay
does not cross the “zero” point. This varying delay creates a unique doubling effect, essentially
an analog precursor to chorusing. (You can hear
ADT-type effects on many classic analog recordings, such as those of the Beatles or Led Zeppelin.)
LFO Rate
LFO Depth
+20 ms
The Wow/Flutter control adjusts the amplitude
of the variable delay tape machine’s wow and
flutter, or the amount of fluctuation in tape
speed. A higher setting results in wider fluctuations in speed. A lower setting results in narrower fluctuations in speed. Wow/Flutter is adjustable from 0 to 1 percent, with a default value
of 0.03 percent.
Rate
The LFO Rate control adjusts the rate of change
in the variable delay. A higher setting results in
faster fluctuations in speed. A lower setting results in slower fluctuations in speed. LFO Rate is
adjustable from 0.05 Hz to 5 Hz, with a default
setting of 0.14 Hz.
You can set the LFO Rate control to synchronize
to the tempo of the current Pro Tools session.
See “Synchronizing Reel Tape Flanger to Session
Tempo” on page 286.
Depth
zero point
-20 ms
Operation with “ADT” Range setting (positive offset)
The LFO Depth control adjusts the amplitude of
the change in variable delay. A higher setting results in wider fluctuations in speed. A lower setting results in narrower fluctuations in speed.
LFO Depth is adjustable from 0 to 100 percent,
with a default value of 65 percent.
When the LFO Depth control is set to zero,
you can still achieve a “manual” flanging or
ADT effect by varying the Range control.
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285
Mix
The Mix control adjusts the amount of fixed delay signal mixed with the variable delay signal in
the final output of the plug-in. The default Mix
value is adjustable from –100 (all fixed delay
signal) to +100 (all variable delay signal) percent, with a default value of 0 (50% fixed delay,
50% variable delay signals).
Invert
(Plug-In Automation Playlist or Control
Surface Access Only)
The Invert parameter inverts the polarity of the
signal coming from the variable delay tape machine, so that complete audio cancellation occurs when the flanger effect crosses the zero
point. The default setting for the Invert parameter is Off.
This parameter is accessible only from the plugin automation playlist or from a supported control surface.
Settings for this parameter are saved with
plug-in presets. If you use a preset for the
TDM, RTAS or AudioSuite version of this
plug-in, any settings for this parameter
will be active.
Noise
(Plug-In Automation Playlist or Control
Surface Access Only)
The Noise parameter controls the level of simulated tape hiss that is added to the processed signal. Noise is adjustable from Off (–INF) to
–24 dB, with a default value of Off.
This parameter is accessible only from the plugin automation playlist or from a supported control surface.
Settings for this parameter are saved with
plug-in presets. If you use a preset for the
TDM, RTAS or AudioSuite version of this
plug-in, any settings for this parameter
will be active.
Synchronizing Reel Tape
Flanger to Session Tempo
You can set the LFO Rate in Reel Tape Flanger to
synchronize to the session tempo (in beats per
minute).
To synchronize the LFO Rate control setting to the
session tempo:
1 In the BPM Sync section, click the On button.
The Tempo/Rate display changes to synchronize
with the current session tempo.
Tempo/Rate
display
On
button
Note Value
display
Dot
button
Triplet
button
BPM Sync controls
To set a rhythmic LFO rate, click the Note
Value to choose from the available note values
(whole, half, quarter, eighth, sixteenth, or
thirty-second note)
2
To adjust the rhythm further, do any of the following:
3
• To enable triplet rhythm delay timing, click
the Triplet (“3”) button so that it is lit.
• To set a dotted rhythm delay value, click the
Dot (“.”) button so that it is lit.
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Reel Tape Flanger Tips
Reel Tape Flanger Presets
 To achieve a flanging effect, set the Range control within the “Flange” range and adjust the
LFO Depth control to a value that is greater than
the offset (so that the variable delay crosses the
“zero” point.)
12-String Moderate-depth ADT setting that
 To achieve an ADT (doubling) effect, set the
Range control within either of the “ADT” ranges
and adjust the LFO Depth control to a value that
is smaller than the offset (so that the variable
delay does not cross the “zero” point).
Flutter Extreme Wow/Flutter setting with flang-
 To achieve a manual flanging effect, set the
LFO Depth control to 0 and vary (or automate)
the Range control within the “Flange” range. For
fine control, hold Control (Windows) or Command (Mac) while varying the Range control.
To add complexity to flanging or ADT effects,
turn up the Wow/Flutter control to introduce
more fluctuation in the variable delay.

 Use Reel Tape Flanger in a send/return configuration to mix the dry signal with an aggressively driven, flanged signal to control the
amount of “grunge” in the final mix.
works well with acoustic guitar sounds
Flutter Flange Moderate-depth flange setting
with Wow/Flutter
ing turned off and a Mix setting that passes only
the variable delay
Manual Flange Settings with LFO Depth set to
zero, ready for manual flanging by adjusting or
automating the Range control
Slow Flange High Depth setting combined with
slow LFO Rate, suitable for flanging vocals
Vocal ADT Settings for creating doubling effect
without flanging “crossover” effect, suitable for
vocals
Vocal Walrus Drive-boosted settings for extreme
vocal doubling effect
Wobble A high LFO Rate setting combined with
a Mix setting that passes only the variable delay.
Works well on sustained parts.
 When you start playback, the LFO sweep always starts at the bottom of the cycle, so each
time you start playback from the same location
(for example, at a bar line), the effect will be applied in the same way.
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Chapter 59: Sci-Fi
Sci-Fi part of the D-Fi family of plug-ins, providing analog synthesizer-type effects. It is
available in TDM, RTAS, and AudioSuite formats. Sci-Fi features effects that include:
• Ring modulation
• Frequency modulation
• Variable-frequency, positive and negative
resonator
• Modulation control by LFO, envelope follower, sample-and-hold, or trigger-andhold
Sci-Fi is designed to mock-synthesize audio by
adding effects such as ring modulation, resonation, and sample & hold, which are typically
found on older, modular analog synthesizers.
Sci-Fi is ideal for adding a synth edge to a track.
Sci-Fi can be used as either a real-time TDM or
RTAS plug-in or as a non-real-time AudioSuite
plug-in.
Purposely Degrading Audio
Contemporary music styles, especially hip-hop,
make extensive use of retro instruments and
processors such as vintage drum machines, samplers, and analog synthesizers. The low bit-rate
resolutions and analog “grunge” of these devices
are an essential and much-desired part of their
sonic signatures. That is why Avid created D-Fi.
The D-Fi suite of plug-ins combines the best of
these instruments of the past with the flexibility
and reliability of the Pro Tools audio production system. The result is a set of sound design
tools that let you create these retro sounds without the trouble and expense of resampling audio
through 8-bit samplers or processing it through
analog synthesizers.
The multichannel TDM version of the
Sci-Fi plug-in is not supported at 192 kHz.
Use the multi-mono TDM or RTAS version
instead.
Sci-Fi
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289
Sci-Fi Controls
Sci-Fi Input Level
Input Level attenuates signal input level to the
Sci-Fi processor. Since some Sci-Fi controls
(such as Resonator) can cause extreme changes
in signal level, adjusting the Input Level is particularly useful for achieving unity gain with the
original signal level. The range of this control is
from –12 dB to 0 dB.
Sci-Fi Effect Types
Sci-Fi provides four different types of effects:
Ring Mod Is a ring modulator—which modu-
lates the signal amplitude with a carrier frequency, producing harmonic sidebands that are
the sum and difference of the frequencies of the
two signals. The carrier frequency is supplied by
Sci-Fi itself. The modulation frequency is determined by the Effect Frequency control. Ring
modulation adds a characteristic hard-edged,
metallic sound to audio.
the effect, producing a hollower sound than Resonator+. The Resonator can be used to produce
metallic and flanging effects that emulate the
sound of classic analog flangers.
Sci-Fi Effect Amount
Effect Amount controls the mix of the processed
sound with the original signal. The range of this
control is from 0–100%.
Sci-Fi Effect Frequency
Effect Frequency controls the modulation frequency of the ring modulator and resonators.
The frequency range is dependent on the effect
type. For Ring Mod, the frequency range of this
control is from 0 Hz to 22.05 kHz. For Freak
Mod, the frequency range is from 0 Hz to
22.05 kHz. For Resonator+, the frequency range
is from 344 to 11.025 kHz. For Resonator–, the
frequency range is from 172 Hz to 5.5 kHz.
You can also enter a frequency value using keyboard note entry.
Freak Mod Is a frequency modulation processor
To use keyboard note entry:
that modulates the signal frequency with a carrier frequency, producing harmonic sidebands
that are the sum and difference of the input signal frequency and whole number multiples of
the carrier frequency. Frequency modulation
produces many more sideband frequencies than
ring modulation and an even wilder metallic
characteristic. The Effect Frequency control determines the modulation frequency of the Freak
Mod effect.
1
Start-click (Windows) or Control-click (Mac)
the Effect Frequency slider to display the pop-up
keyboard.
2 Select the note on the keyboard that you want
for the Effect Frequency.
Resonator+ and Resonator– Add a resonant fre-
quency tone to the audio signal. This frequency
is determined by the Effect Frequency control.
The difference between these two modules is
that Resonator– reverses the phase (polarity) of
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Audio Plug-Ins Guide
Sci-Fi Keyboard Note Entry
Sci-Fi Mod Type Controls
Trigger+Hold Trigger and hold modulation is
The four Mod Type buttons determine the type
of modulation applied to the frequency of the selected effect. Depending on the type of modulation you select here, the sliders below it will
change to provide the appropriate type of modulation controls. If the Mod Amount is set to 0%,
no dynamic modulation is applied to the audio
signal. The Effect Frequency slider then becomes the primary control for modifying the
sound.
similar to sample and hold modulation, with one
significant difference: If the input signal falls
below the threshold set with the Mod Threshold
control, modulation will not occur. This provides interesting rhythmic effects, where modulation occurs primarily on signal peaks. Modulation will occur in a periodic, yet random way
that varies directly with peaks in the audio material. Think of this type of modulation as having the best elements of both sample and hold
modulation and with an envelope follower.
LFO Produces a low-frequency triangle wave as
a modulation source. The rate and amplitude of
the triangle wave are determined by the Mod
Rate and Mod Amount controls, respectively.
Sci-Fi Mod Amount and Mod
Rate Controls
Envelope Follower Causes the selected effect to
dynamically track the input signal by varying
with the amplitude envelope of the audio signal.
As the signal gets louder, more modulation occurs. This can be used to produce a very good
automatic wah-wah-type effect. When you select
the Envelope Follower, the Mod Amount slider
changes to a Mod Slewing control. Slewing provides you with the ability to smooth out extreme
dynamic changes in your modulation source.
This provides a smoother, more continuous
modulation effect. The more slewing you add,
the more gradual the changes in modulation will
be.
If you select Trigger+Hold as a modulation type,
the Mod Rate slider changes to a Mod Threshold
slider, which is adjustable from –95 dB to 0 dB.
It determines the level above which modulation
occurs with the trigger and hold function.
Sample+Hold Periodically samples a random
pseudo-noise signal and applies it to the effect
frequency. Sample and hold modulation produces a characteristic random stair-step modulation. The sampling rate and the amplitude are
determined by the Mod Rate and Mod Amount
controls, respectively.
These two sliders control the amplitude and frequency of the modulating signal. The modulation amount ranges from 0% to 100%. The modulation rate, when LFO or Sample+Hold are
selected, ranges from 0.1 Hz to 20 Hz.
If you select Envelope Follower as a modulation
type, the Mod Rate slider changes to a Mod
Slewing slider, which is adjustable from 0% to
100%.
Sci-Fi Output Meter
The Output Meter indicates the output level of
the processed signal. Note that this meter indicates the output level of the signal—not the input level. If this meter clips, the signal may have
clipped on input before it reached Sci-Fi. Monitor your send or insert signal levels closely to
prevent this from happening.
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Audio Plug-Ins Guide
Chapter 60: TL EveryPhase
TL EveryPhase is an 18-stage analog modeled phaser effects plug-in designed to reproduce classic
phaser effects as well as creating exciting new sounds. It is available in TDM and RTAS formats.
TL EveryPhase plug-in
TL EveryPhase Overview
This section provides an overview of traditional
phasers and the TL EveryPhase phaser.
Traditional Analog Phasers
The phaser (or phase shifter) is a classic sound
effect often heard on guitars or synthesizers.
The sweeping sound of a phaser can vary from
subtle modulation and tremolo on a delicate
guitar track to the most extreme filtered feedBack. Traditionally, phasers were analog effects
devices. Analog phasers delivered the benefits of
a smooth analog sound, but like many analog
devices were often unreliable and introduced
unwanted noise and hum.
A phaser functions by moving a portion of the
incoming audio out of phase and then adding
the processed audio back to the original signal.
Each stage of a multiple stage phaser can be
thought of as a narrow band or notch of the frequency range which is filtered out. As the frequency is adjusted, the classic sweeping phaser
sound is heard.
TL EveryPhase
TL EveryPhase uses proprietary DSP algorithms
to deliver the classic analog phaser sound in digital form, with the added benefits of extensive
synchronization and automation options.
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293
The following figure shows the different modules of TL EveryPhase and how they interact
with the audio signal.
TL EveryPhase Controls
The TL EveryPhase interface is divided into the
following sections of controls:
• Meter (see “TL EveryPhase Meter Section”
on page 294)
• Phaser (see “TL EveryPhase Phaser Section” on page 295)
• Modulation (see “Modulation Section” on
page 296)
• LFO (see “LFO Section” on page 296)
• Tempo (see “Tempo Controls” on page 298)
TL EveryPhase signal flow, processing, and controls
The modulation of the phaser algorithm in
TL EveryPhase can be controlled by a low frequency oscillator (LFO) or by the envelope of an
audio signal using the built-in envelope detector. The Depth control switches TL EveryPhase
between phasing in opposite and identical phasing modes, and feedBack can be taken from any
stage of the phaser by adjusting the Resonance
control.
• Envelope (see “Envelope Section” on
page 298)
TL EveryPhase Meter Section
There are two meters available, and Output meter and a Modulation meter.
Output and Modulation meters
TL EveryPhase provides controls to enable the
LFO to be synchronized to the current tempo of
the Pro Tools session. A variety of LFO triggers
are also provided to ensure that a phase effect
can be created to match the timing of any audio
signal.
The envelope detector in TL EveryPhase provides several options to control the phasing directly from an audio signal. Firstly, the envelope
detector can be driven by the audio of the current track or audio from a side-chain input. The
envelope detector can drive the phaser modulation directly by selecting ENV for the Source in
the Modulation section. Alternatively the envelope detector can be used as a trigger for the LFO
by selecting Envelope under Triggers in the LFO
section.
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Audio Plug-Ins Guide
Output
The Output meter displays the amplitude of the
outgoing audio. In mono mode, a single meter
bar is shown. In mono to stereo and stereo
modes, two meter bars are shown with the left
channel at the top of the meter display. In 5.1
mode, six channels are shown, in the order L C R
Ls Rs LFE from the top of the meter display. The
red clip indicator indicates a channel has
clipped. The clip indicator for each channel can
be cleared by clicking on it.
Modulation
The Modulation meter displays several items at
once. First, the range of phaser sweep set by the
Modulation Width and Manual controls is indicated by the shaded background area. The movement of the phaser itself is indicated by one or
two scanning bars. When TL EveryPhase is instantiated on a mono, stereo, or 5.1 track, a single bar is shown in this meter. When instantiated on a mono track as a mono to stereo plugin, two scanning bars are shown.
the Stages slider. When Resonance set to any
other value, feedBack is taken from the stage indicated by the Resonance slider and a different
feedBack tone is created.
FeedBack
The FeedBack slider feeds the output signal of
TL EveryPhase back into the input, creating a
resonant or singing tone in the phaser when set
to maximum.
Depth
TL EveryPhase Phaser Section
Phaser section
The Depth slider adjusts the depth of the
notches in the phased signal. When set to zero,
TL EveryPhase does not phase the audio signal.
Depth can be set to positive or negative values
which allows for two separate types of phasing
to occur. When Depth is positive, the notches
occur at frequencies that are at opposite phase,
which is a common feature of many analog
phasers. When Depth is negative, the notches
occur at frequencies that have identical phase.
The sound quality of these two types of phasing
can be remarkably different.
Input
Output
The Input slider lets you cut or boost of the input signal level from –24 dB to +12 dB.
The Output slider lets you cut or boost of the
output signal level from –24 dB to +12 dB.
Stages
The Stages slider sets number of phaser stages
from 2 to 18. This changes the character of the
sound as the number of stages controls the number of notches that TL EveryPhase affects.
Resonance
The Resonance slider changes the character of
the feedBack tone created by allowing the feedBack to come from a different stage of the
phaser. When Resonance is set to Norm, feedBack is based on the stage of the phaser set by
Chapter 60: TL EveryPhase
295
Modulation Section
LFO Section
Modulation section
Width
The Width slider determines the amplitude of
the modulation sweep. This is displayed graphically in the modulation meter.
LFO section
When the Modulation section’s Source is set
to the Envelope (ENV), the controls in the
LFO section have no effect on the current
sound.
Manual
The Manual slider offsets the modulation sweep.
This is displayed graphically in the modulation
meter.
Source
Click LFO or ENV to select the source for modulation. When the Source is set to LFO, modulation is controlled by the LFO. When it is set to
Envelope (ENV), modulation is controlled by
the Envelope Detector which listens to the audio
signal. If the side-chain input in the Envelope
section is activated, the side-chain audio is used
instead of the current track.
Direction
Click Up or Down to change the direction of the
modulation.
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Audio Plug-Ins Guide
Rate
The Rate slider adjust the rate of the LFO in
beats per minute. When Link to Tempo is activated, the slider is ignored and the LCD always
displays the current session tempo.
Waveform
Selecting the LFO Waveform
The Waveform selector (Triangle, Ramp, Sine,
etc.) determines the wave shape used by the
LFO. The waveform shape in use is graphically
depicted by the movement of the scanning bars
in the Modulation meter.
LFO Triggers
LFO Triggers
By default, the LFO cycles continuously through
the selected waveform. The LFO can be set to cycle through the selected waveform just once, or
it can be triggered by MIDI Beat Clock, the Envelope, or manually.
Single When the Single trigger is selected, the
LFO will cycle thru the waveform once only and
then stop.
Beat Clock When the Beat Clock trigger is selected, the LFO synchronizes to MIDI Beat
Clock. TL EveryPhase receives Beat Clock signal
every 64th-note. The Duration menu determines
how often the Beat Clock signal triggers TL EveryPhase, ranging from every 16th-note to every
4 bars. When Beat Clock signal is received, the
Beat Clock trigger light blinks brightly. Using
the Beat Clock function enables TL EveryPhase
to produce consistent phasing results, ensuring
that the LFO is always in the same state at each
beat.
In Pro Tools 6.1 and earlier, MIDI Beat
Clock be enabled in Pro Tools. Select MIDI >
MIDI Beat Clock, and enable MIDI Beat
Clock and select TL EveryPhase as a destination.
Envelope When the Envelope trigger is selected,
the LFO is triggered directly by the Envelope detector, which listens to the audio signal. If the
Side-Chain Input selector in the Envelope section is activated, then the side-chain audio signal is used instead. When activated, the Envelope light blinks brighter when an audio signal
is detected. The threshold level can be adjusted
using the Threshold control in the Envelope section.
If the Envelope Detector is completely released
due to previous portions of the audio signal going above threshold, a trigger occurs the next
time the audio goes above the threshold level.
Another trigger will not happen until the Envelope Detector has completely released after the
audio goes below the specified threshold. Thus,
increasing the Release slider will reduce the rate
at which triggers can occur and decreasing the
Release time increases the rate at which triggers
can occur.
Manual When the Manual trigger is selected, the
LFO is triggered manually. This can be especially useful if you want to trigger the LFO using
Pro Tools automation.
With control surfaces and automation, the Manual trigger acts like an on/off switch and triggers
the LFO every time it changes state.
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Tempo Controls
Tempo controls
Link To Tempo
When the Link To Tempo option is enabled, the
LFO rate is set to the Pro Tools session tempo,
and any tempo changes in the session are followed automatically. When Link To Tempo is
enabled, the LFO rate slider is ignored and the
tempo displayed in the LCD always displays the
current session tempo.
The Link To Tempo control is only available
on Pro Tools 6.1 and later. In earlier releases of Pro Tools, manually set the LFO
rate to match the session tempo for the same
effect.
one bar. When Duration is set to 1 beat, the LFO
cycles within the duration of one beat. When
Link to Tempo is activated, the Duration selector sets the LFO rate as a function of the tempo
of the Pro Tools session. The Duration selector
also controls how often the Beat Clock trigger is
activated.
Tempo Display
Tempo Display
The Tempo Display displays the tempo in BPM.
The value in the Tempo Display can also be edited directly by clicking it.
Envelope Section
Duration Selector
Envelope section
When you select Envelope as the Modulation
source, Modulation (as shown in the Modulation Meter) is controlled by the audio signal and
the Envelope Detector section controls.
Selecting Duration
The Duration selector works in conjunction
with the session tempo, LFO rate, and Beat
Clock trigger. By default, Duration is set to 1
bar. At that setting, the LFO cycles once within
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When the Envelope Detector is not in use,
the controls in this section have no effect on
the sound.
Side-Chain Input
Side-Chain Input selector enabled
When the Side-Chain Input selector (the key
icon) is enabled, the audio for the Envelope Detector is taken from the side-chain input rather
than the current track. Select the Side-Chain Input using the Pro Tools key icon at the top of the
plug-in window.
Using TL EveryPhase
This section addresses some common scenarios
in which TL EveryPhase can be used during a
Pro Tools session.
Using TL EveryPhase Presets
TL EveryPhase ships with a wide selection of
factory presets for different phaser sounds. The
following should be noted when using presets:
• Presets which use the Envelope Detector
may need to have the Envelope Threshold,
Attack and Release adjusted appropriately
for the current audio signal.
Threshold
The Threshold slider sets the amplitude level required for the Envelope Detector. The LFO Envelope Detector light blinks brighter when audio
is detected above the threshold.
Attack
• Some presets utilize the Side-Chain Input.
If necessary, ensure that you have a sidechain input assigned, and adjust the Envelope Detector to get the best results.
The Attack slider sets the attack rate of the
Envelope Detector.
• Adjust the input and output levels appropriately for your track to avoid clipping.
Release
The Release slider sets the release rate of the Envelope Detector.
Creating a Single Phased Sound
with TL EveryPhase
A single phased sound (one cycle of the phaser)
can be created using automation of the LFO
manual trigger.
To create a single phased sound:
1
Insert TL EveryPhase on a track.
Select an appropriate LFO Waveform, such as
Ramp.
2
3
Set the Rate to an appropriate value.
Enable the LFO Single trigger so the LFO will
only cycle once.
4
Select the Auto button at the top of the
TL EveryPhase plug-in window.
5
Add LFO Manual Trigger to the automation
list.
6
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Set the Automation mode for the track to Write
or Touch.
7
8
Play the session
At the point where you wish phasing to start,
click on Manual Trigger to start the LFO. The automation for this action will be recorded onto
the track.
9
The Bypass and/or Depth controls can also be
automated to ensure TL EveryPhase does not effect any part of the sound except the specific
section required.
Manually Automating Triggers
If you want the phasing effect of TL EveryPhase
to match an irregular sound (such as a guitar
lead that doesn’t fall on a specific beat), manually automating the LFO Manual Trigger provides an alternative.
You can manually automate the LFO to trigger at
specific points in the session in a similar fashion
to that described above. The following figure
shows a guitar track with automation of the LFO
Manual Trigger at points which match key
phrases in the guitar playing.
Creating a Gradual Phaser Effect
As an alternative to bypassing TL EveryPhase
when an effect isn’t needed, the Depth control
can be automated to introduce and fade out
TL EveryPhase on a track as required.
Adding Other Effects
For different phaser sounds, try using a compressor before or after TL EveryPhase. Other
useful effect plug-ins to try with TL EveryPhase
include distortion, delay, and EQ.
Using TL EveryPhase Beat Clock
Triggers
The Beat Clock trigger lets you trigger the LFO
on specific bars and beats. Using the LFO Duration menu and the Beat Clock trigger, you can
restart the LFO as often as once every 16th-note.
LFO Manual Trigger automation on a guitar track
Alternatively, with an appropriate audio signals,
using the LFO envelope trigger with the correct
threshold settings will trigger the LFO as
needed.
Using the TL EveryPhase SideChain Input
The Side-Chain Input option in TL EveryPhase
lets you direct audio from another track in your
Pro Tools session to the Envelope Detector. This
is achieved by sending the audio from the desired channel to a bus and setting the side-chain
input on TL EveryPhase to the same bus.
This is useful when the tempo and timeline in a
Pro Tools session have been set to match the
music.
Selecting a bus as the Side-Chain Input
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The Side-Chain Input feature lets you control
the TL EveryPhase modulation and LFO using
external audio sources, allowing you to explore
creative possibilities not available with most
phasers.
For example, a side-chain input can be used to
“listen” to a percussion track and create a rhythmic phasing effect on a bass line. This is especially effective in R&B, hip hop and electronic
music.
Consider the following two bar bass line and
drum loop. The bass line is simply a single bass
guitar note which lasts for almost an entire bar.
After starting the transport, adjust the Threshold in the Envelope section until the drum loop
is triggering the Envelope Detector. This is
shown by the Source:Envelope or Envelope trigger light blinking brighter, as well as shown by
the action of the Modulation Meter.
6
7 The Attack and Decay in the Envelope section
can also be adjusted to suit your needs.
The phased bass line is shown below after being
recorded to a separate track. The effect of TL EveryPhase triggered by the drum loop can be seen
in the resulting waveform.
Resulting phased bass line
Bass line and drum loop tracks
The bass line can be phased by the drum loop as
follows:
Instantiate TL EveryPhase on the bass line
track.
1
2
Send the drum loop track to a bus.
3 Set the Side-Chain Input on TL EveryPhase to
listen to the selected bus.
Activate the Side-Chain Input in TL EveryPhase by selecting the key icon in the Envelope
section.
4
On versions of Pro Tools prior to 7.0, RTAS
plug-ins do not provide side-chain processing when used on TDM systems. Use the
TDM version of TL EveryPhase if you require side-chain processing on a TDM system.
For more information on using the SideChain Input, see the Pro Tools Reference
Guide.
5 The Side-Chain audio can modulate the audio
directly by selecting Source:Envelope in the
Modulation section. Alternatively, the SideChain Input can be used to trigger the LFO by selecting the Envelope trigger in the LFO section.
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TL EveryPhase Tips and Tricks
Can’t get the perfect phaser sound? Try some of
these ideas!
• Try a preset. TL EveryPhase includes over
120 presets in eight categories. The categories are merely suggestions—a preset created for guitar may have just the sound you
need for vocals.
• Adjust the Depth. Setting Depth to positive
or negative values allows for two separate
types of phasing to occur. When Depth is
positive, the phaser notches occur at frequencies that are at opposite phase, which
is a common feature of many analog
phasers. When Depth is negative, the
notches occur at frequencies that have
identical phase. Flipping the Depth from
positive to negative or vice versa can have a
dramatic impact on the sound.
• Change the Resonance. If you want to modify the ‘singing’ tones created by high FeedBack settings, try adjusting the Resonance
control. By default, the Resonance slider is
set to ‘Norm’ which is equal to the current
Stages setting. For example, when using TL
EveryPhase with Stages set to 10, setting
the Resonance slider at 2, 4, 6, or 8 stages
will provide a reduced feedback tone. Likewise, to increase feedback tones, set the
Resonance slider to a higher setting.
• Some LFO shapes may create transients or
‘blips’ in the phased sound. This is especially common with the Ramp and Square
Wave LFO shapes. To reduce the transient,
reduce the FeedBack and Stages settings.
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Chapter 61: Voce Plug-Ins
The Voce plug-ins provide a pair of vintage
modulation effect plug-ins that are available in
TDM, RTAS, and AudioSuite formats.
Voce Chorus/Vibrato
Voce Chorus/Vibrato recreates the mechanical
scanner vibrato found in the B-3 Organ. Three
settings of chorus and three settings of vibrato
presented on one cool knob! Fun and easy to use,
it’s a classic effect used for over sixty years.
Inside every B-3 organ, on the end of the driveshaft that spins the tonewheels, you’ll find a mechanical contraption that delays the sound of
the organ. Originally added to make the B-3
sound more like a pipe organ, it imparts frequency variation to the sound.
Although well received by churches, the signature B-3 Chorus/Vibrato graced jazz and rock
recordings ever since. Now you can use this
beautiful effect on any instrument.
Voce Chorus/Vibrato Controls
Simply click the Big Knob to rotate between settings of Vibrato and Chorus. V1 provides the
least amount of vibrato, V2 slightly more, and
V3 the most. Likewise the amount of Chorus increases from C1 to C3.
Option-Click the knob to rotate it in the opposite direction, or click the lettering to select a
specific setting.
Voce Chorus/Vibrato
How Voce Chorus/Vibrato Works
In a large pipe organ, “ranks” of pipes (multiple
pipes designed to emit the same frequency)
aren’t perfectly in tune. The effect goes by
the name “multirank” or, more commonly,
“chorus.”
Voce Chorus/Vibrato Tips and
Tricks
The classic setting for organ is “C3” but you’ll
find other settings useful on a variety of instruments. Some of our favorites include:
Electric Pianos
Many electric pianos feature built-in vibrato.
But if the sound you’re using doesn’t provide a
realistic vibrato (perhaps you’re wrestling with
a sampler), track dry and apply the effect later.
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Guitar
A certain popular guitar amp has a knob that
says “Vibrato” but it’s really just Tremolo.
Tremolo is amplitude modulation; the sound
gets louder and quieter. Vibrato, in contrast,
imparts pitch change. A select few highly sought
after ‘50s Magnatone guitar amps feature a true
tube vibrato (one even does stereo!) You can approximate this sound by recording guitar direct
(or starting with a clean miked sound), applying
Voce Chorus/Vibrato, then using SansAmp ™
PSA-1.
Voce Spin
Voce Spin provides the most accurate simulation of the well-loved rotating speaker. 15 classic recording setups feature horn resonance,
speaker crossover, varying microphone placement—even the “Memphis” sound with the
lower drum’s slow motor unplugged!
How Voce Spin Works
Don Leslie invented the rotating speaker in
1937. His design is simple and elegant: an internal 40-watt tube amplifier feeds a speaker crossover, which splits the signal.
All frequencies below 800 Hz go to a 15” bass
speaker and all frequencies above 800 Hz go to a
compression horn driver.
15” Low Frequency
Loudspeaker
Scoop
Rotation
Direct
Sound
Lower speaker assembly
The large bass speaker is bolted to the cabinet
and a foam drum directly below the speaker reflects the bass outward.
For the high frequencies, a treble horn with two
bells reflects the sound from the compression
horn driver located below.
Only one bell actually produces sound; the other
is merely a counterbalance.
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Then, of course, it spins. Separate belts, pulleys
and motors drive the upper treble horn and the
lower foam drum. Adding to the effect, the upper horn and lower drum spin in opposite directions. Most rotating speakers feature two sets of
motors, allowing both slow (“Chorale”) and fast
(“Tremolo”) rotation speeds.
How Voce Spin Controls
Of course, all that motion creates a rich sound—
but then you have to capture it using microphones. Spin provides fifteen classic recording
setups to choose from, giving you the sounds
you’ve heard on countless records instantly.
Just choose a preset and click Chorale, Tremolo,
or Off. Alternately, click and drag the flip
switch. Short flicks of the wrist land on Off; longer flicks toggle between Chorale and Tremolo.
See also “Voce Spin Additional Controls” on
page 305.
Rover (Slow to Medium) Guitar rotating speaker,
slower variation.
Rover (Medium to Fast) Guitar rotating speaker,
faster variation.
Phaser Medium rotation rate, microphones very
close.
Watery Guitar Fast rotation rate, microphones
close.
Speed Options
Chorale Slow rotation.
Tremolo Fast rotation.
You may also Alt-click (Windows) or Optionclick (Mac) anywhere to toggle between Chorale
and Tremolo speeds.
Spin Presets
122 Model 122 speaker, medium pulleys.
122 (Small Pulley) Small pulleys (fastest rota-
tion).
Off No rotation, but still through the crossover
and speakers (wherever the speakers comes to
rest relative to the microphones!).
Voce Spin Additional Controls
Though the Voce Spin plug-in window contains
only the Chorale/Off/Tremolo control, the following parameters are also available:
122 (Large Pulley) Large pulleys (slowest rota-
• Input Trim
tion).
• Speed Switch
122 (Wide Stereo) Middle pulleys, wide stereo
microphone placement.
122 (Mono) Middle pulleys, one mic each top and
bottom.
21H Model 21H speaker.
• Rotor Balance
• Upper Slow Speed
• Upper Accel Rate
• Upper Decel Rate
• Upper Mic Angle
• Lower Fast Speed
Foam Drum Middle pulleys, microphones close.
• Lower Slow Speed
Memphis Lower drum slow motor unplugged,
• Lower Accel Rate
microphones close.
• Lower Decel Rate
Steppenwolf Lower drum only, loose belts, mi-
• Lower Mic Angle
crophones close.
Rover (Slow to Fast) Guitar rotating speaker,
maximum speed differential.
Using these controls, you can adjust and automate parameters such as input trim (from
–24 dB to +24 dB), set the rotor balance (the
mix between the upper and lower speakers),
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305
specify acceleration and deceleration times (in
seconds) for both the upper and lower speakers,
tweak the fast and slow speeds of each speaker,
and specify the microphone angle for each stereo pair of microphones.
Accessing Voce Spin Controls on a Control
Surface
You can access these additional controls
through Pro Tools plug-in automation, and/or
from a compatible control surface.
When using a control surface, all Voce Spin parameters are available whenever the plug-in is
focused. You only need to enable plug-in automation (as described previously) if you want to
record your adjustments as breakpoint automation.
Accessing Voce Spin Controls
To access additional Voce controls from a control
surface:
Accessing Voce Spin Controls On-Screen
1
All Voce Spin parameters can be adjusted onscreen by editing Pro Tools breakpoint automation data.
To access additional Voce Spin parameters onscreen:
Click the Plug-In Automation button in the
Plug-In window to open the Plug-In Automation
window.
Focus the Voce Spin plug-in on your control
surface. All available parameters are mapped to
encoders, faders, and switches.
Adjust the control currently targeting the desired parameter.
2
If necessary, use the previous/next Page controls to access additional controls.
3
1
In the list of controls at the left, click to select
a control and click Add (or, just double-click the
desired control in the list). Repeat to access and
enable additional controls.
2
Click OK to close the Plug-In Automation window.
To automate your adjustments, be sure to
enable automation for that parameter as described above. See the Pro Tools Reference
Guide for complete track automation instructions.
Voce Spin Tips and Tricks
3
4
In the Edit window, do one of the following:
• Click the Track View selector and select the
automation control you just enabled from
the Voce Spin sub-menu.
– or –
• Reveal an Automation lane for the track,
click the Automation Type selector and select the automation control you just enabled from the Voce Spin sub-menu.
Edit the breakpoint automation for the enabled control.
5
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The “One Mic Way Back In The Corner Of The
Room” Trick
Spin isn’t designed to sound like a rotating
speaker spinning all by itself in a large room.
Spin provides the sound of a miked rotating
speaker, the sound the producer and engineer
hear in the control room. But don’t let that stop
you from getting the sound you want!
To achieve the sound of a distant microphone
capturing the rotating speaker, run Spin using
the wide stereo preset. Now apply a room reverb,
remove any pre-delay, and adjust the wet/dry reverb balance until you get the distant sound
you’re looking for.
Spin into Moogerfooger Lowpass Filter
Generator Leakage
Try using the amplitude modulation effects of
Spin as an LFO driving the Moogerfooger Lowpass Filter!
Of all the sounds to pass through a Leslie, no
sound has been amplified more often than the
sound of B-3 Organ generator leakage. Even
with no notes keyed, a small amount of B-3
sound leaks out.
Distortion and Spin
To simulate overdriving the tube amp powering
the rotating speaker, apply distortion before
Spin, since, in the real-world signal path, the
amp distorts the signal before the speakers
throw the sound around. Among tons of other
great distortion sounds, the SansAmp PSA-1
plug-in provides distortion presets for both the
model 122 and model 147 rotating speakers.
Organ Signal Path
Likewise, when going for classic organ sounds,
route through the Voce Chorus/Vibrato before
Spin, as that’s the signal path in the B-3 organ.
The John Lennon Vocal Thing
In what seems like a particularly dangerous
Beatles studio experiment, a Leslie speaker cabinet was dismembered, a microphone was affixed to the rapidly spinning upper rotor, and
John Lennon attempted to sing into it. Fortunately the deafening wind noise captured by the
microphone put a stop to the proceedings before
anyone got maimed. Feel free just to run the vocal through the rotating speaker—that’s what
they wound up doing.
Reverse Spin
Those reverse-vocal and reverse-guitar tricks
are even more fun when you run ‘em through
Spin. Try reversing the vocal and putting it
through Spin, as well as putting the vocal
through Spin then reversing the processed
vocal.
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Part VIII: Harmonic Plug-Ins
Chapter 62: AIR Distortion
AIR Distortion is an RTAS plug-in that adds
color the audio signal with various types and
varying amounts of distortion.
Output
The Output control lets you lower the Output
level of the distorted signal from 0–100%. At
0%, no distorted signal passes through the output. At 100%, the distorted signal passes
through the output at full volume.
Mix
The Mix control lets you balance the amount of
dry signal with the amount of wet (distorted)
signal. At 50%, there are equal amounts of dry
and wet signal. At 0%, the output is all dry and at
100% it is all wet.
Distortion plug-in window
Distortion Controls
The Distortion plug-in provides a variety of controls for adjusting plug-in parameters.
Drive
The Drive control lets you increase the drive (input volume) of the signal from 0 dB (no distortion) to 60 dB (way too much distortion!) Sometimes an increase or decrease of just 1 of 2
decibels can make a big difference on the
amount and quality of distortion.
The Mix control can be used in conjunction with
the Output control to find just the right balance
of the distorted signal with the input (dry) signal. For example, with Mix set to 50%, equal
amounts of the dry and wet signal pass to the
output. You can then lower the Output control
to decrease the amount of distorted signal being
passed to the output until you get exactly the
right mix between the two signals, and just the
right overall level.
Stereo
When Stereo is enabled, the left and right channels of the incoming stereo signal are processed
separately. When it is disabled, the incoming
stereo signal is summed and processed as mono.
The Stereo button is lit when it is enabled.
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311
Distortion Tone Section
Controls
The Distortion plug-in’s tone controls let you
shape the timbral quality of the distortion.
Pre-Shape The Pre-Shape control lets you in-
crease or decrease a broad gain boost (or attenuation) of treble frequencies in the processed
signal. Pre-Shape is essentially a pre-distortion
tone control that makes the distortion bite at
different frequencies.
Set to 0%, the Pre-Shape control doesn’t affect
the tone at all. Higher settings provide a boost in
the high end of the distorted signal (more treble
distortion), while lower setting suppress the
high end, with some mid-range boost, for a
darker less distorted tone.
High Cut The High Cut control lets you adjust
the frequency for the High Cut filter. To attenuate the high-end of the processed signal, lower
the frequency.
Distortion Clipping Section
The Distortion plug-in’s Clipping controls let
you adjust the DC Bias and the threshold.
DC Bias The DC Bias control lets you change
clipping from being symmetrical to being asymmetrical, which makes it sound richer, and nastier at high settings. The difference is most noticeable at lower Drive settings.
Threshold The Threshold control lets you adjust
the headroom for the dynamic range of the distorted signal between –20.0 dBFS and 0.0 dBFS.
Rather than using the Drive to adjust the signal
level relative to a fixed clipping level, use the
Headroom control to adjust the clipping level
without changing the signal level.
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Distortion Mode Options
Select one of the following options for the
Distortion Mode:
Hard Provides a sharp, immediate distortion of
the signal.
Soft Provides a softer, more gradual distortion
of the signal.
Warp Wraps the waveform back on itself for a
complex distortion tone that changes quickly
from soft to harsh.
Chapter 63: AIR Enhancer
AIR Enhancer is an RTAS plug-in that enhances
the low and high broadband frequencies of an
audio signal.
Enhancer Tune Section Controls
The Tune controls let you set the center frequency for low and high-end enhancement.
Low Adjust the Low control to set the center frequency for the bass boost.
High Adjust the High control to set the center
frequency for the treble boost.
Enhancer Harmonic Generation Section
Controls
Enhancer plug-in window
Enhancer Controls
The Enhancer plug-in provides a variety of controls for adjusting plug-in parameters.
High Gain
The Harmonic Generation controls let you generate additional high-frequency harmonics,
which can brighten up dull signals.
Depth Adjust the Depth control to generate ad-
ditional high frequency harmonics in the signal
(0.0–12.0 dB).
Phase Toggle the Phase control to change the
polarity of the generated harmonics, changing
their phase relationship with the dry signal.
Adjust the High Gain control to boost the high
end.
Low Gain
Adjust the Low Gain control to boost the low
end.
Output
The Output control lets you lower the Output
level from 0.0 dB to –INF dB.
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Chapter 64: AIR Lo Fi
AIR Lo Fi is an RTAS plug-in that you can use to
bit-crush, down-sample, clip, rectify, and mangle an input signal.
Mix
The Mix control adjusts the Mix between the
“wet” (processed) and “dry” (unprocessed) signal. 0% is all dry, and 100% is all wet, while 50%
is an equal mix of both.
AIR Lo Fi Anti-Alias Section
The Anti-Alias section provides control over
anti-aliasing filters that can be used before and
after downsampling to reduce aliasing in the resampled audio signal.
Pre
Lo Fi Plug-In window
AIR Lo Fi Controls
The Lo-Fi plug-in provides a variety of controls
for adjusting plug-in parameters.
Sample Rate
The Sample Rate control resamples the audio
signal at another sample rate.
Bit Depth
The Bit Depth control lets you truncate the bit
depth of the incoming signal from 16 bits all the
way down to 1 bit.
The Pre control adjusts the anti-aliasing filter
cutoff applied to the audio signal before resampling. The filter is applied as a multiplier of the
sample frequency (Fs) between 0.12 Fs and 2.00
Fs.
Post
The Post control adjusts the range of anti-aliasing filter cutoff applied to the audio signal after
resampling. The filter is applied as a multiplier
of the sample frequency (Fs) between 0.12 Fs
and 2.00 Fs.
On
For a much grittier sound, disable the Anti-Alias
filter. The Anti-Alias button is lit when the filter
is enabled.
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315
AIR Lo Fi LFO Section Controls
Wave
The LFO controls let you apply a Low Frequency
Oscillator to modulate the Sample Rate.
Select from the following waveforms for the
LFO.
Rate
Name
Description
When Sync is enabled, the Rate control lets you
select a rhythmic subdivision or multiple of the
beat for the LFO Rate. Select from the following
rhythmic values:
Sine
Provides a sine wave
Tri
Provides a triangle wave
Saw
Provides a saw-tooth wave
• 16 (sixteenth note)
Square
Provides a square wave
• 8T (eighth-note triplet)
Morse
Provides a Morse code-like rhythmic effect
S&H
Provides Sample and Hold (S&H)
modulation
Random
Provides random modulation
• 16D (dotted sixteenth-note)
• 8 (eighth note)
• 4T (quarter-note triplet)
• 8D (dotted eighth-note)
• 4 (quarter note)
• 2T (half-note triplet)
• 4D (dotted quarter-note)
• 2 (half note)
• 1T (whole-note triplet)
• 3/4 (dotted half note)
• 4/4 (whole note)
• 5/4 (five tied quarter notes)
• 6/4 (dotted whole note)
• 8/4 (double whole note)
When Sync is disabled, the Rate control lets you
change the modulation rate independently of
the Pro Tools session tempo.
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Depth
The Depth control lets you adjust the amount of
modulation applied to the Sample Rate.
Sync
Enable Sync to synchronize the LFO Rate to the
Pro Tools session tempo. When Sync is disabled,
you can set the Rate time in Hertz independently
of the Pro Tools session tempo. The Sync button
is lit when it is enabled.
AIR Lo Fi Env Mod Section
Controls
The Env Mod (envelope modulation) section
provides control over an Envelope follower that
can affect the Sample Rate. This is useful for accentuating and enhancing signal peaks (such as
in drum loops) with artificially generated highfrequency aliasing.
Attack
Adjust the Attack control to set the time it takes
to respond to increases in the audio signal level.
Release
Adjust the Release control to set the time it takes
to recover after the signal level falls.
AIR Lo Fi Distortion Section
Controls
The Distortion section provides controls for
adding dirt and grunge to the signal.
Clip
Adds transistor-like distortion to the signal.
Noise
Adds a buzzy, noisy edge to the signal.
Rectify
Acts as a waveshaper, adding aggressive, harsh
distortion to the signal.
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Chapter 65: Lo-Fi
Lo-Fi is part of the D-Fi suite of plug-ins and
provides retro and down-processing effects in
TDM, RTAS, and AudioSuite formats. Features
include:
• Bit-rate reduction
• Sample rate reduction
• Soft clipping distortion and saturation
• Anti-aliasing filter
• Variable amplitude noise generator
Lo-Fi down-processes audio by reducing its
sample rate and bit resolution. It is ideal for emulating the grungy quality of 8-bit samplers.
Lo-Fi can be used as either a real-time TDM or
RTAS plug-in or as a non-real-time AudioSuite
plug-in.
The multichannel TDM version of the Lo-Fi
plug-in is not supported at 192 kHz, use the
multi-mono TDM or RTAS version instead.
Purposely Degrading Audio
Contemporary music styles, especially hip-hop,
make extensive use of retro instruments and
processors such as vintage drum machines, samplers, and analog synthesizers. The low bit-rate
resolutions and analog “grunge” of these devices
are an essential and much-desired part of their
sonic signatures. That is why Avid created D-Fi.
The D-Fi suite of plug-ins combines the best of
these instruments of the past with the flexibility
and reliability of the Pro Tools audio production system. The result is a set of sound design
tools that let you create these retro sounds without the trouble and expense of resampling audio
through 8-bit samplers or processing it through
analog synthesizers.
Lo-Fi Controls
Sample Rate
The Sample Rate slider adjusts an audio file’s
playback sample rate in fixed intervals from
700 Hz to 33 kHz in sessions with sample rates
of 44.1 kHz, 88.2 kHz, or 176.4 kHz; and from
731 Hz to 36 kHz in sessions with sample rates
of 48 kHz, 96 kHz, or 192 kHz. Reducing the
sample rate of an audio file has the effect of degrading its audio quality. The lower the sample
rate, the grungier the audio quality.
Lo-Fi
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The maximum value of the Sample Rate control
is Off (which effectively means bypass).
The range of the Sample Rate control is
slightly different at different session sample
rates because Lo-Fi’s subsampling is calculated by integer ratios of the session sample
rate.
Anti-Alias Filter
The Anti-Alias control works in conjunction
with the Sample Rate control. As you reduce the
sample rate, aliasing artifacts are produced in
the audio. These produce a characteristically
dirty sound. Lo-Fi’s anti-alias filter has a default
setting of 100%, automatically removing all
aliasing artifacts as the sample rate is lowered.
This control is adjustable from 0% to 100%, letting you add precisely the amount of aliasing
you want back into the mix. This slider only has
an effect if you have reduced the sample rate
with the Sample Rate control.
Sample Size
The Sample Size slider controls the bit resolution of the audio. Like sample rate, bit resolution affects audio quality and clarity. The lower
the bit resolution, the grungier the quality. The
range of this control is from 24 bits to 2 bits.
Quantization
Lo-Fi applies quantization to impose the selected bit size on the target audio signal. The
type of quantization performed can also affect
the character of an audio signal. Lo-Fi provides
you with a choice of Linear or Adaptive quantization.
Linear Linear quantization abruptly cuts off
sample data bits in an effort to fit the audio into
the selected bit resolution. This imparts a characteristically raunchy sound to the audio that
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becomes more pronounced as the sample size is
reduced. At extreme low bit-resolution settings,
linear quantization will actually cause abrupt
cut-offs in the signal itself, similar to gating.
Thus, linear resolution can be used creatively to
add random percussive, rhythmic effects to the
audio signal when it falls to lower levels, and a
grungy quality as the audio reaches mid-levels.
Adaptive Adaptive quantization reduces bit
depth by adapting to changes in level by tracking and shifting the amplitude range of the signal. This shifting causes the signal to fit into the
lower bit range. The result is a higher apparent
bit resolution with a raunchiness that differs
from the harsher quantization scheme used in
linear resolution.
Noise Generator
The Noise slider mixes a percentage of pseudowhite noise into the audio signal. Noise is useful
for adding grit into a signal, especially when you
are processing percussive sounds. This noise is
shaped by the envelope of the input signal. The
range of this control is from 0 to 100%. When
noise is set to 100%, the original signal and the
noise are equal in level.
Distortion/Saturation
The Distortion and Saturation sliders provide
signal clipping control.
The Distortion slider determines the amount of
gain applied and lets clipping occur in a smooth,
rounded manner.
The Saturation slider determines the amount of
saturation added to the signal. This simulates
the effect of tube saturation with a roll-off of
high frequencies.
Output Meter
The Output Meter indicates the output level of
the processed signal. Note that this meter indicates the output level of the signal—not the input level. If this meter clips, the signal may have
clipped on input before it reached Lo-Fi. Monitor your send or insert signal levels closely to
prevent this from happening.
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Chapter 66: Recti-Fi
Recti-Fi is part of the D-Fi suite of plug-ins and
provides additive harmonic processing effects
through waveform rectification. Recti-Fi is
available in TDM, RTAS, and AudioSuite formats. Recti-Fi features the following effects:
• Subharmonic synthesizer
• Full wave rectifier
• Pre-filter for adjusting effect frequency
• Post-filter for smoothing generated waveforms
Recti-Fi provides additive synthesis effects
through waveform rectification. Recti-Fi multiplies the harmonic content of an audio track and
adds subharmonic or superharmonic tones,
Purposely Degrading Audio
Contemporary music styles, especially hip-hop,
make extensive use of retro instruments and
processors such as vintage drum machines, samplers, and analog synthesizers. The low bit-rate
resolutions and analog “grunge” of these devices
are an essential and much-desired part of their
sonic signatures. That is why Avid created D-Fi.
The D-Fi suite of plug-ins combines the best of
these instruments of the past with the flexibility
and reliability of the Pro Tools audio production system. The result is a set of sound design
tools that let you create these retro sounds without the trouble and expense of resampling audio
through 8-bit samplers or processing it through
analog synthesizers.
Recti-Fi
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Recti-Fi Controls
Recti-Fi Pre-Filter Control
The Pre-Filter control filters out high frequencies in an audio signal prior to rectification.
This is desirable because the rectification process can cause instability in waveform output—
particularly in the case of high-frequency audio
signals. Filtering out these higher frequencies
prior to rectification can improve waveform stability and the quality of the rectification effect.
If you wish to create classic subharmonic synthesis effects, set the Pre-Filter and Post-Filter
controls to a relatively low frequency, such as
250 Hz.
The range of the Pre-Filter is from 43 Hz to
21 kHz, with a maximum value of Thru (which
effectively means bypass).
Recti-Fi Rectification Controls
Positive Rectification This rectifies the waveform so that its phase is 100% positive. The audible effect is a doubling of the audio signal’s
frequency.
Positive rectification
Negative Rectification This rectifies the wave-
form so that its phase is 100% negative. The audible effect is a doubling of the audio signal’s
frequency.
Negative rectification
Normal waveform
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Alternating Rectification This alternates between rectifying the phase of the first negative
waveform excursion to positive, then the next
positive excursion to negative, and so on,
throughout the waveform. The audible effect is a
halving of the audio signal’s frequency, creating
a subharmonic tone.
Recti-Fi Gain Control
Gain lets you adjust signal level before the audio
reaches the Post-Filter. This is particularly useful for restoring unity gain if you have used the
Pre-Filter to cut off high frequencies prior to
rectification. The range of this control is from
–18dB to +18dB.
Recti-Fi Post-Filter
Alternating rectification
Alt-Max Rectification This alternates between
holding the maximum value of the first positive
excursion through the negative excursion period, switching to rectify the next positive excursion, and holding its peak negative value until the next zero crossing. The audible effect is a
halving of the audio signal’s frequency, and creating a subharmonic tone with a hollow, square
wave-like timbre.
Waveform rectification, particularly alternating
rectification, typically produces a great number
of harmonics. The Post Filter control lets you remove harmonics above the cutoff frequency and
smooth out the sound. This is useful for filtering
audio that contains subharmonics. To create
classic subharmonic synthesis effects, set the
Pre-Filter and Post-Filter to a relatively low frequency.
The range of the Post-Filter control is 43 Hz to
21 kHz, with a maximum value of Thru (which
effectively means bypass).
Recti-Fi Mix Control
Mix adjusts the mix of the rectified waveform
with the original, unprocessed waveform.
Recti-Fi Output Meter
Alt-Max rectification
The Output Meter indicates the output level of
the processed signal. Note that this meter indicates the output level of the signal—not the input level. If this meter clips, the signal may have
clipped on input before it reached Recti-Fi.
Monitor your send or insert signal levels closely
to prevent this from happening.
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Chapter 67: Reel Tape Saturation
Reel Tape Saturation is part of the Reel Tape
suite of tape-simulation effects plug-ins that are
available in TDM, RTAS, and AudioSuite formats.
Reel Tape Saturation simulates the saturation
effect of an analog tape machine, modeling its
frequency response, noise and distortion characteristics, but without any delay or wow and
flutter effects.
Reel Tape Saturation can be placed on mono,
stereo, or multichannel tracks.
Reel Tape Common Controls
All Reel Tape plug-ins share the following
controls:
Drive
Drive controls the amount of saturation effect
by increasing the input signal to the modeled
tape machine while automatically compensating
by reducing the overall output. Drive is adjustable from –12 dB to +12 dB, with a default value
of 0 dB.
Output
Reel Tape Saturation
How Reel Tape Saturation Works
For years, engineers have relied on analog tape
to add a smooth, warm sound to their recordings. When driven hard, tape responds with gentle distortion rather than abrupt clipping as in
the digital domain. Magnetic tape also has a frequency-dependent saturation characteristic that
can lend punch to the low end, and sweetness to
the highs.
Reel Tape Saturation models the sonic characteristics of analog tape, including the effects of
tape speed, bias setting, and calibration level of
the modeled tape machine.
Output controls the output signal level of the
plug-in after processing. Output is adjustable
from –12 dB to +12 dB, with a default value of
0 dB.
Tape Machine
The Tape Machine control lets you select one of
three tape machine types emulated by the plugin, each with its own sonic characteristics:
US Emulates the audio characteristics of a
3M M79 multitrack tape recorder.
Swiss Emulates the audio characteristics of a
Studer A800 multitrack tape recorder.
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Lo-Fi Simulates the effect of a limited-bandwidth analog tape device, such as an outboard
tape-based echo effect.
Tape Formula
The Tape Formula control lets you select either
of two magnetic tape formulations emulated by
the plug-in, each with its own saturation characteristics:
Classic Emulates the characteristics of
Ampex 456, exhibiting a more pronounced saturation effect.
Hi Output Emulates the characteristics of
Quantegy GP9, exhibiting a more subtle saturation effect.
Reel Tape Saturation
Controls
In addition to the Drive, Output, Tape Machine,
and Tape Formula controls, Reel Tape Saturation has the following controls:
Speed
The Speed control adjusts the tape speed in ips
(inches per second). Tape speed affects the frequency response of the modeled tape machine.
Available tape speeds include 7.5 ips, 15 ips, and
30 ips, with a default setting of 15 ips.
Noise
Reel Tape Saturation produces noise only during playback and recording, and not when the
transport is stopped.
The Noise control adjusts the level of simulated
tape noise that is added to the processed signal.
The characteristics of the noise depend on the
Speed, Bias, and Tape Machine settings, and the
relative level of the noise depends on the Drive,
Cal Adjust, and Tape Formula settings.
Noise is adjustable from Off (–INF) to –24 dB,
with the default value being Off.
Bias
The Bias control simulates the effect of underor over-biasing the modeled tape machine. Bias
is adjustable from –6 dB to +6 dB, with a default
value of 0.0 dB. The 0.0 dB value represents a
standard overbias calibration of 3 dB for analog
tape machines, so the control acts as a bias offset
rather than as an absolute bias control.
Cal Adjust
Cal Adjust simulates the effect of three common
calibration levels on the modeled tape machine
and magnetic tape formulations.
With the evolution of tape formulations, it was
possible to increase the fluxivity level, or magnetic strength, of the signals on tape. Over the
years, this resulted in an elevation of recorded
levels relative to a standard reference fluxivity
(185 nW/m at 700 Hz). The Cal Adjust value expresses the elevated level in dB over this standard reference level.
The Cal Adjust control does not affect the overall gain, but does affect the amount of saturation
effect for a given input signal.
Available Cal Adjust values are:
• +3 dB (equivalent to 250 nW/m)
• +6 dB (equivalent to 370 nW/m)
• +9 dB (equivalent to 520 nW/m)
The default value is +6 dB.
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Reel Tape Saturation Tips
 Use Reel Tape Saturation on individual tracks
to round out sharp transients or add color to sustained tones.
 Use Reel Tape Saturation on a group of tracks
(for example drums) to add cohesiveness to the
sound of the group.
 Use Reel Tape Saturation on a Master Fader to
apply analog tape-style compression to a mix.
Reel Tape Saturation Presets
The sonic effect of Reel Tape Saturation depends
on many factors, including the signal level of the
source material; these presets are just starting
points. With some experimentation, Reel Tape
Saturation can yield warmer-sounding results
than conventional digital compression.
Bass Drum Rounds out and adds consistency to
bass drum hits.
Bass Gtr Adds consistency and warmth to bass
guitar sound while avoiding compression artifacts.
Snare Drum Reduces harsh peaks resulting from
EQ-boosted snare drum or rim shots.
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Chapter 68: SansAmp PSA-1
SansAmp PSA-1 is a guitar amp simulator plug-in that is available in TDM, RTAS, and AudioSuite formats. Punch up existing tracks or record great guitar sounds with the SansAmp PSA-1. Capture bass
or electric guitar free of muddy sound degradation and dial in the widest range of amplifier, harmonic
generation, cabinet simulation and equalization tone shaping options available!
SansAmp PSA-1
How the PSA-1 Works
B. Andrew Barta of Tech 21, Inc. introduced the
SansAmp Classic in 1989. A guitar player with
both a trained ear and electronics expertise, Andrew and Tech 21 pioneered the market for tube
amplifier emulation.
SansAmp’s FET-hybrid circuitry captures the
low-order harmonics and sweet overdrive
unique to tube amplifiers. And pushed harder,
SansAmp also generates cool lo-fi and grainy
sound textures that still retain warmth.
SansAmp also features a proprietary speaker
simulator which emulates the smooth, even response of a multiple-miked speaker cabinet—
free of the harsh peaks, valleys and notches associated with single miking or poor microphone
placement.
Finally, SansAmp provides two extremely sweet
sounding tone controls (high and low) that
sound great on most anything.
Tube sound, speaker simulation, warm equalization and cool lo-fi textures—no wonder thousands of records feature the classic sounds of
SansAmp!
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PSA-1 Controls
Use the eight knobs to dial in your desired tone
or effect.
Pre-Amp
Determines the input sensitivity and pre-amp
distortion. Increasing the setting produces an
effect similar to putting a clean booster pedal
ahead of a tube amp, overdriving the first stage.
For cleaner sounds, use settings below the
unity-gain point.
Buzz
Controls low frequency break up and overdrive.
Boost the effect by turning clockwise from the
center point indicated by the arrows. As you increase towards maximum, the sound becomes
(you guessed it) buzzy, with added harmonic
content. For increased clarity and definition
when using distortion, position the knob at its
midpoint or towards minimum.
Punch
Sets midrange break up and overdrive. Decreasing from the center produces a softer, “Fender”style break up. Increasing the setting produces a
harder, heavier distortion. At maximum, it produces a sound similar to a wah pedal at midboost position placed in front of a Marshall amp.
Crunch
Brings out upper harmonic content and, on guitars, pick attack. For cleaner sounds or
smoother high end, decrease as needed.
Drive
Increases the amount of power amp distortion.
Power amp distortion is associated with the
“Vintage Marshall” sound—using SansAmp,
you can produce the effect even at low levels.
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Low
Provides a tone control specially tuned for maximum musicality when used to EQ low frequencies on instruments. Boost or cut by ±12 dB by
turning from the center point indicated by the
arrows.
High
Boosts or cuts high frequencies ±12 dB.
Level
Boosts or cuts the overall gain to re-establish
unity after adding distortion or equalizing the
signal.
PSA-1 Tips and Tricks
Peace and Unity
A little known fact: The arrows in the SansAmp
controls indicate the unity-gain position.
Louder and Cleaner
For best results, don’t set the Pre-Amp level
lower than unity gain when the Drive knob is at
9 o’clock or higher. However, if you want a crystal-clear sound and the Drive control is already
near minimum, decrease Pre-Amp to further remove distortion.
Pre-Amp Versus Drive
To create varying types of overdrive, vary PreAmp in relation to Drive. A high Pre-Amp setting emphasizes pre-amp distortion (see
“Mark 1” preset), while high Drive settings emphasize power amp distortion (see “Plexi” preset).
Part IX: Noise Reduction
Plug-Ins
Chapter 69: DINR
Digidesign Intelligent Noise Reduction (DINR)
plug-in provides BNR, broadband and narrowband noise reduction. BNR can be used for suppressing such unwanted elements as tape hiss,
air conditioner rumble, and microphone preamp noise.
The Broadband Noise Reduction module (BNR)
removes many types of broadband and narrowband noise from audio material. It is best suited
to reducing noise whose overall character
doesn’t change very much: tape hiss, air conditioner rumble, and microphone preamp noise.
In cases where recorded material contains several types of noise, the audio can be processed
repeatedly according to the specific types of
noise.
DINR is available for Pro Tools|HD systems as a
real-time TDM version and as an AudioSuite
version of the BNR.
DINR LE is available for Pro Tools host-based
systems as an AudioSuite-only version of the
BNR.
The TDM version of BNR is not supported at
sample rates above 96 kHz. The AudioSuite
version of BNR supports 192 kHz.
BNR TDM
How Broadband Noise
Reduction Works
The Broadband Noise Reduction module uses a
proprietary technique called Dynamic Audio
Signal Modeling ™ to intelligently subtract the
noise from the digital audio file. Noise is removed with multiple downward expanders that
linearly decrease the gain of a signal as its level
falls.
Creating a Noise Signature
The first step in performing broadband noise reduction is to create what is called a noise signature by selecting and analyzing an example of
the noise within the source material. Using this
noise signature, a noise contour line is created
which is used to define the thresholds for the
downward expanders that will perform the
broadband noise reduction. The noise contour
represents an editable division between the
noise and non-noise audio signals.
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At the same time, DINR also creates a model of
what the non-noise audio signal looks like.
DINR then attempts to pull apart these two
models, separating the bad from the good—the
noise from the desired audio. The noise portion
can then be reduced or eliminated.
The noise reduction itself is achieved through
the use of multiple downward expanders. The
threshold of these expanders is set so that the
noise signal will fall below them and be decreased while the desired audio signal will remain above them, untouched.
The Contour Line
Once the signal level has fallen below the specified Contour Line (which represents BNR’s
threshold), the downward expanders are activated and decrease the gain of the signal as its
level falls. Over five hundred individual downward expanders are used linearly across the audio spectrum to reduce the effects of unwanted
noise.
Psychoacoustic Effects of Noise Reduction
One of the psychoacoustic effects associated
with broadband noise reduction is that listeners
often perceive the loss of noise as a loss of high
frequencies. This occurs because the noise in the
higher frequency ranges fools the ear into thinking the original signal has a great deal of energy
in that range. Consequently, when the noise is
removed it feels as if there has been a loss of
high-frequency signal. DINR’s High-Shelf EQ is
useful for compensating for this effect. See
“High-Shelf EQ” on page 338.
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Limitations of Noise Reduction
It is important to understand that there is a certain amount of trade-off inherent in any type of
noise reduction system. Implementing noise reduction means that you have to choose the best
balance between the following three things:

The amount of noise removed from the signal

The amount of signal removed from the signal

The number of artifacts added to the signal
DINR gives you a considerable amount of control over the above three elements, and lets you
maximize noise reduction while minimizing signal loss and artifact generation. However, as
powerful as it is, DINR does have limitations. In
particular, there are two instances in which
DINR may not yield significant results:
 Cases in which the noise components of the
audio are so prominent that they obscure the actual signal components of the audio.
 Cases in which the noise amplitude of a 24-bit
file is less than –96 dB. DINR is not designed to
recognize noise that is lower than this level.
BNR Spectral Graph
The BNR Spectral Graph displays the noise signature and the editable noise Contour Line. The
Spectral Graph’s horizontal axis shows frequency, which is displayed in Hertz, from 0 Hz
to one-half the current audio file’s sample rate.
The Spectral Graph’s vertical axis shows amplitude, which is displayed in dB, from 0 dB to
–144 dB (below full-scale output of the audio).
The Noise Signature
The jagged line is a graph of noise. This is called
a noise signature. It is created when you use the
Learn button in the Broadband Noise Reduction
window. Once you have the noise signature of an
audio file, you will be able to begin removing the
noise by generating and editing a threshold or
Contour Line (covered next) between the noise
and the desired audio signal.
Spectral Graph showing the noise signature
The Contour Line
The line with a series of square breakpoints is
called the noise contour line. The Contour Line is
an editable envelope which represents the division between the noise and the non-noise signal
in the current audio file. The Contour Line is
created by clicking the Fit or AutoFit button in
the Broadband Noise Reduction window after
you have learned a section of noise. By moving
this envelope up or down, or by moving the individual breakpoints, you can modify which signals are removed and which remain.
Spectral Graph showing the Contour Line
The noise modeling process treats audio below
the line as mostly noise, and audio above the
line as mostly signal. Therefore, the higher you
move the Contour Line upwards, the more audio
is removed. To maximize noise reduction and
minimize signal loss, the Contour Line should
be above any noise components, but below any
signal components.
To fine-tune the broadband noise reduction,
move breakpoints at different locations along
this line to find out which segments remove the
noise most efficiently. Editing the Contour Line
to follow the noise signature as closely as possible will also help maximize noise reduction and
minimize signal loss. See “Editing the Contour
Line” on page 343.
Broadband Noise Reduction
Controls
BNR provides audio processing controls, controls Contour Line controls, and controls for
navigating the Spectral Graph.
BNR Audio Processing Controls
Noise Reduction Amount
This slider controls how much the noise signal is
reduced. It is calibrated in decibels. A setting of
0 dB specifies no noise reduction. Increasing
negative amounts specify more noise reduction.
The default value is 0 dB.
NR Amount, Response, Release, and Smoothing
In many cases, as much as 20–30 dB of noise reduction can be used to good effect. However, because higher amounts of noise reduction can
generate unwanted audio artifacts, you may
want to avoid setting the NR Amount slider to
its maximum value.
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Response
Smoothing
This slider adjusts how quickly the downward
expanders and noise reduction process responds
to the overall changes in the noise in milliseconds. Depending on the character of the noise,
different settings of this control will produce
varying amounts of artifacts in the signal, as the
modeling process attempts to track the noise
signal faster or slower.
This slider controls the rate at which noise reduction occurs once the threshold is crossed. It
lets you reduce the audibility of any artifacts
generated in the modeling process, at the expense of noise reduction accuracy. This is done
by limiting the rate of change of the Response
and Release controls to the specified Smoothing
setting. As soon as the frequency threshold is
reached, the full NR amount value is immediately applied according to Response and Release
settings. When the frequency threshold is
reached, DINR will ramp to the NR Amount
level. Settings range from 0 to 100%. A setting of
0% specifies no smoothing. A setting of 100%
specifies maximum smoothing.
The Response speed ranges from 0 ms to 116 ms.
A setting of 116 ms (slow) specifies that the
modeling process should not attempt to track
very fast changes in the noise character. A setting of 0 ms (fast) specifies that the modeling
process should attempt to follow every change
in the noise character very closely.
A faster setting can yield more noise removal,
but it may generate more artifacts. This is similar to how a noise gate produces chatter when attempting to track highly dynamic material. A
slower setting will allow slightly less noise removal, but will generate much fewer artifacts.
Release
This slider is used in conjunction with the Response slider. It controls how quickly DINR reduces the amount of noise reduction when the
amount of noise present in the audio diminishes. Release times range from 0 ms to 116 ms.
Like the Response control, a faster setting can
yield more noise removal, but it may also generate artifacts. You may want to avoid setting this
control to its slowest position, since it will cause
the noise tracking to slow to the point that the
other controls seem to have no effect.
High-Shelf EQ
The High-Shelf EQ (Hi Shelf) is a noiseless filter
that can be applied after noise reduction has
been performed in order to compensate for a
perceived loss of high-frequency content. It is
unique because it operates only on the signal,
not on any remaining noise. The Freq slider controls the center frequency of the filter. Values
range from 20 Hz to 22 kHz.
The Gain slider controls the gain of the filter.
Values range from –12 dB to +6 dB. The HighShelf EQ can be enabled and disabled by clicking
the Enable button.
High-Shelf EQ
You can also use the High-Shelf EQ to reduce the
amount of high frequencies in a signal. This is
particularly useful if you are working with older
recordings that are band-limited, since the
high-frequency content in these is probably
made up of noise and not signal.
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BNR Contour Line Controls
Fit
Clicking the Learn button creates a noise signature based on the audio segment currently selected on screen. There are two Learn modes:
Learn First Audio mode and Learn Last Audio
mode.
The Fit button computes a noise Contour Line
with approximately 30 breakpoints to fit the
shape of the current noise signature. The Contour Line can then be edited to more closely fit
the noise signature or to reduce specific frequency bands by dragging, adding or deleting
breakpoints.
Learn button
Fit button
Learn First Audio Mode Learn First Audio mode
Pressing the Up Arrow or Down Arrow keys on
your computer keyboard lets you raise or lower
the entire Contour Line, or a selected portion of
the Contour Line. The Left/Right arrows lets you
move a selection left or right. To select a portion
of the Contour Line with multiple breakpoints,
Control-drag (Windows) or Command-drag
(Mac) to highlight the desired area.
Learn
is the default Learn mode. It is designed for use
with audio that has an identifiable noise-only
section that you can locate and pre-select. To
use this mode, locate and select the noise-only
portion of the audio, click the Learn button,
start playback, and BNR will build a noise signature based on the first 16 milliseconds of audio
playback. First Audio Learn mode can be
thought of as a trigger-learn mode, since noise
capturing is triggered by the first audio that
DINR receives.
Learn Last Audio Mode Learn Last Audio mode
is designed to let you locate and identify a segment of noise on-the-fly as you listen to audio
playback. In this mode, you first Alt-click (Windows) or Option-click (Mac) the Learn button,
then initiate audio playback. When you hear the
portion of audio that contains the noise you
want to identify and remove, click the Learn
button a second time. BNR will build a noise signature based on the last 16 milliseconds of audio
playback. The Spectral Graph displays data in
real-time in Learn Last Audio mode.
After you use the Fit function, BNR will automatically boost the entire Contour Line 6 dB
above the noise signature so that all noise components of the audio file are below the Contour
Line. You may want to adjust the Contour Line
downwards as needed to modify the character of
the noise reduction.
Super Fit
The Super Fit button creates a noise Contour
Line consisting of over five hundred breakpoints in order to follow the shape of the noise
signature more precisely.
Super Fit button
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Auto Fit
The Auto Fit function is designed to generate a
noise curve for audio that lacks a noise-only
portion for DINR to learn. Clicking Auto Fit
computes this generic noise curve based on the
points contained within the currently selected
audio, then fits the Contour Line to it. To use the
Auto Fit function, you must first make a selection in the Spectral Graph by Control-dragging
(Windows) or Command-dragging (Mac).
Auto Fit button
If the selected audio has both noise and desired
sound components, you can generate an approximate noise-only Contour Line by selecting a
frequency range that appears to be mostly noise,
then pressing the auto fit button. You can then
edit the resulting noise Contour Line to optimize the noise reduction.
Move Breakpoints Up/Down/Left/Right
These arrows behave differently depending on
whether or not there is a selection of points
along the Contour Line.
BNR Spectral Graph Navigation
Controls
Scroll Left/Right
These buttons scroll the Spectral Graph to the
left or right, respectively.
Scroll Left/Right buttons
To scroll the Spectral Graph (Mac only), use
Control-Option-Left Arrow or Control-Option-Right Arrow.
Zoom Out/In
Clicking on these buttons zooms in or out of the
Spectral Graph. This lets you view and edit the
noise contour with greater precision. If you have
selected a breakpoint or breakpoints, press
Alt+Start+Plus (Windows) or Control+Option+Plus (Mac) to zoom the beginning of the
selection to the center of the screen. Press
Alt+Start+Minus (Windows) or Control+Option+Minus (Mac) to zoom back out.
Zoom Out/In buttons
Move Breakpoints Up/Down/Left/Right buttons
No Selection: When there is no selection, the Up
and Down arrows move the entire Contour Line
up or down by 1 dB, respectively, and the Left
and Right arrows scroll the display left and
right.
With a Selection: Clicking these buttons moves a
selected breakpoint or breakpoints up, down,
right, or left. If there is currently a selection in
the Spectral Graph, clicking the left and right arrow buttons will move the selected breakpoints
left or right. The Up and Down arrows will move
the selected breakpoints up or down, respec-
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Audio Plug-Ins Guide
tively. Alt-Start key-clicking (Windows) or Control-Option-clicking (Mac) the Arrow keys on
your computer keyboard performs the same
function.
To use Broadband Noise Reduction:
Undo
2
Clicking the Undo button undoes the last edit to
the Spectral Graph Display. The Undo button
does not undo changes made to slider positions.
From the Insert pop-up on the track with the
noise, select BNR. The Broadband Noise Reduction window appears.
1
In the Edit window, select the noisiest portion
of the track—ideally, a segment with as little of
the desired signal as possible. This will make it
easier for BNR to accurately model the noise. If
the track contains a segment comprised of noise
only, select that portion.
3
Undo button
• Start audio playback, and in the Broadband
Noise Reduction window, click Learn. BNR
samples the first 16 milliseconds of the selected audio and creates its noise signature.
Using Broadband Noise
Reduction
– or –
Before you start using BNR, take a moment to
think about the nature of the noise in your session and where it’s located: Is it on a single
track, or several tracks? Is it a single type of
noise, or several different types? The answers to
these questions will affect how you use BNR.
If there is a single type of broadband noise on a
single track, insert the BNR plug-in onto the
track. Solo the track to make it easier hear as you
remove the noise. If a single track contains different types of noise, you may need to use more
than one DINR insert to remove the other types
of noise. If multiple tracks contain the same
noise, you may want to bus them all to an Auxiliary Input so you can use a single DINR plug-in
insert. This will minimize the amount of DSP
you use.
Do one of the following:
• Locate and identify noise on the fly, during
playback, using BNR’s Learn Last Audio
mode. To do this, Alt-click (Windows) or
Option-click (Mac) Learn. Begin playback,
and when you hear the segment that you
want DINR to sample as noise, click Learn a
second time. BNR will build a noise signature based on the 16 milliseconds of audio
immediately preceding the second click.
Click Fit. BNR will fit a Contour Line to the
noise signature just created. If you want to create
a Contour Line that follows the noise signature
even more precisely, click Super Fit. A Contour
Line with five hundred breakpoints is created.
4
To audition the effects of the noise reduction
interactively, in the Edit window, select a portion of audio containing the noise. Then select
Options > Loop Playback and press the Spacebar
to begin looped audio playback.
5
Adjust the NR amount slider to reduce the
noise by the desired amount. To compare the
audio with and without noise reduction, click
Bypass.
6
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341
7 To fine-tune the effects of the noise reduction,
adjust the Response, Release, and Smoothing
sliders to achieve optimal results.
To further increase noise reduction, edit the
Contour Line. The quickest way to do this is to
move the entire Contour Line upwards. In the
Spectral Graph, Control-drag (Windows) or
Command-drag (Mac) to select the entire waveform range. Then click the Move Breakpoint Up
button. The higher you move the Contour Line
above the noise signature, the more noise is removed. See “Editing the Contour Line” on
page 343.
8
If you feel that some of high end frequencies of
the audio have been lost due to the noise reduction process, try using the High-Shelf EQ to
compensate. To do this, click BNR’s Hi Shelf button and adjust the frequency and gain sliders until you are satisfied with the results.
9
If you are happy with the results of the noise reduction, use the Plug-In Settings menu to save
the settings so that you can use them again in
similar sessions.
To enable Learn Last Audio mode, Alt-click
(Windows) or Option-click (Mac) the Learn
button. This button flashes red when armed
for Learn Last Audio mode. When you hear
the target noise, click Learn a second time.
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Audio Plug-Ins Guide
Performing Noise Reduction on
Audio that Lacks a Noise-Only
Portion
Ideally, audio that you want to perform noise reduction on will have a noise-only portion at the
beginning or end of the recording that DINR can
analyze and learn. Unfortunately this is not always the case, and in many recordings some
amount of signal is always mixed with the noise.
Obviously, analyzing such audio will produce a
noise signature that is based partially on signal.
Luckily, DINR has provisions for cases such as
this, and this is where the Auto Fit feature comes
in.
If your audio file lacks a noise-only portion for
DINR to analyze, you can still obtain reasonable
results by selecting and learning a segment of
audio that has a relatively low amount of signal
and a high amount of noise (as in a quiet passage). By then selecting a frequency range of the
noise signature and using the Auto Fit function
to generate a generic noise curve, you can recompute the Contour Line based on this selection.
Some editing of the newly generated Contour
Line will probably be necessary to yield optimum results, since it is not based entirely on
noise from your audio file. See “Editing the Contour Line” on page 343.
To generate a Contour Line for audio that lacks a
noise-only portion:
In the Edit window, select a segment of audio
with a relatively low amount of signal and a high
amount of noise.
1
2 Click the Inserts pop-up on the track with the
noise and select BNR. The Broadband Noise Reduction window appears.
3 Click Learn to create a preliminary noise signature.
4
Click Fit to fit a Contour Line to it.
In BNR’s Spectral Graph, Control-drag (Windows) or Command-drag (Mac) to make a selection. Select points where the high-frequency
noise components are most evident. In general,
the flatter areas of the Spectral Graph, are better,
since they represent quieter areas where there is
probably less signal and more noise.
5
6 Click Auto Fit. DINR computes a generic noise
curve and corresponding Contour Line based on
your selection. If you want to remove the selection in the Spectral Graph Display, Control-click
(Windows) or Command-click (Mac) once.
7 Follow the steps given in the previous section
removing the noise using the NR Amount slider
and other controls.
Since the Contour Line is not based entirely on
noise from your audio file, you may also want to
edit its envelope in order to fine-tune the noise
reduction. See “Editing the Contour Line” on
page 343.
8
Editing the Contour Line
One of the most effective ways to fine-tune the
effects of broadband noise reduction is to edit
the Contour Line. The Contour Line treats audio
below the line as mostly noise, and audio above
the line as mostly signal. Therefore, the higher
your move the Contour Line upwards, the more
audio is removed.
To maximize noise reduction and minimize signal loss, the Contour Line should be above any
noise components, but below any signal components. To fine-tune the broadband noise reduction, try moving individual breakpoints at different locations along this line to find out which
segments remove the noise most efficiently. For
more dramatic results, try moving the entire
Contour Line upwards. One drawback of the latter technique is that it will typically remove a
considerable amount of signal along with the
noise.
Remember that high-frequency noise components are typically more evident in the flatter,
lower amplitude areas of the Spectral Graph. Try
editing the Contour Line in these areas first.
To hear the changes you make to the Contour Line
in real time:
Select the target audio in Pro Tools’ Edit window. Make sure the selection is at least a second
or two in length. If the selection is too short, you
won’t be able to loop playback.
1
2
Select Options > Loop Playback.
3
Begin playback.
Noise components on the Spectral Graph
Chapter 69: DINR
343
To edit the Contour Line:
4
To move a breakpoint, click directly on it and
drag it to the desired position. Moving a breakpoint higher increases noise reduction at that
range. Moving a breakpoint lower decreases
noise reduction at that range.
5 To delete a breakpoint, Alt-click (Windows) or
Option-click (Mac) the breakpoint. As long as
you click and hold the mouse, you will delete all
breakpoints that the cursor passes over.
1
To create a new breakpoint, click on the Contour Line.
Using BNR AudioSuite
BNR AudioSuite is identical to the real-time version of BNR, with the addition of two features to
enhance the noise reduction process. These features are:
Dragging a breakpoint
To move multiple breakpoints, Control-drag
(Windows) or Command-drag (Mac) to select
the desired breakpoints. Click the appropriate
Move Breakpoint button (below the Spectral
Graph) to move the selected breakpoints in 1 dB
increments. Control-Shift-drag (Windows) or
Command-Shift-drag (Mac) to extend your selection.
2
Audition Lets you listen specifically to the noise
portion being removed from the target material.
This makes it easier to fine-tune noise reduction
settings to maximize noise reduction and minimize signal loss.
Post-Processing Applies post-processing to the
audio file to help remove undesirable artifacts
that are a result of noise reduction.
To enable either of these features, click the corresponding button. To disable them, click again.
Moving selected breakpoints
To move the entire Contour Line, Control-drag
(Windows) or Command-drag (Mac) to select
the entire range. Click the appropriate Move
Breakpoint button (below the Spectral Graph) to
move the selected breakpoints in 1 dB increments. The higher you move the Contour Line
above the noise signature, the more noise is removed.
3
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Audio Plug-Ins Guide
BNR AudioSuite
Preparing to Render a Clip
Processing a Clip
To prepare a clip to process with the BNR
AudioSuite plug-in:
To begin processing:
Select the desired clips in the target tracks or
the Clip List. Only tracks and clips that are selected will be processed.
Adjust the AudioSuite File controls. These settings will determine how the file is processed
and what effect the processing will have on the
original clips. Here are some guidelines:
From the Pro Tools AudioSuite menu, choose
BNR.
 Decide where the selected clip should be processed:
3 Click Learn to capture the noise signature of
the selected material. If you have selected more
than one track or clip, BNR will build the noise
signature based on the first selected track or clip
when used in Mono mode, or the first two selected track or clip when used in Stereo mode.
• To process the selected clip only in the
track in which it appears, choose Playlist
from the Selection Reference pop-up.
1
2
Click Fit or Super Fit to create a Contour Line
that matches the noise signature.
4
1
– or –
• To process the selected clip in the Clip List
only, choose Clip List from this pop-up.
 Decide if you want to update every occurrence
of the selection clip:
5 Click Preview to begin playback of the selected
material.
• To process and update every occurrence of
the selected clip throughout your session,
enable Use In Playlist (and also choose Clip
List from the Selection Reference pop-up).
Adjust BNR controls and fine-tune the noise
reduction using the techniques explained above
(See “Using Broadband Noise Reduction” on
page 341.)
6
To hear the noise components that are being
removed, click Audition. Adjusting BNR’s controls while toggling this on and off will let you
fine-tune the noise reduction. It also lets you
hear exactly how much signal is being removed
with the noise, and adjust your controls accordingly.
– or –
• If you do not want to update every occurrence of the selected clip, disable Use In
Playlist.
7
 If you have selected multiple clips for processing and want to create a new file that connects
and consolidates all of these clips together,
choose Create Continuous File from the File
mode pop-up menu.
If unwanted artifacts are generated by the
noise reduction process, click Post-processing.
For best results, set the Response and Release
controls to zero.
8
BNR AudioSuite does not allow destructive
processing, so the Overwrite Files option is
not available in the File mode pop-up menu.
From the Destination Track pop-up, choose
the destination for the replacement audio.
2
3
Click Render.
Chapter 69: DINR
345
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Audio Plug-Ins Guide
Part X: Dither Plug-Ins
Chapter 70: Dither
Dither is a dither-generation plug-in that is
available in TDM and RTAS formats.
The Dither plug-in minimizes quantization artifacts when reducing the bit depth of an audio
signal to 16-, 18-, or 20-bit resolution.
For more advanced dithering, use the POW-r
Dither plug-in. See “POW-r Dither” on
page 351.
The Dither plug-in has user-selectable bit resolution and a noise shaping on/off option.
If you are mixing down to an analog destination with any 24-bit capable interface,
you do not need to use Dither. This allows
maximum output fidelity from the 24-bit
digital-to-analog convertors of the interface.
Dither Controls
The Dither plug-in has a Bit Resolution button
and a Noise Shaping button.
Bit Resolution Button
Dither plug-in
Whenever you are mixing down or bouncing to
disk and your destination bit depth is lower than
24-bit, insert a dither plug-in on a Master Fader
track that controls the output mix.
Using a dither plug-in on a Master Fader is preferable to an Auxiliary Input because Master
Fader inserts are post-fader. As a post-fader insert, the dither plug-in can process changes in
Master Fader level.
Use this pop-up menu to choose one of three
possible resolutions for the Dither processing.
Set this control to the maximum bit resolution
of your destination.
16-bit Recommended for output to digital de-
vices with a maximum resolution of 16 bits, such
as DAT and CD recorders.
18-bit Recommended for output to digital de-
vices with a maximum resolution of 18 bits.
For more information on using Dither, see
the Pro Tools Reference Guide.
Chapter 70: Dither
349
20-bit Recommended for output to digital de-
vices that support a full 20-bit recording data
path, such the Sony PCM-9000 optical mastering
recorder, or the Alesis ADAT XT 20. Use this
setting for output to analog devices if you are
using a 20-bit audio interface, such as the
882|20 I/O audio interface. The 20-bit setting
can also be used for output to digital effects devices that support 20-bit input and output, since
it provides for a lower noise floor and greater
dynamic range when mixing 20-bit signals directly into Pro Tools.
The Dither plug-in only provides eight
channels of uncorrelated dithering noise. If
Dither is used on more than eight tracks, the
dithering noise begins to repeat and dither
performance is impaired. For example, if
two Quad Dithers are used, both Quad instances of Dither will have all of their dither
noise uncorrelated. However, any additional instances of the Dither plug-in will
begin to repeat the dithering noise.
Noise Shaping Button
The Noise Shaping button engages or disengages
Noise shaping. Noise shaping is on when the
button is highlighted in blue.
Noise shaping can further improve audio performance and reduce perceived noise inherent
in dithered audio. Noise shaping uses filtering
to shift noise away from frequencies in the middle of the audio spectrum (around 4 kHz), where
the human ear is most sensitive.
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Audio Plug-Ins Guide
Chapter 71: POW-r Dither
POW-r Dither is a dither-generation plug-in
that is available in TDM and RTAS formats.
The POW-r Dither plug-in is an advanced type
of dither that provides optimized bit depth reduction. It is designed for final-stage critical
mixdown and mastering tasks where the highest
possible fidelity is desired when reducing bit
depth. For more information on dithering, see
Chapter 70, “Dither.”
POW-r Dither Controls
POW-r Dither provides a variety of controls for
adjusting plug-in parameters.
Bit Resolution
Use this pop-up menu to choose either 16- or 20bit resolutions for POW-r Dither processing. Set
this control to the maximum bit resolution of
your destination.
16-bit Recommended for output to digital de-
vices with a maximum resolution of 16 bits, such
as DAT and CD recorders.
POW-r Dither plug-in
The POW-r Dither plug-in does not run on
third-party applications that use DAE.
The multichannel TDM version of the POWr Dither plug-in is not supported at
192 kHz. Use the multi-mono TDM or RTAS
version instead.
20-bit Recommended for output to devices that
support a full 20-bit recording data path.
Noise Shaping
Noise shaping can further improve audio performance and reduce perceived noise inherent
in dithered audio. Noise shaping uses filtering
to shift noise away from frequencies in the middle of the audio spectrum (around 4 kHz), where
the human ear is most sensitive.
The POW-r Dither plug-in is not appropriate for truncation stages that are likely to be
further processed. It is recommended that
POW-r Dither be used only as the last insert
in the signal chain (especially when using
Type 1 Noise Shaping).
Chapter 71: POW-r Dither
351
The POW-r Dither plug-in provides three types
of noise shaping, each with its own characteristics. Try each noise shaping type and choose the
one that adds the least amount of coloration to
the audio being processed.
Type 1 Has the flattest frequency spectrum in
the audible range of frequencies, modulating
and accumulating the dither noise just below the
Nyquist frequency. Recommended for less stereophonically complex material such as solo instrument recordings.
Type 2 Has a psychoacoustically optimized low
order noise shaping curve. Recommended for
material of greater stereophonic complexity.
Type 3 Has a psychoacoustically optimized high
order noise shaping curve. Recommended for
full-spectrum, wide-stereo field material.
For more information on using Dither, see
the Pro Tools Reference Guide.
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Audio Plug-Ins Guide
Part XI: Sound Field Plug-Ins
Chapter 72: AIR Stereo Width
AIR Stereo Width is an RTAS plug-in that you
can use to create a wider stereo presence for
mono audio signals.
Comb Adds artificial width to the signal by M-S
encoding then adding a delayed version of the M
component to the S component. This creates a
comb filtering effect that shifts some frequencies to the left and others to the right.
Phase In this mode the Low/Mid/High controls
set the centre frequencies of 3 phase shifters
which shift the relative phase of the left and
right channels, giving a much more subtle effect
than Comb mode.
Delay
Stereo Width plug-In window
The Delay control lets you specify the duration
of delay used in Phase mode (0–8 ms)
Stereo Width Controls
Width
The Stereo Width plug-in provides a variety of
controls for adjusting plug-in parameters.
The Width control sets the final width of the
generated stereo field.
Mode
Process Section Controls
The Mode control lets you specify the method by
which the Stereo Width plug-in will create the
artificial stereo field. Choices include the following:
The Process controls boost or cut the Low, Mid
and High-frequency bands of the generated stereo signal.
Adjust Adjusts the existing stereo width of the
signal by M-S encoding, equalizing the S component with the Low/Mid/High controls and
boosting/attenuating it with the Width control,
then M-S decoding back to stereo. The Delay
control delays the right signal relative to the left
for an additional widening effect (known as
“Haas panning”).
Stereo Width Trim Section Controls
The Trim controls adjust the perceived center/source of the generated stereo signal.
Level The Level control sets the volume of the
perceived center of the stereo signal.
Pan The Pan control sets the position left-to-
right of the perceived center of the stereo signal
Chapter 72: AIR Stereo Width
355
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Audio Plug-Ins Guide
Chapter 73: Down Mixer
Avid Down Mixer is an AAX plug-in (Native)
that can be used to automatically mix greaterthan-stereo multichannel tracks (such as 5.1)
down to stereo (Pro Tools HD and Pro Tools
with Complete Production Toolkit only) or stereo tracks down to mono.
When inserting Down Mixer on a compatible
greater-than-stereo multichannel track, the
channel format of the track output changes to
stereo.
When inserting Down Mixer on a stereo track,
the channel format of the track output changes
to mono.
Avid Down Mixer plug-in, 5.1 to Stereo shown
Down Mixer supports 44.1 kHz, 48 kHz,
88.2 kHz, 96 kHz, 176.4 kHz and 192 kHz sample rates.
Avid Down Mixer plug-in, Stereo to Mono shown
Down Mixer supports the following greaterthan-stereo multichannel formats:
The Source section of the Down Mixer plug-in
provides controls that let you mute, invert the
phase, and adjust the level of each input channel
to the Down Mixer.
• LCR
• LCRS
Source
• 5.1
Mute
• 7.1 SDDS
When enabled, the Mute button mutes the channel input to the Down Mixer.
• 7.1
Chapter 73: Down Mixer
357
Phase
When enabled, the Phase button inverts the
phase of the channel input to the Down Mixer.
Level
You can adjust the level of the channel input to
the Down Mixer from –45 dB to +12 dB. For stereo to mono down mixing, both the Left and
Right channels are mixed to summed mono. For
greater-than-stereo multichannel down mixing,
the following rules apply:
• All left-channel sources (L, Lc, Ls, Lss, Lsr)
feed to the left channel (L) of the down
mixer.
• All right-channel sources (R, Rc, Rs, Rss,
Rsr) feed to the right channel (R) of the
down mixer.
• The center channel (C) and low-frequency
channel (LFE) are panned center into the
stereo field of the down mixer.
Meter
The level meters for source channels always
show the input level (pre-fader) for the channel
regardless of the Source Level setting.
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Audio Plug-Ins Guide
Downmix
The Downmix section of the Down Mixer
plug-in provides output meters and a single
fader to adjust the output level of the Down
Mixer from –45 dB to +12 dB.
Chapter 74: SignalTools
The SignalTools metering plug-ins provide two
metering modules:
• SurroundScope
• PhaseScope
The SignalTools plug-ins are available in TDM
and RTAS formats at all sample rates.
SignalTools SurroundScope
(Pro Tools HD and Pro Tools with Complete
Production Toolkit Only)
SurroundScope is a plug-in that provides surround metering for multichannel track types
from 3 channels (LCR) to 8 channels (7.1 surround). Stereo and mono tracks are not supported.
This version of SurroundScope is compatible
with sessions that used the previous versions
of SurroundScope.
SurroundScope plug-in
SignalTools Surround Display
SurroundScope detects the multi-channel format of the track and displays each channel in the
signal in a circle around the Surround Display.
SurroundScope Surround Display (5.0 shown)
Chapter 74: SignalTools
359
The Surround Display generates a composite
image that indicates relative signal strength in
the displayed channels.
 A circle in the center of the display indicates a
surround signal that is panned equally to all
channels.
An irregular shape that is closer to one side of
the display indicates that the channels on that
side have a stronger signal.

 A teardrop shape that points toward a single
channel indicates that the signal is panned to
that channel.
SignalTools Lissajous Meter Display
The PhaseScope Lissajous Meter displays the relationship between the amplitude and phase of a
stereo signal, enabling you to monitor stereo
imaging graphically.
A “Lissajous curve” (also known as a Lissajous figure or Bowditch curve) is a type of
graph that is able to describe complex harmonic motion. To learn more, search the
Web or your local library for information on
its origins and its two principal developers,
Jules Antoine Lissajous and Nathaniel
Bowditch.
SignalTools PhaseScope
(TDM and RTAS)
PhaseScope is a multichannel metering plug-in
that provides signal level and phase information
for stereo tracks only. (Mono and LCR or greater
multichannel tracks are not supported.) This is
useful for troubleshooting phase problems and
for visualizing the stereo width of a track when
mixing.
PhaseScope Lissajous Meter Display
The Lissajous Meter display is divided into four
quadrants, with left and right channels arranged
diagonally. When audio is panned predominantly to a particular speaker channel, a diagonal line appears, indicating the channel.
The Lissajous Meter displays in-phase material
as a vertical line and out-of-phase material as a
horizontal line.
PhaseScope plug-in
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Audio Plug-Ins Guide
SignalTools Display Options
Both SignalTools plug-ins offer two display options: Phase Meter Display and Leq(A) Meter
Display.
SurroundScope With SurroundScope you can
select the two channels to compare by clicking
the channel buttons around the Surround Display. Selected channels are indicated in blue.
To choose a display option:
 Click the corresponding button in the Options
section of the plug-in window.
Selecting SurroundScope channels for phase
metering
SignalTools display options
SignalTools Phase Meter
Display
The Phase Meter indicates the phase coherency
of two channels of a multi-channel signal.
PhaseScope With PhaseScope, the left and right
channels are always compared.
Signal Tools Leq(A) Meter
Display
The Leq(A) Meter display lets you view the true
weighted average of the power level sent to any
channel or combination of channels (except the
LFE channel) in a multichannel track.
SignalTools Phase Meter
The Phase Meter is green when the channels are
positively out of phase (values from 0 to +1) and
red when the channels are negatively out of
phase (values from 0 to –1).
The Leq(A) Meter display shows a floating average for the level over the interval chosen in the
Window menu. For example, with a setting of 2
seconds, the display shows the average value for
the most recent 2 seconds of audio playback.
At the center or zero position, the signal is a perfect stereo image. At the +1 position, the signal
is a perfect mono image. At the –1 position, the
signal is 100% out of phase.
SignalTools Leq(A) meter and controls
Chapter 74: SignalTools
361
Selecting Channels for Leq(A) Metering
SurroundScope With SurroundScope, you can
select any combination of channels for Leq(A)
metering by clicking the channel buttons
around the Surround Display. Selected channels
are indicated in green.
Reset The Reset button lets you manually reset
the start time of the Leq(A) measurement window.
Auto Reset When enabled, causes the start time
of the Leq(A) measurement window to be automatically reset whenever playback starts in
Pro Tools.
Hold on Stop When enabled, causes the Leq(A)
measurement window timer to pause when playback stops, and resume when playback begins
again.
Selecting SurroundScope channels for Leq (A)
metering
PhaseScope With PhaseScope, you can select either or both channel for Leq(A) metering by
clicking the channel buttons in the corners of
the Lissajous display. Selected channels are indicated in green.
In any of the Loop Transport modes, the
measurement start time is automatically reset each time playback goes back to the beginning of the loop.
SignalTools Level Meters
SignalTools lets you choose the type of metering
and the style of peak hold used, and lets you adjust the reference mark for metering.
SignalTools Meter Types
Clicking the meter types button lets you choose
the type of metering you want to use. Each meter
type has a different metering scale and response.
Selecting PhaseScope channels for Leq(A) metering
Signal Tools Leq(A) Metering
Controls
Window The Leq(A) window menu lets you
choose the length of time the signal is measured
before an average value is calculated. Settings
range from 1 second to 2 minutes.
SignalTools level meter types button
Peak (Default meter type) Uses the metering
scale in EQ III and Dynamics III plug-ins.
When the Leq(A) meter is in INF (infinite) mode
it is constantly averaging the signal without a
floating averaging window.
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Audio Plug-Ins Guide
RMS (Root Mean Square) was used in previous
versions of the Avid SurroundScope plug-in and
uses the same “true” RMS metering scale.
The “true” RMS meter scale is not the same
as the AES 17 RMS scale. For a sine wave
with a peak value of –20 dBFS, the “true”
RMS meter will show a value of –23 dBFS.
(The same sine wave will show a value of
–20 dBFS on an AES 17 RMS meter.‚
SignalTools Meter Peak Hold
Options
Clicking the peak hold button lets you choose
the style of peak hold when peaks are shown in
the plug-in meters.
Peak + RMS Uses a multi-color display to show
both types of metering. Peak metering is shown
in conventional green color, while RMS metering is shown in blue.
VU (Volume Unit) Uses AES standards for signal
level indication.
BBC Uses IEC-IIa standards for signal level in-
dication. This style of metering suppresses short
duration peaks that would not affect broadcast
program material. Reference calibration (4 dB)
is –18 dBFS.
Nordic Uses IEC Type I standards for signal level
indication and provides greater resolution for
readings between –10 dBu and +4 dBu. Reference calibration (0 dB) is –18 dBFS.
DIN Uses IEC Type I standards for signal level
indication and provides greater resolution for
readings between –10 dBu and +5 dBu. Reference calibration (–9 dB) is –18 dBFS.
SignalTools level peak hold button
3 Sec Hold Displays peak levels for 3 seconds
Inf Hold Displays peak levels until meters are
cleared
No Hold Does not display peak levels
SignalTools Meter Reference
Mark
Dragging the reference mark to a different location on the meter scale adjusts the level of the
reference mark for the meter display. The mark
is set by default to the reference level for the corresponding meter type.
VENUE Provides Peak metering behavior with a
meter scale calibrated specifically for VENUE
systems. Reference calibration (0 dB) is
–20 dBFS.
Meter values are always displayed on control surfaces in dBFS values, regardless of
the Meter Type setting.
SignalTools level reference mark
SignalTools meters also change color to
show different ranges of level. The relative
range of color automatically adjusts to follow the current Reference Mark setting in all
meter types (except Peak+RMS).
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Audio Plug-Ins Guide
Chapter 75: TL AutoPan
TL AutoPan is an automatic panning plug-in
that is available in TDM and RTAS formats. TL
AutoPan pans a mono input to a multichannel
(stereo, LCR, quad, or 5.0) output based on a
LFO, envelope follower, MIDI Beat Clock, or
manual automation. TL AutoPan is ideal for
rhythmic panning effects based on your
Pro Tools session tempo. It also provides an
easy and elegant way to automate panning to
multichannel surround formats for post-production.
RTAS on Pro Tools host-based systems only
supports mono-to-stereo.
TL AutoPan Controls
TL AutoPan provides output meters, panner
controls, LFO controls, tempo controls, and envelope controls.
TL AutoPan Output Meters
The Output meters display the amplitude of the
outgoing audio. In mono-to-stereo mode, a two
meter bar is shown. In mono-to-LCR, quad, or
5.0 mode, three, four, or five channels are shown
respectively.
Output meters (L, C, R, Ls, Rs)
The Clip indicator lights red when the channel
has clipped. The clip indicator for each channel
can be cleared by clicking it.
TL AutoPan plug-in, TDM version
Chapter 75: TL AutoPan
365
TL AutoPan Panner Controls
The Panner section provides different controls
for different output channel configurations.
TL AutoPan in mono-to-stereo and mono-toLCR formats provide controls common to all
output configurations: Output, Width, and
Manual. TL AutoPan mono-to-quad and monoto-5.0 formats provide additional controls depending on the Path selection: Angle and Place,
or Spread. Additionally, the Panning Source selector, Panning display, and Path selectors are
common to all output channel configurations.
Output
affected by the setting of the Width slider. For
full manual control, set the Width slider to 0%.
When the Width slider is at 100%, the Manual
slider has no effect on the pan position. When
Width is set to 50%, the LFO sweeps the position
through 50% of its range and the Manual slider
lets you move the position of that 50% range.
Angle
The Angle slider adjusts the orientation of the
panning field from –90° to +90°. At 0°, the panning field is oriented strictly left/right. At –90°
or +90°, the panning field is oriented strictly
front/back.
The Output slider lets you cut or boost the output signal level from –24 dB to +12 dB.
Panner section, mono-to-5.0, left to right path selected
Panner section, mono-to-stereo, left to right path
selected
Width
Place
The Width slider controls the width of the panning field. At 100%, the panning field is at its
widest. At 0%, the panning field is centered and
stationary. The Width slider effectively determines the amount of LFO or Envelope control on
the pan position.
The Place slider adjusts the front/back placement of the panning field. At 0%, the panning
field is centered front/back. At +100%, it is
placed all the way front. At –100%, it is placed
all the way back.
Manual
The Place slider is only available with mono-toquad and mono-to-5.0 formats, and a left to
right or right to left path selected.
The Manual slider directly controls the pan position, this lets you manually control the pan position from a control surface or by using automation. The amount of manual control is
366
The Angle slider is only available with mono-toquad and mono-to-5.0 formats, and a left to
right or right to left path selected.
Audio Plug-Ins Guide
Spread
Panning Display
The Spread slider opens or constricts the field of
panning. At 100%, the spread of the panning
field is at its greatest. At 0%, the spread of the
panning field is completely constricted, and the
sound is centered and stationary (left/right and
front/back).
The Panning display graphically represents the
panning field and the location of the sound
source within that field.
Panning display, mono-to-5.0, left to right path
selected
Sound Location Indicator This bright yellow
light indicates the location of the sound source.
Panning Field Indicator This is the grey line on
which the yellow Sound Location indicator travels and indicates the panning field.
Panner section, mono-to-5.0, clockwise path selected
Path
The Spread slider is only available with monoto-quad and mono-to-5.0 formats, and a circular
path (clockwise or counterclockwise) selected.
Panning Source
Click LFO or ENV to select the source for panning. When the Source is set to LFO, panning is
controlled by the LFO and its controls (see
“TL AutoPan LFO Controls” on page 368). When
the Source is set to Envelope (ENV), panning is
controlled by the Envelope Detector and its controls (see “TL AutoPan Envelope Controls” on
page 370). The Envelope Detector can be triggered by the panned audio signal, or by a sidechain input (see “Using the Side-Chain Input”
on page 371).
The Path selectors determine whether the audio
signal pans left to right, right to left, or in a circular motion clockwise, or counterclockwise.
The circular path selectors (clockwise and counterclockwise) are only available with mono-toquad and mono-to-5.0 formats.
Path selectors, left to right path selected
Panning Source buttons
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367
TL AutoPan LFO Controls
The LFO section provides controls for the Low
Frequency Oscillator that can be used to modulate panning. The controls in the LFO section
only affect the panning if LFO is selected as the
panning source in the panning section (see
“Panning Source” on page 367).
Waveform
The Waveform selector determines the wave
shape used by the LFO. The waveform shape in
use is graphically depicted by the movement of
the Sound Location indicator in the Panning
display.
Selecting the LFO Waveform
LFO section
LFO Triggers
When the Panner section is set to Envelope
(ENV), the controls in the LFO section have
no effect on panning.
Rate
By default, the LFO cycles continuously through
the selected waveform. The LFO can be set to cycle through the selected waveform just once, or
it can be triggered by MIDI Beat Clock, the Envelope, or manually.
The Rate slider adjusts the rate of the LFO in
beats per minute. When Link to Tempo is activated, the slider is ignored and the Tempo LCD
always displays the current session tempo (see
“Tempo LCD” on page 370).
LFO Triggers
Single When the Single trigger is selected, the
LFO will cycle thru the waveform once only and
then stop.
Beat Clock When the Beat Clock trigger is selected, the LFO synchronizes to MIDI Beat
Clock. TL AutoPan receives Beat Clock signal
every 64th-note. The Duration menu determines
how often the Beat Clock signal triggers TL Au368
Audio Plug-Ins Guide
toPan, ranging from every 16th-note to every 4
bars. When Beat Clock signal is received, the
Beat Clock trigger light blinks brightly. Using
the Beat Clock function enables TL AutoPan to
produce consistent panning results, ensuring
that the LFO is always in the same state at each
beat.
Envelope When the Envelope trigger is selected,
the LFO is triggered directly by the Envelope Detector, which analyzes the amplitude of the audio signal. If the Side-Chain Input selector in
the Envelope section is activated, then the sidechain audio signal is used instead. When activated, the Envelope light blinks brighter when
an audio signal is detected. The threshold level
can be adjusted using the Threshold control in
the Envelope section.
If the Envelope Detector is completely released
due to previous portions of the audio signal going above threshold, a trigger occurs the next
time the audio goes above the threshold level.
Another trigger will not happen until the Envelope Detector has completely released after the
audio goes below the specified threshold. Increasing the release time reduces the rate at
which triggers can occur and decreasing the release time increases the rate at which triggers
can occur.
TL AutoPan Tempo Controls
Link To Tempo
When the Link To Tempo option is enabled, the
LFO rate is set to the Pro Tools session tempo,
and any tempo changes in the session are followed automatically. In addition, the LFO rate
slider is ignored and the tempo displayed in the
LCD always displays the current session tempo.
Tempo controls
Duration Selector
The Duration selector works in conjunction
with the session tempo, LFO rate, and Beat
Clock trigger. By default, Duration is set to 1
bar. At that setting, the LFO cycles once within
one bar. When Duration is set to 1 beat, the LFO
cycles within the duration of one beat. When
Link to Tempo is enabled, the Duration menu allows the LFO rate to be set as a function of the
tempo of the Pro Tools session. The Duration
menu also controls how often the Beat Clock
trigger is activated.
Manual When the Manual trigger is selected, the
LFO is triggered manually. This can be especially useful if you want to trigger the LFO using
Pro Tools automation.
With control surfaces and automation, the Manual trigger acts like an on/off switch and triggers
the LFO every time it changes state.
Selecting Duration
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369
Tempo LCD
Threshold
The Tempo LCD displays the tempo in BPM. The
value in the Tempo LCD can also be edited directly by clicking it and typing a new value.
The Threshold slider sets the amplitude level required for the Envelope Detector. The LFO Envelope Detector light blinks brighter when audio
is detected above the threshold.
Attack
Tempo LCD
TL AutoPan Envelope Controls
When Envelope (ENV) is selected as the Panning
source, Panning (as shown in the Panning display) is controlled by the audio signal and the
Envelope section controls.
The Attack slider sets the attack rate of the Envelope Detector.
Release
The Release slider sets the release rate of the Envelope Detector.
Using TL AutoPan
Envelope section
When Envelope (ENV) is not selected as the
Panning Source, the controls in this section
have no effect on the sound.
Side-Chain Input
When the Side-Chain Input selector (the key
icon) is enabled, the audio for the Envelope Detector is taken from the side-chain input rather
than the current track. Select the Side-Chain Input using the Pro Tools Key Input selector at the
top of the plug-in window.
Side-Chain Input selector enabled
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TL AutoPan can be used for dynamic panning effects based on a Low Frequency Oscillator
(LFO), an amplitude envelope (ENV), or manual
control. TL AutoPan makes it easy to pan to the
beat of a music track, as well as panning “flyaround” effects. The following section describes
two possible scenarios for using TL AutoPan:
panning to the beat for rhythmic panning effects
and surround panning effects for post production.
Panning to the Beat
TL AutoPan lets you synchronize the LFO to
MIDI Beat Clock for rhythmic panning effects.
To synchronize TL AutoPan to MIDI Beat Clock:
1 Make sure that your session tempo matches the
tempo of the music.
Insert a mono-to-stereo instance of
TL AutoPan on the mono audio track containing
the audio you want to pan. The track’s channel
width changes from mono-to-stereo.
2
In the TL AutoPan Plug-In window, enable
Link To Tempo. This sets the LFO rate to follow
the session tempo.
3
From the LFO Waveform selector, select Half
Sine.
5
Try automating the Manual control instead
of using the LFO to create a more erratic
panning of the “mosquito” sound.
Select the desired duration from the Duration
selector. For example, select 2 Beats.
4
5 Select the desired waveform for the LFO from
the Waveform selector. For example, select 4
Step Triangle.
Enable Beat Clock for the LFO Trigger. This
ensures that the LFO is synchronized to the beat.
6
Try automating Rate to alter the speed of the
panned sound over time.
6
Play back the session to hear the panning effect.
7
Post Production Panning
(Pro Tools HD and Pro Tools with Complete
Production Toolkit Only)
TL AutoPan lets you pan a mono track to a
greater than stereo (LCR, Quad, or 5.0) output
in a surround path. This is especially useful for
post-production applications. The following example describes how to use TL AutoPan to pan a
“mosquito” sound in 5.0 surround.
Adjust the Rate slider as desired.
Play back the session to hear the “mosquito”
flying around your head.
7
Using the Side-Chain Input
The Side-Chain Input option in TL AutoPan lets
you direct audio from another track in your
Pro Tools session to the Envelope Detector. This
is achieved by sending the audio from the desired channel to a bus and setting the side-chain
input on TL AutoPan to the same bus.
For more information on using the SideChain Input, see the Pro Tools Guide.
To pan a mono track to 5.0 with TL AutoPan:
Insert a mono-to-5.0 instance of TL AutoPan
on the mono track containing the audio you
want to pan. The track’s channel width changes
from mono-to-5.0.
1
Select a 5.0 output path from the track’s Output selector.
2
In the TL AutoPan Plug-In window, select a
clockwise or counter-clockwise Path as desired.
3
Adjust the Spread and Width sliders as desired.
4
Try automating Spread and Width to alter
the positioning of the panned sound.
Chapter 75: TL AutoPan
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Part XII: Instrument Plug-Ins
Chapter 76: Boom
Boom plug-in window, main controls and sections
Boom is a virtual drum machine that features a
broad range of electronic percussion sounds,
paired with a simple, drum-machine-style pattern sequencer. Boom is as an RTAS plug-in that
is part of the Avid Virtual Instrument collection
of plug-ins.
Drum patterns can be created from scratch, or
adapted from one of the included preset patterns. Patterns can be triggered and switched in
real time with the mouse or using MIDI data, facilitating the rapid creation of evolving drum
patterns.
Boom comes with 10 drum kits inspired by classic electronic drum machines. Each individual
sound in a kit can have its volume, pan, pitch,
and decay manipulated and automated in real
time.
Sounds can be shaped to fit the needs of your
production, and even given further animation
over time using automation.
Each pattern is one bar long, with sixteen 16thnote steps. Up to 16 patterns, along with kit and
control settings, can be saved with each Preset.
Chapter 76: Boom
375
Boom Controls
The intuitive control layout for Boom lets you
quickly get a feel for various sections within the
interface. Within no time, you'll be well on your
way to creating fresh and exciting new drum
parts.
Boom Matrix Display
The Matrix display provides a visual overview of
the current pattern in Boom’s sequencer, and is
a quick way to keep track of the pattern’s
rhythm and velocity, as well as what step Boom
is playing at any given time.
step through two levels of lowered velocity, reducing that step’s volume. Clicking the LED
again will silence that step, and turn off its light.
Right-clicking an LED will toggle its on-off
state, preserving the current velocity setting.
The Pattern display above the Matrix shows
which of the 16 patterns in the current preset is
being shown in the Matrix display.
The Copy and Clear buttons above the Matrix are
used to copy or erase patterns when in Pattern
Select mode.
For more information on using the Copy
and Clear buttons, see “Creating a Drum
Pattern Using Boom” on page 380 and
“Playing with Patterns in Boom” on
page 381.
Boom Instrument Section
Controls
Each of Boom’s 10 instruments has a set of controls that set its pan position, volume level, tuning (pitch), and decay (length).
Matrix display
Each horizontal row corresponds with one of
Boom’s 10 instrument channels, and each vertical column represents one of the 16 rhythmic
steps that make up a pattern.
When an LED in the grid is dark, no note is sequenced to play the indicated instrument on
that step.
When an LED in the grid is lit red, the corresponding instrument is sequenced to play at that
step. The brighter the LED is lit, the higher that
step’s velocity has been set.
You can click each LED directly to add or remove a note on that step. When a dark LED is
first clicked, that step is set to play at high velocity. Clicking it a 2nd or 3rd time will cycle that
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Audio Plug-Ins Guide
Boom instrument strips
Pan Sets the current instrument’s pan position
Boom Transport Controls
in the stereo field.
Level Sets the current instrument’s volume.
Tuning Sets the current instrument’s pitch.
Decay Sets the current instrument’s length.
S Solos the selected instrument, letting it play
while temporarily silencing the other instruments. More than one instrument can be soloed
at a time.
M Mutes the selected instrument, silencing it
until the M button is pressed again.
Adjuster Calibrates the sound of the current in-
strument in varying ways.
Sample Selector Sets the current instrument’s
sample (10 samples available for each instrument).
Transport controls
Start and Stop These controls start and stop
Boom’s pattern sequencer. When the Pro Tools
transport is stopped, Boom’s sequencer can play
and stop freely.
When Pro Tools is playing, pressing Play on
Boom’s transport causes Boom to play in sync
with Pro Tools.
Boom Kit Selector
Boom Global Controls
Kit Selector menu
The Global Controls affect all instruments at
once.
The Kit Selector menu gives you access to the 10
preset drum kits in Boom.
The available kits are as follows:
Global controls
Shuffle Adds a variable amount of rhythmic
swing to the currently playing pattern.
Volume Controls the plug-in’s overall output
volume.
Dynamics Scales the difference in volume between the pattern sequencer’s three possible Velocity levels.
Name
Description
Urban 1
R&B-style sounds: Fat kicks,
cracking snares.
Urban 2
Variations of the above.
Dance 1
Classic club-style drums: Fat,
electronic, and punchy.
Dance 2
As above, but more organic,
loopy and percussive.
Electro
Electronic, noisy, distorted
sounds.
Eight-O
A classic analog drum machine
kit.
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377
Name
Description
Nine-0
A classic analog/digital drum
machine kit.
Fat-8
A more processed version of
Eight-0. Aggressive and compressed, with a lot of impact.
Fat-9
A more processed version of
Nine-0. Fatter and crunchier.
Retro
Classic array of analog drum
machine sounds.
Boom Edit Mode Switch
Edit Mode switch
The Edit Mode switch lets you select whether to
edit the current pattern, or choose between the
16 available patterns in the current preset.
Pat Edit Lets you create and edit drum patterns.
Pat Sel Lets you switch between the patterns in
the current preset.
Boom Speed Switches
Patterns can be selected using MIDI notes at
any time, regardless of Edit Mode switch position.
Boom Event Bar
Speed switches
The Speed switches change Boom’s rhythmic relationship with the current tempo set in
Pro Tools.
The switch on the left has three modes. In X1
mode, Boom’s sequencer plays at the same
tempo as the master tempo in Pro Tools. In X2
mode, Boom plays twice as fast. In X1/2 mode,
Boom plays half as fast.
The switch on the right enables Triplet mode. In
Triplet mode, Boom plays only the first 12 steps
in the sequence. The last 4 steps turn grey, indicating that they will not be played.
The 12 steps play in the same amount of time
Boom would normally play all 16, for the creation of triplet grooves.
Trying different combinations of Speed switch
settings on-the-fly can create interesting rhythmic variations.
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Audio Plug-Ins Guide
The Event Bar is where most of the work of creating and playing patterns in Boom is done.
Event Bar, showing the 16 Event switches, in various
states
In Pattern Edit mode, the sixteen numbered
Event switches that make up the Event Bar each
correspond with a 16th note step in the current
pattern. The rhythm of the currently selected instrument is shown. By default, in an empty pattern, all of the Event switches will be dark, indicating that the selected instrument will not play
on any of the 16 steps.
When an Event switch is selected, it lights to
show that the selected instrument will play at
that step in the pattern.
Clicking (or triggering via MIDI or a control
surface) a step’s switch a 2nd or 3rd time will cycle it through two levels of lowered velocity, reducing that step’s volume. Clicking the switch
again will silence that step, and turn off its light.
Right-clicking an Event switch will toggle its onoff state, preserving the current velocity setting.
In Pattern Select mode, the Event switches
choose between the 16 patterns in the current
preset.
Boom Info Display
Boom has an Info display that shows the setting
of the currently selected control.
1/16 Boom starts playing the selected pattern
from one of the first five steps in the pattern,
corresponding with the incoming MIDI note’s
place in the current quarter-note.
Notes played on the first or third quarter note of
the current bar will trigger the current pattern
from step 1. Notes played on the 2nd or 4th
quarter note will trigger the pattern from step 5.
Off Boom starts playing the selected pattern in
sync whenever triggered by a MIDI note, without synchronizing to the Pro Tools transport.
Pattern Chaining On/Off
This lets you turn the Pattern Chain function on
or off.
Inserting Boom on a Track
Info display
Boom Setup Page
Click the Setup button to view the Setup page. It
has two parameters you can set that change
Boom’s behavior.
To use an instrument plug-in to its best advantage, insert it on a stereo Instrument track in
your Pro Tools session.
To insert an instrument plug-in on an Instrument
track:
Create a new stereo Instrument track (recommended) in your Pro Tools session:
1
• Choose Track > New.
• Select 1 new Stereo Instrument track in
Ticks.
• Click Create.
Setup button
Sync Mode
This sets the way Boom synchronizes with
Pro Tools when patterns are triggered using
MIDI notes. The modes are as follows:
Click the Pro Tools Track Insert selector and
select an instrument.
2
Beat Boom starts playing the selected pattern
from the step that corresponds with the incoming MIDI note’s place in the current bar.
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379
If needed, you can now record-enable the instrument track to enable the use of a MIDI controller to play the instrument and/or help in
creating MIDI sequences within the sequencer
in Pro Tools.
3
See the Pro Tools Reference Guide for instructions on how to use the MIDI sequencer
in Pro Tools.
Creating a Drum Pattern
Using Boom
This section will get you started on the process
of creating beats with Boom. First, you’ll need
an empty pattern to edit.
When you have a satisfying rhythm created for
your first instrument (such as Bass Drum), select the next instrument you want to add to the
pattern (such as Snare Drum), and repeat the
above process.
5
Continue adding parts until you’re satisfied
with the pattern.
6
Saving a Boom Pattern as a
Preset
You can save Boom settings, (in this case, your
drum patterns and instrument settings) as plugin settings files (.tfx), or presets.
To clear a pattern:
Set the Edit Mode switch to Pattern Select
mode.
1
Click one of the Event switches to select the
pattern you want to clear. For now, click the
Event switch marked 1.
2
Click the Clear button above the Matrix display. All notes in the selected pattern will be
cleared.
To save a Boom pattern:
Click the Plug-In Settings Select button to
open the Plug-In Settings dialog.
3
1
To create a new pattern:
2 Choose “Save Settings As...”, choose a name for
your preset, and click “ Save”.
1
Set the Edit Mode switch to Pattern Edit mode.
2
Press play on Boom’s transport.
In the Instrument Section, find the first instrument whose pattern you want to edit, and
click its Instrument Name area. The selected Instrument Strip’s background color will become
highlighted, indicating that it is selected.
3
In the Event Bar, try out various rhythms by
toggling the Event switches on and off.
4
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Plug-Ins Settings Select Button
Audio Plug-Ins Guide
Playing with Patterns in
Boom
Now that you’ve started creating patterns, let’s
take a look at how to switch between and create
variations of them.
To create a simple fill:
1 Select Snare Drum, and toggle Event switches
13–16 to on. This will give you a simple 4-note
snare roll at the end of the pattern.
Try switching between patterns 1 and 2 to hear
the new pattern and the fill you’ve added.
2
To switch between patterns on-the-fly:
1
Press Play on Boom’s transport.
Set the Edit Mode switch to Pattern Select
mode.
2
Click the Event switch marked 2. A different
pattern will start to play, and its notes will appear in the Matrix display.
Event switches 13–16, toggled on to create a roll
3
Press Event switch 1, and you’ll see and hear
your original pattern return. Make a copy of it,
so that you can create a variation of the pattern.
4
To copy a pattern:
Click the Copy button above the Matrix display. Event switch 1 lights up red, and Event
switches 2–16 blink, indicating that they are
available to receive a copy of the selected pattern.
1
Controlling Boom with MIDI
Boom becomes much more powerful when controlled using MIDI. Boom responds to two main
ranges of MIDI notes:
C1–D#2 Plays each of the instruments in the
current drum kit. Used primarily when using
Pro Tools MIDI or instrument tracks to control
Boom, rather than Boom’s built-in pattern sequencer. These mappings closely match the
General MIDI standard, for ease of use with preexisting MIDI sequences.
Click Event switch 2. The Event Bar returns to
its normal state, and pattern 1 will be copied to
pattern 2.
MIDI Note
Instrument Played
C1
Kick
If you decide before doing so that you don’t
want to copy the pattern after all, just press any
other button or move any other control besides
the Event switches, and the Copy action will be
cancelled.
C#1
Rim
D1
Snare
D#1
Clap
Press Event switch 2, then set the Edit Mode
switch back to Pattern Edit mode. You’ll see that
you now have an identical copy of pattern 1 to
work with. Make a simple edit to the pattern, by
adding a rhythmic fill.
E1
Snare
F1
Lo Tom
F#1
Clsd Hat
G1
Lo Tom
G#1
Clsd Hat
2
3
4
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381
MIDI Note
Instrument Played
A1
Hi Tom
A#1
Open Hat
B1
Hi Tom
C2
Hi Tom
C#2
Crash
D2
Hi Tom
D#2
Ride
Playing Boom Patterns Using
MIDI
Much like you can switch between patterns by
clicking on various Event switches, you can play
and switch between patterns using MIDI data.
This lets you create interactive changes in your
beat over time, and record the MIDI data so that
the same sequence can be played back and edited
once it’s been recorded.
To play patterns using a MIDI controller
C3–D#4 Each note triggers one of the 16 patterns in the current preset, switching between
them on the fly
The first set of notes lets you play and sequence
Boom’s sounds directly like any other software
instrument. The second set lets you switch between and create sequences of Boom’s patterns.
See the Pro Tools Reference Guide for
instructions on how to use the MIDI
and instrument tracks in Pro Tools.
1 Record-enable the Instrument track on which
Boom is inserted.
Use the octave switches on your MIDI controller to make sure you have access to MIDI notes
C3–C4. You may need to play a number of notes
to find the right range, but once you have, you’ll
notice that Boom plays a pattern each time you
play a note.
2
If you press and hold a note, the corresponding
pattern plays until you release the note.
3
4 If you play legato within that range while holding down the first note, Boom switches patterns,
starting the new pattern at the same rhythmic
step where the previous pattern left off.
This MIDI data can be recorded and edited using
the MIDI sequencer in Pro Tools, letting you
create complex sequences of drum patterns.
If you do not have a MIDI controller, you can use
the Pencil tool in Pro Tools to create a sequence
of MIDI notes to trigger patterns over time.
See the Pro Tools Reference Guide for instructions on how to use MIDI and instrument tracks in Pro Tools to control instrument plug-ins.
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Creating Boom Pattern
Chains
Pattern Chains let you to create sequences of
drum patterns in real time, either with the
mouse and keyboard, or with MIDI notes.
To create a pattern chain with the mouse and
keyboard
1
Press Play on Boom’s transport.
2
Switch Boom into Pattern Select mode.
Click the Event switch that corresponds with
the pattern you want to start the chain with.
3
Hold Control on the computer keyboard and
click the Event switches that correspond with
each pattern you want to add to the chain.
If you let go of a note, its pattern will be removed from the chain. If you let go of all but one
note, the chain will stop, and Boom will go back
to playing one pattern repeatedly.
4
This MIDI data can be recorded and edited using
the MIDI sequencer in Pro Tools, let you more
easily create sequences of drum patterns.
If you do not have a MIDI controller, you can use
the Pencil tool in Pro Tools to create a sequence
of MIDI notes that can trigger Pattern Chains.
Pattern chaining can be turned on and off in
the Setup Page. When off, the above keyboard and MIDI behavior does not occur.
4
Boom will play the patterns you’ve chosen in
the order they were added to the chain.
5
Using the MIDI Learn
Function on Avid Virtual
Instruments
To remove a pattern from the chain, hold Control and click its Event switch.
6
To go back to playing patterns individually,
click any Event switch without holding down
Control.
7
To create a pattern chain with MIDI notes
Press and hold a MIDI note between C3 and D3
on your MIDI controller, triggering a pattern.
1
Press and hold a second note in that range
along with the first. A Pattern Chain will be created, and Boom will alternate between playing
the first pattern you chose, and the second.
2
In addition to pre-assigned MIDI controllers
(such as Sustain Pedal and Volume), you can assign MIDI controllers to parameters within an
Avid Virtual Instrument plug-in for automation
or real-time control from a MIDI keyboard or
control surface. MIDI assignments are saved
with the session.
Press and hold another note, and it will be
added to the chain. Boom will play all three patterns in sequence. If you hold down more notes,
their patterns will be added to the chain.
3
Chapter 76: Boom
383
To assign an Avid Virtual Instrument parameter to
a MIDI controller, do one of the following:
Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Assign and
choose a controller number from the pop-up
MIDI CC list.

– or –
 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Learn from
the menu and move the desired knob or controller on your MIDI keyboard or sequencer. The instrument plug-in will set this MIDI controller to
the parameter you have chosen.
Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
To remove a MIDI controller assignment:
 Control-click or Right-click (Mac), or Rightclick (Windows) an assigned control and choose
Forget to remove its MIDI controller assignment .
All Avid Virtual Instrument plug-ins have
pre-defined parameter assignments for Avid
and supported third-party hardware control
surfaces.
Set Min/Max
These options let you scale incoming MIDI controller data so that the chosen control does not
go below or above a certain value.
To set the Min/Max level:
MIDI CC
Function
120
All Sound Off
121
Reset Controllers
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, choose Set Min or Set
Max, and select the desired lower or upper limit
for the current control.
123
All Notes Off
Invert Range
124
Omni Off (Not used in Plug-Ins)
125
Omni On (Not used in Plug-Ins)
126
Mono On (Not used in Plug-Ins)
127
Mono Off (Not used in Plug-Ins)
This option lets you invert incoming MIDI controller data so that the chosen control reacts in
inverse proportion to the assigned MIDI controller.
To invert a control’s response:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, and select Invert
Range.
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Audio Plug-Ins Guide
Chapter 77: Bruno and Reso
Bruno and Reso are a pair of TDM plug-ins that
process audio using a sound generation technique known as cross-synthesis.
• Editable ADSR envelope generator
Cross-synthesis generates complex sound textures by using an audio track as a tone source
then applying a variety of synthesizer-type effects to that tone source. Bruno and Reso each
use a different sound generation method:
• Time-slice switching using envelope triggering or MIDI beat clock
Bruno uses time-slicing, a technique whereby
timbres are extracted from the source audio during playback and crossfaded together. This
crossfading between signals can create a rhythmic pulse in the sound as the timbre changes.
• Supports sample rates up to 192 kHz

Reso uses a resonator, which adds harmonic
overtones to the source audio through a short
signal delay line with a feedback loop.

In both cases, the processed sound can then be
played in real time or sequenced using the MIDI
recording and playback capabilities of
Pro Tools.
• Portamento
• Velocity-sensitive gain and detuning
• Voice-stacking
• Side-chain input for control using an external audio source
• Online help
Reso Features
Reso features include:
• Harmonic resonance generation
• Up to 62 voices of polyphony (on
Pro Tools|HD Accel systems)
• Multi-voice detuning
• Resonant low pass filter
• Editable ADSR envelope generator
• Portamento
Bruno/Reso Features
• Velocity-sensitive resonance, damping,
gain, and detuning
Bruno Features
• Harmonic switching using envelope triggering or MIDI beat clock
Bruno features include:
• Time-slice tone generation with adjustable
crossfade
• Polyphony: Up to 62 voices of polyphony
(on Pro Tools|HD Accel systems)
• Multi-voice detuning
• Voice-stacking
• Side-chain input for control using an external audio source
• Supports sample rates up to 192 kHz
• Online help
Chapter 77: Bruno and Reso
385
Bruno/Reso DSP
Requirements
Bruno and Reso each require one full DSP chip
on a Pro Tools|HD card.
Play Bruno/Reso with the on-screen keyboard
or by MIDI control (see “Playing Bruno/Reso” on
page 386).
3
Adjust Bruno/Reso controls to get the effect
you want.
4
DSP and Voice Polyphony
The maximum number of Bruno/Reso voices
available per DSP chip depends on the sample
rate of the session and the type of DSP cards in
your system.
HD Accel On Pro Tools|HD systems equipped
with an HD Accel card, Bruno and Reso provide
up to 62 voices at their maximum setting. The
62-voice versions of Bruno and Reso require one
entire DSP chip on an HD Accel card. Polyphony
is reduced by half for sessions at 88.2 kHz and
96 kHz.
HD Core and HD Process On Pro Tools|HD sys-
tems not equipped with an HD Accel card, Bruno
and Reso provide a maximum of 24 voices of polyphony. Polyphony is reduced by half for sessions at 88.2 kHz and 96 kHz (up to 14 voices).
Inserting Bruno/Reso onto an
Audio Track
To use Bruno/Reso in a Pro Tools session, you
must add it to a track as an insert. Once
Bruno/Reso is inserted on the track, you can adjust its controls to get the effect that you want,
then play the plug-in using the on-screen keyboard, an external MIDI controller, or an Instrument track.
To add Bruno/Reso as a track Insert:
Click the Insert selector on the desired track
and select Bruno or Reso.
1
Click Play on the Pro Tools Transport to start
audio playback.
2
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Audio Plug-Ins Guide
Playing Bruno/Reso
To generate sound, Bruno/Reso must be played
during audio playback. You can play
Bruno/Reso in two ways:
 In real time, using either the on-screen keyboard or an external MIDI controller.
– or –

Using MIDI.
Using the On-Screen Keyboard
The simplest way to play Bruno/Reso is to use its
on-screen keyboard. You can click one note at a
time or use keyboard latch to hold multiple
notes.
The on-screen keyboard
Notes played with the on-screen keyboard are
triggered at a MIDI velocity of 92.
To play Bruno/Reso with the on-screen keyboard:
1
Open the plug-in window for Bruno/Reso.
Click Play on the Pro Tools Transport to start
audio playback.
2
Click the on-screen keyboard. Bruno/Reso will
only produce sound while audio plays on the
source track.
3
To latch keys on the on-screen keyboard:
To play Bruno/Reso with a MIDI controller:
Click the Latch bar, then click multiple keys.
Chords can be played in this way.
1
Start audio playback.
2
Play your MIDI keyboard while audio plays.
1
2
To turn off a latched key, click it a second time.
To turn off key latching entirely, click the
Latch bar a second time.
3
Bruno/Reso only produces sound during audio
playback on the source track.
Using MIDI Playback
Saving a Bruno or Reso setting while keys
are latched also saves the latched keys.
Using a MIDI Keyboard Controller
You can play Bruno/Reso live using a MIDI keyboard controller. You can also use the MIDI keyboard controller to record your performance on
an Instrument track or a MIDI track routed to
Bruno/Reso for playback.
You can also play Bruno/Reso using a Pro Tools
Instrument or MIDI track. Use a separate Instrument or MIDI track for each Bruno/Reso
plug-in.
To play Bruno/Reso using an Instrument or MIDI
track:
1
Insert Bruno or Reso on an audio track.
Click the Instrument or MIDI track’s MIDI
Output selector and choose Bruno or Reso. If you
are using multiple Bruno or Reso plug-ins, they
will all appear in this pop-up. Route the Instrument or MIDI track to the correct one.
2
To configure Bruno/Reso for MIDI input:
1
Insert Bruno/Reso on an audio track.
2 Choose Track > New and specify 1 new Instrument or MIDI track, then click Create. Create a
separate Instrument or MIDI track for each
Bruno/Reso plug-in you use.
3
Start Pro Tools playback.
3
Click the track’s MIDI Output selector and select Bruno or Reso.
Using an External Key Input
with Bruno/Reso
If you are using multiple Bruno/Reso plug-ins,
they will all appear in this pop-up. Route the Instrument or MIDI track to the correct one.
(Side-Chain Processing)
4
Record-enable the Instrument or MIDI track.
Test your MIDI connection by playing notes on
your MIDI keyboard. The corresponding notes
should highlight on Bruno/Reso’s on-screen keyboard.
5
Bruno and Reso feature side-chain processing
capabilities. Side-chain processing lets you trigger certain controls from a separate reference
track or external audio source. The source used
for triggering is referred to as the key input.
You can use this capability to control the rate at
which Bruno performs sample switching or Reso
toggles its harmonics back and forth using the
dynamics of another signal (the key input).
Chapter 77: Bruno and Reso
387
Typically, a rhythm track such as a drum kit is
used to trigger these controls and create rhythmic timbral changes that match the groove of
the key input.
To use a key input for side-chain processing:
Bruno Controls
Bruno uses time-slicing for tone generation, extracting timbres from the audio track during
playback and cross-fading them together at a
user-selectable rate.
Click the Key Input selector and choose the input or bus with the audio you want to use to trigger the plug-in.
1
Selecting a Key Input
Click the Key Input button (the button with the
key icon above it) to activate external side-chain
processing.
2
3 Begin playback. The plug-in uses the input or
bus that you chose as a side-chain input to trigger the effect.
To hear the audio source you have selected to
control side-chain input, click the Key Listen
button (the button below the Ear icon).
4
Remember to disable Key Listen to resume
normal plug-in monitoring.
Adjust other controls to create the desired effect.
Bruno
This crossfading can create a rhythmic pulse in
the sound as the timbre changes. This makes
Bruno ideal for creating tonal effects with a continuously shifting timbre—similar to the wave
sequencing found on synthesizers such as the
PPG, Prophet VS, Korg Wavestation, and
Waldorf XT.
5
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Audio Plug-Ins Guide
By carefully choosing the type of source audio,
the crossfade length, and the type of switching,
you can create unique and complex sound
textures.
Bruno Timbre Controls
External Key Enables switching from a separate
reference track or external audio source. The
source used for triggering is referred to as the
key input and is selected using the Side-chain Input pop-up. You can assign either an audio input
channel or a TDM bus channel.
Typically, a drum track is used as a key input so
that switching occurs according to a definite
rhythmic pattern.
Timbre controls
Crossfade
Crossfade sets the rate at which Bruno extracts
timbres from the source audio and crossfades
from one time slice to the next. The range of this
control is from 2 to 40 Hz (cycles per second) in
a 44.1 kHz or 48 kHz session, and from 4 to 40
Hz in a 96 kHz session.
The higher the crossfade frequency, the smaller
the time slice, and the faster Bruno moves between slices. A higher frequency crossfade
would retain more characteristics of the original
audio source and would have a pulsed or wavesequenced feel.
The lower the crossfade frequency, the larger
the time slice, and the slower Bruno moves between slices. A lower frequency crossfade would
have fewer characteristics of the original source
and a more rounded or gradually evolving
sound.
Switch
Switch causes Bruno to switch directly between
time-sliced samples without crossfading them.
This adds a distinct rhythmic pulse to the timbral changes.
Switching can be controlled by triggering (using
the dynamics of the source audio or an external
key input) or by MIDI clock.
Key Listen When enabled, Key Listen monitors
the source of the key input. It is often useful to
do this in order to fine tune Bruno’s settings to
the key input. See “Using an External Key Input
with Bruno/Reso” on page 387.
Threshold Sets the level in decibels above which
switching occurs. When the audio input level
rises above the Threshold level, Bruno will
switch directly to a new time-slice. The range of
this control is from a low of –48 dB (maximum
switching) to a high of 0.0 dB (no switching). If
no key input is used, the dynamics of the source
audio will trigger switching. If a key input is
used, the dynamics of the key input signal will
trigger switching. Threshold-based switching
can be used at the same time as Key Input-based
switching.
MIDI Clock Triggers switching in sync with a
MIDI Beat Clock signal. This creates a very regular, highly rhythmic wave sequencing effect
that is ideal for sessions arranged around MIDI
beat clock. This control can be set to quarter,
eighth, or sixteenth notes, or dotted triplet values of the same.
For quick numeric entry of MIDI beat clock
values, type “4,” “8,” or “16” for quarter
notes, eight notes, or sixteenth notes. Add
“t” for triplets, or “d” for dotted note values.
Typing “4t” for example, enters a quarter
note triplet value. Typing “16d” enters a
dotted sixteenth note value.
Chapter 77: Bruno and Reso
389
Timbrometer
Gain Velocity
Gain Velocity sets the velocity sensitivity of the
Gain Amount control. This gives you touch-sensitive control over Bruno’s volume using a MIDI
keyboard.
Timbrometer
This multicolor waveform display shows the amplitude and duration of the audio signal generated by Bruno as well as the frequency of timbral
changes and whether they are crossfaded or
switched.
Red and blue waveform segments indicate timbral changes that are crossfaded. Green waveform segments indicate timbral changes that are
hard switched.
Bruno Amplitude Controls
This control is adjustable from a low of –24 dB
(maximum velocity sensitivity) to a high of
0.0 dB (no velocity sensitivity).
If you set Gain Velocity to –24 dB, a soft strike
on a key will reduce gain up to –24 dB. A hard
strike will have a maximum output level equal to
the current dB setting of the Gain Amount control.
Conversely, if Gain Velocity is set to 0.0 dB,
Bruno’s volume will not change no matter how
hard or soft you strike a key on your MIDI controller.
Gain Velocity only has an effect when you
play Bruno with a velocity-sensitive MIDI
controller.
Mix
Mix adjusts the mix of the processed audio with
the original, unprocessed audio.
Amplitude controls
Gain Amount
Gain Amount attenuates output level gain. Since
some of Bruno’s controls can cause extreme
changes in signal level, this is particularly useful
for preventing clipping and achieving unity gain
with the original signal level. This control is adjustable from a low of –96 dB (no gain) to a high
of 0.0 dB (maximum gain).
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Audio Plug-Ins Guide
Spread
When Bruno is used in stereo, the Spread control can be used to pan multiple voices within
the stereo field. This control is adjustable from
0% (no stereo spread) to 100% (maximum stereo
spread).
Voice stacking has a direct effect on stereo
Spread. For example, setting Voice Stack to 1
and Spread to 100% will randomly pan each note
played. Setting Voice Stack to 4 and Spread to
100%, will pan two of the four voices hard left,
and two voices hard right.
ADSR Envelope Generator
Bruno Pitch Controls
The ADSR (attack, decay, sustain, release) Envelope Generator controls Bruno’s amplitude envelope. This amplitude envelope is applied to a
sound each time a note is struck.
The four envelope elements can be adjusted by
dragging the appropriate breakpoint, or by typing in a numeric value.
Attack Controls the amount of time in millisec-
onds that the sound takes to rise from zero amplitude to its full level. The longer the attack, the
more time it takes for the sound to reach maximum volume after the a note is struck. This control is adjustable from 0.0 to 5000 milliseconds.
Decay Controls the amount of time in milliseconds that the sound takes to fall from its peak
Attack level to the Sustain level. This control is
adjustable from 0.0 ms to 5000 ms.
Sustain Level Controls the amplitude level in dB
that is reached after the decay time has elapsed.
The amplitude level stays constant as long as a
MIDI note remains depressed. This control is
adjustable from –96 dB (no sustain) to 0.0 dB
(maximum sustain).
Release Controls the amount of time in milliseconds that the sound takes to fall from the
Sustain level to zero amplitude after a note is released. This control is adjustable from 0.0 ms to
5000 ms.
Pitch controls
Glide
Glide, also known as portamento, determines
the amount of time it takes for a pitch to glide
from the current note to the next note played.
This effect is commonly found on synthesizers.
Glide is adjustable from a low of 0.0% (no glide)
to a high of 100% (maximum glide). A setting of
100% will take the longest time to travel from
the current note to the next note played. The effect is also dependent on the interval (distance
of pitch) between the two notes: The larger the
interval, the more noticeable the effect.
Bend Range
Bend Range sets the maximum interval of pitch
bend that can be applied to Bruno with a MIDI
controller’s pitch bend wheel. This control is adjustable from 0 semitones (no bend) to 12 semitones (1 octave).
Master Tune
Master Tune can be used to tune the pitch of
Bruno’s output to another instrument. By default, this control is set to 440.0 Hz It can be adjusted from a low of 430.0 Hz to a high of
450.0 Hz.
Chapter 77: Bruno and Reso
391
Detune Amount
Bruno Voice Controls
Detuning is a common sound-thickening technique used on synthesizers and many effects devices. Bruno’s Detune Amount control sets the
maximum amount of pitch detuning that occurs
when multiple voices are stacked together using
Voice Stacking. Using a combination of voice
stacking and detuning, you can create timbres
that are exceptionally fat.
Voices can be detuned up to 50.0 cents. (One
cent is equal to 1/100th of a semitone.)
Voice controls
Detune Velocity
These controls set Bruno’s voice polyphony and
allocation.
Detune Velocity controls how MIDI key velocity
affects voice detuning. This gives you velocitysensitive control over voice detuning when you
play Bruno with a MIDI keyboard.
This control is adjustable from a low of 0.0 cents
(no velocity-sensitive detuning) to a high of
50.0 cents (maximum velocity-sensitive detuning).
If Detune Velocity is set to 0.0 cents, detuning
will not change no matter how hard you strike a
key on your MIDI controller. Conversely, if you
set Detune Velocity to 50.0 cents, a hard strike
will detune voices a maximum of 50.0 cents (in
addition to the detuning specified with the Detune Amount control).
Detune Velocity has an effect only when you
play Bruno with a velocity-sensitive MIDI
controller.
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Audio Plug-Ins Guide
Mode
Mono (Monophonic) In this mode, Bruno responds monophonically, producing a single note
even if more than one is played simultaneously
(though multiple voices can be stacked on the
same note using the Voice Stacking control).
Monophonic mode gives voice priority to the
most recently played note.
Poly (Polyphonic) In this mode, Bruno responds
polyphonically, producing as many notes as are
played simultaneously (up to 62 on
Pro Tools|HD Accel systems). The number of
notes that can be played simultaneously depends on the Voice Stacking setting chosen. A
voice stack setting of 1, for example, allows up
to 62 individual notes simultaneously. A voice
stack setting of All allows only one note at a
time, but will stack all 62 voices on that note,
producing an extremely fat sound.
Voice Stack
Voice Stack selects the number of voices that are
used, or stacked when you play a single note. The
number of voices that you choose to stack will
directly affect polyphony. Selecting a larger
number of stacked voices will reduce the number of notes that you can play simultaneously.
If all available voices are being used, playing an
additional note will replace the first note played
in the chord.
Reso Controls
Reso synthesizes new harmonic overtones from
the source audio signal, creating harmonically
rich timbres with a metallic, synthesizer-like
character.
Voice Stack
The sample rate of your session also affects polyphony. For example, in a 96 kHz session,
Bruno can simultaneously play up to:
• 32 notes in a 1-voice stack
• 16 notes in a 2-voice stack
• 4 notes in a 4-voice stack
• 2 notes in an 8-voice stack
• 1 note in an 12-voice (All) stack
The 62-voice Bruno requires an HD Accel
card.
In a 44.1 kHz or 48 kHz session on a
Pro Tools|HD system not equipped with an
HD Accel card, Bruno can simultaneously play
up to:
Reso
Reso Timbre Controls
• 24 notes in a 1-voice stack
• 12 notes in a 2-voice stack
• 6 notes in a 4-voice stack
• 3 notes in an 8-voice stack
• 1 note in a 24-voice (All) stack
Voice counts for Bruno for 44.1 kHz and 48 kHz
sessions are the same on Pro Tools|HD-series
systems not equipped with an HD Accel card.
Timbre controls
Chapter 77: Bruno and Reso
393
Reso Resonance Controls
Resonance Amount
Resonance Amount controls the intensity of
harmonic overtones produced by the Resonator.
Increasing the Resonance Amount will increase
the overall harmonic content of the sound while
increasing the sustained portions of the generated harmonics.
The frequency content of the input signal largely
determines what harmonics are generated by the
resonator. For this reason, the character of the
resonance will change according to the type of
audio that you process.
Resonance Velocity
Resonance Velocity increases or decreases resonance according to how hard a MIDI key is
struck and how much resonance is initially specified with the Resonance Amount control.
Resonance Velocity is adjustable from a low of
–10 to a high of +10. With positive values, the
harder the key is struck, the more resonance is
applied. With negative values, the harder the
key is struck, the less resonance is applied.
The effectiveness of this control depends on the
Resonance Amount setting. For example, if Resonance Amount is set to 0, setting the Resonance Velocity to a negative value will have no
effect, since there is no resonance to remove.
Similarly, if the Resonance Amount control is
set to 10, setting Resonance Velocity to +10 will
have no effect since the resonance is already at
its maximum.
For optimum effect, set the Resonance Amount
to a middle value, then set Resonance Velocity
accordingly for the desired effect.
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Resonance Velocity has an effect only when
you play Reso with a velocity-sensitive
MIDI controller.
Reso Damping Controls
Damping Amount
Damping causes the high-frequency harmonics
of a sound to decay more rapidly than the low
frequency harmonics. It lets you control the
brightness of the signal generated by Reso's Resonator and is particularly useful for creating
harp or plucked string-like textures.
The range of this control is from 0 (no damping)
to 10 (maximum damping). The greater the
amount of damping, the faster the high-frequency harmonics in the audio will decay and
the duller it will sound.
Damping Velocity
Damping Velocity increases or decreases damping according to how hard a MIDI key is struck
and how much damping is initially specified
with the Damping Amount control.
Damping Velocity is adjustable from a low of
–10 to a high of +10. With positive values, the
harder the key is struck, the more damping is
applied. With negative values, the harder the
key is struck, the less damping is applied (which
simulates the behavior of many real instruments).
The effectiveness of this control depends on the
Damping Amount setting. For example, if
Damping Amount is set to zero, setting the
Damping Velocity to a negative value will have
no effect, since there is no damping to remove.
Similarly, if the Damping Amount control is
set to 10, setting Damping Velocity to +10 will
have no effect since damping is already at its
maximum.
For optimum effect, set the Damping Amount to
a middle value, then set Damping Velocity accordingly for the desired effect.
Damping Velocity only has an effect when
you play Reso with a velocity-sensitive
MIDI keyboard controller.
The resonator adds harmonic overtones to the
source audio signal that are integer multiples of
the fundamental frequency of the signal. The
Harmonics control selects between all of these
harmonics, or just the odd-numbered intervals.
Your choice will affect the timbre of the sound.
All
Adds all of the harmonic overtones generated by
the resonator. In synthesizer parlance, this produces a somewhat buzzier, sawtooth wave-like
timbre.
External Key
Toggles the harmonics from a separate reference
track or an external audio source. The source
used for toggling is referred to as the key input
and is selected using the Side-chain Input popup. You can assign either an audio input channel
or a TDM bus channel.
Typically, a drum track is used as a key input so
that toggling occurs according to a definite
rhythmic pattern.
Key Listen
When enabled, monitors the source of the key
input. It is useful to do this to fine tune Reso’s
settings to the key input.
See “Using an External Key Input with
Bruno/Reso” on page 387.
Threshold
Odd
Adds only the odd-numbered harmonic overtones generated by the resonator. In synthesizer
parlance, this produces a somewhat more hollow, square wave-like timbre.
Reso Toggle Controls
Reso can automatically toggle between the All
and Odd harmonics settings, producing a rhythmic pulse in the timbre.
Harmonic toggling can be controlled either by
triggering (using the dynamics of the source audio itself, or those of an external key input) or
by MIDI Beat Clock.
Sets the level in decibels above which toggling
occurs. When the audio input level rises above
the Threshold level, Reso will toggle its harmonics setting. The range of this control is from a
low of –48 dB (maximum toggling) to a high of
0.0 dB (no toggling). If no key input is used, the
dynamics of the source audio will trigger toggling. If a key input is used, the dynamics of the
key input signal will trigger toggling. Threshold-based switching can be used at the same
time as Key Input-based switching.
MIDI Clock
Triggers toggling in sync with a MIDI Beat Clock
signal. This creates a very regular, highly rhythmic wave sequencing effect that is ideal for sessions arranged around MIDI beat clock. This
control can be set to quarter, eighth, or sixteenth notes, or dotted triplet values of the
same.
Chapter 77: Bruno and Reso
395
For quick numeric entry of MIDI beat clock
values, type “4,” “8,” or “16” for quarter
notes, eight notes, or sixteenth notes. Add
“t” for triplets, or “d” for dotted note values.
Typing “4t” for example, enters a quarter
note triplet value. Typing “16d” enters a
dotted sixteenth note value.
Reso Amplitude Controls
If you set Gain Velocity to –24 dB, a soft strike
on a key will reduce gain up to –24 dB. A hard
strike will have a maximum output level equal to
the current dB setting of the Gain Amount control.
Conversely, if Gain Velocity is set to 0.0 dB,
Reso’s volume will not change no matter how
hard or soft you strike a key on your MIDI controller).
Gain Velocity only has an effect when you
play Reso with a velocity-sensitive MIDI
keyboard controller.
Mix
Mix adjusts the mix of the processed audio with
the original, unprocessed audio.
Spread
Amplitude controls
Gain Amount
Gain Amount attenuates output level gain. Since
resonation can cause extreme changes in signal
level, this is particularly useful for preventing
clipping and achieving unity gain with the original signal level. This control is adjustable from
a low of –96 dB (no gain) to a high of 0.0 dB
(maximum gain).
Gain Velocity
Gain Velocity sets the velocity sensitivity of the
Gain Amount control. This gives you touch-sensitive control over Reso’s volume using a MIDI
keyboard.
This control is adjustable from a low of –24 dB
(maximum velocity sensitivity) to a high of
0.0 dB (no velocity sensitivity).
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Audio Plug-Ins Guide
When Reso is used in stereo, the Spread control
can be used to pan multiple Reso voices within
the stereo field. This control is adjustable from
0% (no stereo spread) to 100% (maximum stereo
spread).
Voice stacking affects stereo Spread. For example, setting Voice Stack to 1 and Spread to 100%
will alternately pan each note played right and
left. Setting Voice Stack to 4 and Spread to
100%, will pan two of the five voices hard left,
and two voices hard right.
ADSR Envelope Generator
The ADSR (attack, decay, sustain, release) Envelope Generator controls Reso’s amplitude envelope. This amplitude envelope is applied to a
sound each time a note is struck.
The four envelope elements can be adjusted by
dragging the appropriate breakpoint, or by typing in a numeric value.
Attack Controls the amount of time in milliseconds that the sound takes to rise from zero amplitude to its full level. The longer the attack, the
more time it takes for the sound to reach maximum volume after the a note is struck. This control is adjustable from 0.0 to 5000 milliseconds.
Decay Controls the amount of time in milliseconds that the sound takes to fall from its peak
Attack level to the Sustain level. This control is
adjustable from 0.0 ms to 5000 ms.
Sustain Level Controls the amplitude level in dB
that is reached after the decay time has elapsed.
The amplitude level stays constant as long as a
MIDI note remains depressed. This control is
adjustable from –96 dB (no sustain) to 0.0 dB
(maximum sustain).
Release Controls the amount of time in milli-
seconds that the sound takes to fall from the
Sustain level to zero amplitude after a note is released. This control is adjustable from 0.0 ms to
5000 ms.
Reso Pitch Controls
Glide is adjustable from a low of 0.0% (no glide)
to a high of 100% (maximum glide). A setting of
100% will take the longest time to travel from
the current note to the next note played. The effect is also dependant on the interval (distance
of pitch) between the two notes: The larger the
interval, the more noticeable the effect.
Bend Range
Bend Range sets the maximum interval of pitch
bend that can be applied to Reso with a MIDI
controller’s pitch bend wheel. This control is adjustable from 0 semitones (no bend) to 12 semitones (1 octave).
Master Tune
Master Tune can be used to tune the pitch of
Reso’s output to another instrument. By default,
this control is set to 440.0 Hz It can be adjusted
from a low of 430.0 Hz to a high of 450.0 Hz.
Detune Amount
Detuning is a common sound-thickening technique used on synthesizers and many effects devices. Reso’s Detune Amount control lets you set
the maximum amount of pitch detuning that occurs when multiple voices are stacked together
using Voice Stacking. Using a combination of
voice stacking and detuning, you can create timbres that are exceptionally fat.
Voices can be detuned up to 50.0 cents. (One
cent is equal to 1/100th of a semitone.)
Pitch controls
Detune Velocity
Glide
Detune Velocity controls how MIDI key velocity
affects voice detuning. This gives you touchsensitive control over voice detuning when you
play Reso with a MIDI keyboard.
Glide, also known as portamento, determines
the amount of time it takes for a pitch to glide
from the current note to the next note played.
This effect is commonly used on synthesizers.
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This control is adjustable from a low of 0.0 cents
(no velocity-sensitive detuning) to a high of
50.0 cents (maximum velocity-sensitive detuning).
If Detune Velocity is set to 0.0 cents, detuning
will not change no matter how hard or soft you
strike a key on your MIDI controller. Conversely, if you set Detune Velocity to 50.0 cents,
a hard strike will detune voices a maximum of
50.0 cents.
Detune Velocity only has an effect when you
play Reso with a velocity-sensitive MIDI
keyboard controller.
Reso LPF/Voice Controls
LPF and Voice controls
Reso LPF (Low Pass Filter) Controls
The range of this control is from 0 to 10.
Follower The Follower is an envelope follower
that lets the filter cutoff frequency dynamically
follow the amplitude of the source audio signal.
The range of this control is from a low of –10 to
a high of +10. With positive values, the louder
the source audio, the higher the cutoff frequency
and the wider the filter will open for a brighter
sound. With negative values, the louder the
source audio, the lower the cutoff frequency and
the more the filter will close for a duller sound.
The effectiveness of the Follower depends on the
filter’s Frequency setting. For example, setting
the Follower to +10 and selecting a low Frequency setting will sweep the filter wide on loud
passages. However, if the cutoff frequency is at
its maximum, setting the Follower to +10 will
not sweep the filter at all since it is already completely open.
Reso’s Low Pass Filter is a single resonant filter
that is applied to all of Reso’s voices.
When used with high Q settings and a relatively
low cutoff frequency, the Follower can be used
to produce an automatic wah-wah-type effect.
Frequency The Frequency control sets the cut-
Mono (Monophonic)
off frequency of the Low Pass Filter in Hertz. All
frequencies above the selected cutoff frequency
will be attenuated.
The range of this control is from 20 Hz to
20 kHz.
Q Sometimes referred to as resonance on synthesizers, Q adjusts the height of the resonant peak
that occurs at the filter’s cutoff frequency.
398
Increasing the Q increases the volume of frequencies near the filter’s cutoff frequency (suppressing the more remote frequencies) and adds
a nasal quality to the audio. High Q settings let
you create wah-wah type effects, particularly
when the filter is swept with the Follower.
Audio Plug-Ins Guide
In this mode, Reso responds monophonically,
producing a single note even if more than one is
played simultaneously (though multiple voices
can be stacked on the same note using the Voice
Stacking control). Monophonic mode gives
voice priority to the most recently played note.
Poly (Polyphonic)
In this mode, Reso responds polyphonically,
producing as many notes as are played simultaneously (up to 62 on Pro Tools|HD Accel systems). The number of notes that can be played
simultaneously depends on the Voice Stacking
setting chosen. A voice stack setting of 1, for example, allows up to 62 individual notes simultaneously. A voice stack setting of All allows only
one note at a time, but will stack all 62 voices on
that note, producing an extremely fat sound.
Polyphony will be reduced by half at
96 kHz.
Voice Stack
Voice Stack selects the number of voices that are
used, or stacked when you play a single note. The
number of voices that you choose to stack will
directly affect polyphony. Selecting a larger
number of stacked voices will reduce the number of notes that you can play simultaneously.
The sample rate of your session will also affect
polyphony.
In a 96 kHz session, Reso on Pro Tools|HD Accel
systems can simultaneously play up to:
• 32 notes in a 1-voice stack
• 16 notes in a 2-voice stack
• 4 notes in a 4-voice stack
• 2 notes in an 8-voice stack
• 1 note in an 14-voice (All) stack
In a 44.1 kHz or 48 kHz session on Pro Tools|HD
systems not equipped with an HD Accel card,
the standard Reso module can simultaneously
play up to:
• 28 notes in a 1-voice stack
• 14 notes in a 2-voice stack
• 7 notes in a 4-voice stack
• 3 notes in an 8-voice stack
• 1 note in a 28-voice (All) stack
If all available voices are being used, playing an
additional note will replace the first note played
in the chord.
Voice Stack
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Chapter 78: Click
Click is a metronome plug-in that is available in
TDM and RTAS formats.
Creating a Click Track
The Click plug-in creates an audio click during
session playback that you can use as a tempo reference when performing and recording. The
Click plug-in receives its tempo and meter data
from the Pro Tools application, enabling it to
follow any changes in tempo and meter in a session. The Click plug-in is a mono-only plug-in.
Several click sound presets are included.
To create a click track with the Click plug-in:
1
Ensure that the Options > Click is enabled.
2
Choose Track > Create Click Track.
Pro Tools creates a new Auxiliary Input track
named “Click” with the Click plug-in already inserted. In the Edit window, the track’s Track
Height is set to Mini.
To manually create a click track with the Click
plug-in:
Select Options > Click to enable the Click option (or enable the Metronome button in the
Transport).
1
Click plug-in
2
Click Controls
3
MIDI In LED Illuminates each time the Click
plug-in receives a click message from the
Pro Tools application, indicating the click
tempo.
Accented Controls the output level of the accent
beat (beat 1 of each bar) of the audio click.
Create new a mono Auxiliary Input track and
insert the Click plug-in.
Select a click sound preset.
4 Choose Setup > Click/Countoff and set the Click
and Countoff options as desired.
The Note, Velocity, Duration, and Output
options in this dialog are for use with MIDI
instrument-based clicks and do not affect
the Click plug-in.
Unaccented Controls the output level of the unaccented beats of the audio click.
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Click Options dialog
5 Begin playback. A click is generated according
to the tempo and meter of the current session
and the settings in the Click/Countoff Options
dialog.
Refer to the Pro Tools Reference Guide for
more information on configuring Click options.
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Chapter 79: DB-33
DB-33 Organ page overview
DB-33 is a virtual organ that recreates the
sounds and controllability of classic tonewheel
organs, and the rotary-speaker cabinets they are
often played through. DB-33 can also be used as
an insert effect on an audio track. DB-33 is an
RTAS plug-in that is part of the Avid Virtual Instrument collection of plug-ins.
DB-33 Controls
DB-33’s control layout has two main pages. The
Organ page contains most of the main tonal controls, and the Cabinet page contains the controls
concerning the rotating-speaker cabinet. Once
you’ve got a feel for the various sections within
the interface, you’ll soon be creating classic vintage organ sounds.
DB-33 Organ Page Controls
The Organ page holds the controls that effect the
tone of the organ itself. These controls are also
the most likely to be manipulated while the instrument is played.
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403
Tonewheels
Scanner Vibrato
Tonewheel organs are based on a system of spinning, serrated metal wheels whose motion is
translated into sound by magnetic pickups.
Their condition effects the overall tone of the
organ.
Like many vintage organs, DB-33 features vibrato/chorus to animate the organ sound.
Scanner Vibrato controls
Tonewheels control
The Tonewheels control sets the condition of the
tonewheels, and even provides a couple of nonstandard choices.
Tonewheel
Description
Dirty
Worn-out tone generator. Drifting
pitch, leaking
Used
Like Dirty, but not as extreme
New
Brand-new tonewheels. Clean
tone
Syn 1
Triangle wave, for emulating synthesized organ sounds
Syn 2
Square wave, for emulating synthesized organ sounds
On-Off Turns the chorus/vibrato effect on and
off.
Vibrato and Chorus Sets the chorus/vibrato effect’s mode. The following options are available:
Wave Shape
Description
V1, V2, V3
Vibrato (pitch modulation) effect.
Higher numbered modes offer
stronger modulation
C1, C2, C3
Chorus (timbral modulation)
effect. Higher numbered modes
offer stronger modulation
Drawbars
The most often-used set of controls on most
tonewheel organs, drawbars are used to manipulate the mixture between the various harmonics generated by the tonewheel mechanism.
Drawbars
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From left to right, each drawbar controls a different part of the harmonic spectrum, ranging
from low fundamentals to high harmonics.
Short/Long Sets the length of the harmonic
When a drawbar is pulled out (downward), the
volume of the corresponding harmonic is increased. When pushed in, (upward) it is decreased.
the 3rd or 2nd harmonic.
Key Click
Originally an artifact of the mechanical nature
of tonewheel organs, DB-33 gives you control
over the clicking sound made when a key is
played.
burst.
3rd/2nd Sets the harmonic that is added, either
Master Level
This control sets the overall volume level. To
control the level of signal going to the rotary
speaker simulation (thus affecting tonal character), use the Organ control in the Input section
of the Cabinet page. See “DB-33 Cabinet Page
Controls” on page 406. for more information.
Master Level control
Key Click control
Turn the knob to set the level of click, from zero
to full.
Percussion
DB-33’s percussion feature adds a short burst of
additional harmonics at the beginning of each
note played.
Turn this knob to set the overall output volume.
This control is set, by default, the MIDI CC 7.
Rotation Speed Switch
This control switches the speed of rotation for
the rotating speaker cabinet.
Rotation Speed switch
Percussion controls
On/Off Turns the percussion feature on and off.
Move this switch to the left (Slow) to set the rotating speaker cabinet to slow rotation. Move
the switch to the right (Fast) for fast rotation.
Move it to the center (Brake) to stop the rotation, or to slow it temporarily before switching
to another speed.
Loud/Soft Sets the volume of the added harmonic burst.
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The exact speed of each of the Rotation Speed
switch’s modes is set in the Speed Control section in the Cabinet page.
DB-33 Cabinet Page Controls
To access the Cabinet page, click the Cabinet
button. The Cabinet page provides the controls
pertaining to the rotating speaker cabinet and
the organ’s tube preamp. These controls determine the overall tone of the instrument.
Organ Sets the volume of the tone generator signal before it enters the pre-amp. If you’re hearing unwanted distortion, try turning this control down.The control is set to MIDI CC 11, an
expression pedal, to emulate the volume pedal
used for expressive purposes on many classic organs.
Tube Pre-Amp
The Tube Pre-amp section offers control over
the preamplifier that precedes the rotating
speaker cabinet.
Cabinet page overview
Input
The Input section contains the level controls for
the external input (using the preamp and rotating speaker simulation as an effect for other signals in Pro Tools) and the organ’s signal.
Increasing either of these inputs to very high
levels drives the tube pre-amp harder, sometimes leading to pleasant (or not-so-pleasant)
distortion.
Input controls
External Sets the volume of incoming signal
when DB-33 is used as an insert effect.
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Tube Pre-Amp controls
Character Effects the tonal balance of the signal.
Turned to the left, the lows are cut, and high and
midrange harmonics are emphasized. Turned to
the right, lows are boosted and highs are cut.
Drive Sets the amount of gain in the pre-amp,
ranging from clean to distorted.
High Cut Sets the amount of treble roll-off. Used
in conjunction with the Character control (set to
a low value), it creates a mid-heavy but not treble-boosted tone.
Mics
The Mics section controls the balance between
the high and low-end speakers in the rotating
speaker cabinet, and the stereo separation of the
rotating speaker’s movement.
Rotation Speed Switch A duplicate of the switch
on the Organ page, it is present here for ease of
testing speed modes while setting other parameters.
Slow Rate Sets the speed of speaker rotation
when the rotation speed switch is set to Slow
mode.
Fast Rate Sets the speed of speaker rotation
when the rotation speed switch is set to Fast
mode.
Mics controls
Drum/Horn Controls the mix between the lowend speaker (drum) and the high-end speaker
(horn).
Spread Sets the angle, and thus, the stereo response of the two virtual “mics” that pick up the
organ’s signal. Fully left, the mics are placed at
90 degrees from each other. Fully right, the mics
are placed at 180 degrees from each other, accentuating the motion of the signal as the horn
spins.
Speed Control
Acc/Dec Sets the amount of time it takes for the
rotating speaker to move from one speed to another.
DB-33 Info Display and
Organ/Cabinet Switches
Info display and Organ/Cabinet switches
DB-33 has an Info display that shows the setting
of the currently selected control. To the left of
the Info display are the switches that toggle the
main window between the Organ and Cabinet
pages.
The Speed Control section affects the rotating
speaker cabinet’s speed of rotation, and the time
it takes to change between speed modes.
Speed controls
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Inserting DB-33 on a Track
To use an instrument plug-in to its best advantage, insert it on a stereo Instrument track in
your Pro Tools session.
DB-33 can also be used as an insert effect.
To insert DB-33 on an Instrument track:
Create a new stereo Instrument track (recommended) in your Pro Tools session:
1
• Choose Track > New.
• Select 1 new Stereo Instrument track in
Ticks.
• Click Create.
Click the Pro Tools Track Insert selector and
select an instrument.
2
If needed, you can now record-enable the instrument track to enable the use of a MIDI controller to play the instrument and/or help in
creating MIDI sequences within the sequencer
in Pro Tools.
3
See the Pro Tools Reference Guide for instructions on how to use the MIDI sequencer
in Pro Tools.
Using the MIDI Learn
Function on Avid Virtual
Instruments
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Audio Plug-Ins Guide
In addition to pre-assigned MIDI controllers
(such as Sustain Pedal and Volume), you can assign MIDI controllers to parameters within an
Avid Virtual Instrument plug-in for automation
or real-time control from a MIDI keyboard or
control surface. MIDI assignments are saved
with the session.
To assign an Avid Virtual Instrument parameter to
a MIDI controller:
 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Assign and
choose a controller number from the pop-up
MIDI CC list.
-or Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Learn from
the menu and move the desired knob or controller on your MIDI keyboard or sequencer. The instrument plug-in will set this MIDI controller to
the parameter you have chosen.
Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
MIDI CC
Function
120
All Sound Off
121
Reset Controllers
123
All Notes Off
124
Omni Off (Not used in Plug-Ins)
125
Omni On (Not used in Plug-Ins)
126
Mono On (Not used in Plug-Ins)
127
Mono Off (Not used in Plug-Ins)
To remove a MIDI controller assignment:
 Control-click or Right-click (Mac), or Rightclick (Windows) an assigned control and choose
Forget to remove its MIDI controller assignment .
All Avid Virtual Instrument plug-ins have
pre-defined parameter assignments for Avid
and supported third-party hardware control
surfaces.
Set Min/Max
These options let you scale incoming MIDI controller data so that the chosen control does not
go below or above a certain value.
To set the Min/Max level:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, choose Set Min or Set
Max, and select the desired lower or upper limit
for the current control.
Invert Range
This option lets you invert incoming MIDI controller data so that the chosen control reacts in
inverse proportion to the assigned MIDI controller.
This is useful, for example, when you’re assigning the drawbars in DB-33 to a set of MIDI fader
controls, but you want the action to be reversed,
so that the faders work like the drawbars on a
physical organ.
To invert a control’s response:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, and select Invert
Range.
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Chapter 80: Mini Grand
Mini Grand plug-in window
Mini Grand is a simple virtual piano instrument
with seven different acoustic piano sounds to
suit a broad range of musical styles and production needs. Mini Grand is an RTAS plug-in that
is part of the Avid Virtual Instrument collection
of plug-ins.
Mini Grand Controls
Six selectable models of room ambience can be
used to place Mini Grand’s sound into an optimum spatial environment.
You can use the on-screen keyboard to audition
the sound, if a MIDI controller is not within
reach.
The main panel contains controls for choosing
the desired piano model, type and amount of
room simulation, dynamic response, and overall
output level.
By familiarizing yourself with the main controls, you’ll be well on your way to creating perfect piano parts for every occasion.
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411
Mini Grand Main Controls
Room This control switches between various
Model This knob selects between seven different
room ambiences. Effects range from natural reverbs to special effects.
piano models that range from dark and mellow
(Atmo) to bright and aggressive (Dance).
Room control
Mix The Mix knob blends the desired amount of
Model knob
room ambience, selected using the Room knob,
into the piano tone.
Dynamic Response This control adjusts the response of the piano sound to incoming MIDI velocity data. Higher settings give more dynamic
sensitivity, and lower settings create a more
even dynamic response.
If you’re using a MIDI keyboard that tends to
output high velocities without much effort,
turning this control down can help compensate
creating a more natural feel.
Dynamic response control
Tuning Scale This control toggles between Equal
tuning, where the piano’s relative pitch is normal, and Stretched, where the piano’s higher
notes are tuned slightly higher, so they are more
in tune with the overtones of the lower note.
Mix knob
Level Turn this knob to control the overall out-
put volume of Mini Grand.
Level control
Mini Grand Info Display and
Setup Button
Info display (left) and Setup Button (right)
Tuning scale
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Mini Grand has an Info display that shows the
setting of the currently selected control.
To the right of the Info display is the Setup button, which opens the Setup page. The Setup page
offers control of Mini Grand’s Eco Mode, which
reduces CPU load by deactivating string resonances, and the polyphony selector, which sets
Mini Grand’s maximum polyphony.
Room Ambiences
Shaping Mini Grand’s Sound
Mini Grand offers a wide range of tones that can
fit into many genres of music. It is helpful, when
familiarizing yourself with the plug-in, to try
various combinations of sounds and ambience
settings.
Room
Description
Soft
Mellow, sweet reverb
Bright
Heavier early reflections, accentuated highs
Studio
Controlled, tight ambience
Chamber
Longer reverb time with more diffused reflections
Hall
Longest reverb time, biggestsounding room
Ambient
Few reflections, very spatial
Piano Models
See the Pro Tools Reference Guide for instructions on how to access and save Presets
within Mini Grand.
Model
Description
Atmo
Quiet and dark, accentuated lowmids and muted highs
Soft
Mellow and rounded, with accentuated lows and low-mids, and
mellow highs
Inserting Mini Grand on a
Track
Ballad
Dynamic, but understated, warm
lows
Real
Natural and dynamic, little processing
To use an instrument plug-in to its best advantage, insert it on a stereo Instrument track in
your Pro Tools session.
Bright
Controlled and percussive, lowered bass and low-mids
Hard
Loud, pointed, with accentuated
highs, lowered bass and lowmids
Dance
To insert an instrument plug-in on an Instrument
track:
Create a new stereo Instrument track (recommended) in your Pro Tools session:
1
• Choose Track > New.
• Select 1 new Stereo Instrument track in
Ticks.
Loud, aggressive and gritty,
strong upper mid and treble presence, scooped lows
• Click Create.
Click the Pro Tools Track Insert selector and
select an instrument.
2
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413
If needed, you can now record-enable the instrument track to enable the use of a MIDI controller to play the instrument and/or help in
creating MIDI sequences within the sequencer
in Pro Tools.
3
See the Pro Tools Reference Guide for instructions on how to use the MIDI sequencer
in Pro Tools.
Using the MIDI Learn
Function on Avid Virtual
Instruments
In addition to pre-assigned MIDI controllers
(such as Sustain Pedal and Volume), you can assign MIDI controllers to parameters within an
Avid Virtual Instrument plug-in for automation
or real-time control from a MIDI keyboard or
control surface. MIDI assignments are saved
with the session.
To assign an Avid Virtual Instrument parameter to
a MIDI controller:
 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Assign and
choose a controller number from the pop-up
MIDI CC list.
-or-
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 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Learn from
the menu and move the desired knob or controller on your MIDI keyboard or sequencer. The instrument plug-in will set this MIDI controller to
the parameter you have chosen.
Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
MIDI CC
Function
120
All Sound Off
121
Reset Controllers
123
All Notes Off
124
Omni Off (Not used in Plug-Ins)
125
Omni On (Not used in Plug-Ins)
126
Mono On (Not used in Plug-Ins)
127
Mono Off (Not used in Plug-Ins)
To remove a MIDI controller assignment:
 Control-click or Right-click (Mac), or Rightclick (Windows) an assigned control and choose
Forget to remove its MIDI controller assignment .
All Avid Virtual Instrument plug-ins have
pre-defined parameter assignments for Avid
and supported third-party hardware control
surfaces.
Set Min/Max
These options let you scale incoming MIDI controller data so that the chosen control does not
go below or above a certain value.
To set the Min/Max level:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, choose Set Min or Set
Max, and select the desired lower or upper limit
for the current control.
Invert Range
This option lets you invert incoming MIDI controller data so that the chosen control reacts in
inverse proportion to the assigned MIDI controller.
To invert a control’s response:
 Control-click or Right-click (Mac) or
Right-Click (Windows) a control, and select
Invert Range.
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Chapter 81: Structure Free
Structure Free is a sampler plug-in that brings
the world of Structure compatible sample libraries to any Pro Tools system and delivers superior performance and reliability thanks to its direct integration with Pro Tools. Structure Free
is an RTAS plug-in that is part of the Avid Virtual Instrument collection of plug-ins. Structure
Free comes with its own 600 MB sample library
to get you started.
For detailed information about the full
version of Structure, see the A.I.R. Virtual
Instruments Plug-Ins Guide.
Structure Free Features

64-voice multitimbral sound engine
 Loads all Structure compatible sample libraries (native Structure, SampleCell, SampleCell II,
Kontakt, Kontakt 2, and EXS 24)
Structure Free Keyboard
Section Controls
The Structure Free Keyboard section provides
the following controls:
Keyboard The Keyboard provides 88 keys for
playing Structure Free, six Smart Knobs, and a
context sensitive Info display, as well as the
Master volume control for the whole plug-in.
You can play and control Structure Free by
clicking the keys, using MIDI input from a MIDI
keyboard, or using MIDI data in an Instrument
or MIDI track in Pro Tools. When Structure Free
receives MIDI data, the keys reflect the MIDI
note input.
Keyboard
Full compatibility with all Structure versions
(Structure), you can easily upgrade and still use
your Pro Tools sessions created with Structure
Free. You can also open sessions which originally used Structure with Structure Free


Sample playback using disk streaming or RAM
 Support of all common bit depths and
sample rates up to 192 kHz
 Easy real-time sound manipulation using
Smart Knobs
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417
Smart Knobs The Smart Knobs are special controls which can be assigned to one or more
Structure Free parameters in the currently selected patch. These parameters can then be remote controlled at the same time by moving the
Smart Knob. This comes in handy for easily designing complex sounds or quickly adjusting a
patch to suit your session in terms of feel, timbre, enveloping, or any other sensible sound
shaping parameter. In Structure Free’s factory
content, each patch has Smart Knobs pre-assigned to important parameters. The Smart
Knob can be named in the field above each knob.
Info Display The Info display above the Keyboard section is a context-sensitive text display.
When you load something into Structure Free, it
displays a progress bar. When loading a commented patch, it displays the Patch comment.
When editing controls, it displays parameter
name and value.
Info display
Editing a Patch Comment
To edit a patch comment:
1
Select a patch.
2
Double-click into the Info display.
Smart Knobs
3
Type in your comment.
To display a control’s current value:
4
Press Enter.

Click the control without moving the mouse.
Key Switches Key Switches are special MIDI
notes or keys that are assigned to controls and
act as a switch. For example, they can switch between different Smart Knob settings for a patch.
Master (Output Volume) The Master control adjusts the volume of all Structure Free outputs to
Pro Tools. All patches are mixed down to the
Main output by default, and then output to the
Instrument, Auxiliary Input, or Audio track on
which Structure Free is inserted.
Adjusting the Master output control
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The Display does not show parameter values
of incoming automation, as multiple parameters in different patches could be
changing simultaneously. Only values edited using the mouse are shown.
Structure Free Patch List
Controls
In the Patch list on the left side of Structure
Free, you can create, select, mix, MIDI-assign,
route, and group patches.
Click a Patch module to select it for editing in
the Parameter panel. The handle on the left of
the selected patch is lit. When a patch is selected
all of its parameters are displayed in the Parameter panel on the right and assorted into subpages.
Structure Free Patches. See “Structure Free
Browser Page Controls” on page 423 for more
information on how to add a folder to your favorites.
Mute Button Mutes the patch.
Solo Button Solos the patch.
Volume Fader Adjusts the Patch volume.
Panorama Fader Adjusts the patch’s position in
the stereo panorama.
MIDI Channel Selector Selects the channel on
which the patch receives MIDI data.
Structure Free Patch Menu
The Structure Free Patch menu provides the following controls:
Load New Patch The Load Patch entry brings up
a dialog for selecting a patch that will be added
below the currently selected patch in the Patch
list.
Patch list
Structure Free Patch Module
Controls
Add Patch The Add Patch submenu lets you add
a new patch to the end of the Patch list. Like the
Quick Browse Menu, it gives access to your Favorite folders for loading patches.
Duplicate Patch The duplicate Patch entry adds
an exact copy of the selected patch below it in
the Patch list.
Patch module showing Quick Browse menus (left) and
Panorama Faders (right)
Quick Browse Menu for Favorite Folders Gives
quick access to the factory content folders and
folders that have been added to the favorites.
Click the double arrow to bring up the favorite
folders menu from which you can directly select
Remove Patch The Remove Patch entry unloads
the selected patch removing it from the Patch
list.
Remove All Patches The Remove All Patches en-
try clears the Patch list of all loaded patches.
Click OK in the prompted security dialog if you
really want to clear the whole Patch list.
Cut Patch The Cut Patch entry copies the selected patch to the clipboard and removes it
from the Patch list.
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419
Copy Patch The Copy Patch entry copies the selected patch to the clipboard.
Paste Patch The Paste Patch entry inserts the
copied patch on the clipboard at the end of the
Patch list.
Paste Patch Parameter The Paste Patch Parame-
ter entry inserts only the parameter settings of
the copied patch to the selected patch.
Loading a Patch from the Structure Free Patch
Menu
Samples to Session Folder function to transfer
the loaded samples to your computer’s disk.
After transferring the samples, Structure Free
can load the concerned patches without requiring the source CD, DVD, or network folder.
Selected Patch copies the samples of the selected patch to disk.
All Patches copies the samples of all patches of
the Structure Free instance to disk.
Session copies the samples of all patches of all
Structure Free instances in your session to disk.
To load a patch from the Patch menu:
1
Go to the Patch menu and click Load Patch.
In the following file dialog, locate and select a
patch.
2
3
Click OK.
Automation and Structure Free
Patches
Structure Free automatically assigns an automation channel to each patch, each of which provides automation for the most important Patch
parameters like level, solo, mute, and Smart
Knobs. In the Pro Tools plug-in automation dialog, the automatable parameters for each channel are distinguishable by the corresponding letter. For example, A Level for the Volume fader
of the patch assigned to automation channel A.
Automation channels are assigned subsequent
to the patches in the Patch list by default. The
currently selected patch’s assignment is displayed in the Patch menu.
Copying Structure Free Samples
to Session Folder
If you have loaded patches from removable
media like a CD, DVD, or over the network into
Structure Free, a yellow exclamation mark symbol indicates the affected patches. Use the Copy
420
Audio Plug-Ins Guide
Adding Additional Structure
Free Patches
Additional factory patches for Structure Free
can be downloaded from Avid’s website
(www.avid.com).
To access additional Structure Free patches
through the Quick Browse Menu, you must manually add them to the “Structure QuickStart”
folder.
To add Structure Free patches:
Download the Structure Free patches from the
Avid website (www.avid.com). After downloading, make sure the patches are uncompressed.
1
Drag the uncompressed downloaded patches
into the Structure QuickStart folder, located on
your computer at the following location:
2
• Applications/Digidesign/Structure/Structure QuickStart (Mac)
– or –
• Applications/Digidesign/Structure/Structure QuickStart (Windows)
Structure Free Main Page Controls
After inserting Structure Free, the Main page is selected by default. Coming from the Browser page,
click the Main tab to access the parameters for patches. The Main page provides easy access to all useful parameters like transposition and filter within two sub-pages. If a patch gets selected Structure
Free switches automatically to the Main page.
A patch’s parameters on Main page
Structure Free Edit Sub-Page
Controls
Structure Free provides two Edit subpages, accessible from the Structure Free Main page. The
Edit sub-pages provide patch editing controls.
To access the Edit sub-pages for the selected
patch:
Click the sub-page tabs in the Parameter
panel.

Structure Free Edit 1 Sub-Page Controls
Octave Transposes the incoming MIDI notes for
the patch in octave steps.
Semi Transposes the incoming MIDI notes for
the patch in semitone steps.
Fine Tune Tunes the patch up and down in cents.
Pitch Bend Up Sets the upward pitch bend range
for the patch in semitones.
Pitch Bend Down Sets the downward pitch bend
range for the patch in semitones.
Selecting the Edit 1 sub-page
Max Polyphony Sets the maximum number of
voices available for the patch.
Key Range Sets the key range in which the patch
plays. You can define the upper and lower borders and a transition.
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421
FX Send On Activates the Effect Send for the
patch.
FX Send Level Adjusts the level sent from the
patch to the Effect Send.
Structure Free Edit 2 Sub-Page Controls
Filter Section
Filter Type Selects a filter type.
Cutoff Adjusts the filter cutoff frequency.
Envelope Level Adjusts how strongly the filter
envelope modulates filter cutoff.
Filter Envelope Section
Attack Sets the time needed for the filter enve-
lope to reach its maximum value.
Hold Adjusts the length of the Filter envelope’s
Hold time.
Decay Adjusts the time for the filter envelope
needed to fall from hold level to sustain level.
Sustain Adjusts the level of the sustain segment.
The envelope’s signal remains on this level as
long as the note is held.
Release Adjusts the time for the envelope’s release segment to fall to zero when the note is released. Use shorter times for an immediate closing of the filter. Longer times cause the filter
cutoff to decay slowly.
422
Audio Plug-Ins Guide
Amplifier Section
Vel Sens (Velocity Sensitivity) Adjusts the envelope velocity sensitivity (range in dB between
lowest and highest velocity).
Amp Envelope Section
Attack Softens the attack phase of Instruments
by applying an amplitude envelope to the start
of each Instrument hit. Move the control to the
right to increase the time needed for the attack
to rise to full amplitude.
Hold Adjusts the length of the Amp envelope’s
Hold time at the end of the attack phase.
Decay Shortens the played instrument hits by
applying an amplitude decay after the hold time.
Sustain Adjusts the level of the sustain segment.
The envelope’s signal remains at this level as
long as the note is held.
Release Adjusts the time for the release segment
to fall to zero when the note is released. Use
shorter times for an immediate stop of the
sound. Longer times cause the sound to fade out.
Structure Free Browser Page Controls
The Browser lets you search and display the local file system, as well as letting you load by dragging
and dropping. The Browser is not a file manager. Modifying operations such as copying, moving, or
deleting are not available.
Browser page
Browser Controls
Next directory
Add folder to favorites
Previous directory
New folder
Delete
Folder history
Directory up
Show favorites folders
Refresh view
Browser controls
Previous Directory Navigates to the previous
The Browser page provides the following controls:
folder.
Patch Activates the displaying of only patches.
Next Directory Navigates to the next folder.
Parts Activates the displaying of only parts.
Directory Up Navigates one folder level up.
Sample Activates the displaying of only sam-
Show Favorites Shows your Favorite folders.
ples.
Show All Activates the displaying of all file
types.
Add to Favorites Adds the selected folder to
your Favorite folders (accessible through the up
and down arrows in the patch module).
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423
New Folder Creates a new folder.
Delete Deletes the selected file or folder.
Folder History Shows the 20 last selected folders.
Inserting Structure Free on a
Track
To insert Structure Free on an Instrument track:
Create a new stereo Instrument track in
Pro Tools.
1
To load a patch from the browser:

Drag a patch into the Patch list to load it.
Click the track’s Insert selector and choose
Structure Free from the list.
2
To replace a patch using the browser:
 Drag a patch onto another in the Patch list to
replace it at the same position using the previous
settings for MIDI input, Individual output, and
Automation channel.
To create a New Structure Free Patch from an
Audio File
 Drag one or more audio files into the Patch list
to load; a new patch is created.
Using Structure Free
The following section helps you to explore
Structure Free’s basic concepts with a hands-on
approach. You will touch the most important
functions, understand the basic concepts and
make the first guided steps to get Structure Free
to sound.
For details on how to assign MIDI controllers, see “Using the MIDI Learn Function on
Avid Virtual Instruments” on page 426.
Making Sound with Structure
Free
To make sound with Structure Free:
1 If you have a MIDI keyboard available and prefer to use it, connect it to Structure Free’s MIDI
input, and route it to Structure Free on MIDI
channel 1. If there is no MIDI keyboard available, you can play Structure Free by clicking the
keyboard on screen, or using MIDI input from
the Instrument track in Pro Tools.
Play some notes on your MIDI keyboard. If all
is well so far, you are hearing a sine wave signal
from the default Sine Wave Patch at the top of
the Patch list.
2
The default Sine Wave patch
3
424
Audio Plug-Ins Guide
Next, load a Structure Free patch.
Loading a Structure Free Patch
To load a patch from the Browser:
Click the Browser tab in the Parameter panel
to display the Browser page.
1
Finding Missing Structure Free
Samples
If a loaded patch does not find its samples because folders have been renamed or moved to
another location, you can use the Find Missing
Samples file dialog to point Structure Free to the
new location of the samples. Patches which are
missing samples are indicated by a red exclamation mark symbol.
Missing samples
Browsing for patches
To find missing samples for a patch:
Click your way through the folders to access
the QuickStart content folder. If you chose the
suggested path during installation, it is located
here, depending on your OS:
2
Windows Program Files\Digidesign\
Structure\Structure QuickStart
Mac OS X /Applications/Digidesign/
Structure/Structure QuickStart
Click the Patch menu and select Find Missing
Samples from the menu.
1
In the following dialog, navigate to the new
sample location and click OK.
2
Full Recursive Search 3 Searches for missing
samples in the specified folder and all its subfolders.
Drag the Patch named 01 Six String Guitar.patch onto the Sine Wave Patch to load it and
Using Structure Free Smart
Knobs
replace the Sine Wave patch. A red frame around
the patch when dragging indicates that you are
replacing the existing patch with the new one.
Wait until the Loading message in the display
beneath the Parameter panel disappears.
The Smart Knobs are special controls which can
be assigned to one or more Structure Free parameters in the currently selected patch.
3
After loading, the multi-purpose display
shows a short description of the Patch, and the
Parameter panel above displays its Patch parameters.
4
Play some notes and chords. Adjust the Patch
volume using the horizontal fader on the Patch
module in the Patch list.
5
To use Structure Free Smart Knobs:
Every Patch has six Smart Knob assignments
which are (in the factory content) pre-assigned
to useful parameters. You can use them to easily
adjust a patch to fit your session. Select the
patch to display its Smart Knob assignments in
the Keyboard section.
1
2
Set the Smart Knob for Delay Mix to 30%.
3
Set the Smart Knob for Chorus Mix to 65%.
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425
Play some notes and chords. Set the other
Smart Knobs at will.
4
Using the MIDI Learn Function
on Avid Virtual Instruments
Smart Knob
Using Structure Free Key
Switches
Key Switches are special MIDI notes or keys that
are assigned to switch control values instead of
triggering notes. For example, they can switch
between different Smart Knob settings for a
Patch or mute certain parts within a patch.
To use Key Switches with Structure Free:
Load Patch 04 Electronic Drum Kit.patch, and
play with it on your keyboard.
In addition to pre-assigned MIDI controllers
(such as Sustain Pedal and Volume), you can assign MIDI controllers to parameters within an
Avid Virtual Instrument plug-in for automation
or real-time control from a MIDI keyboard or
control surface. MIDI assignments are saved
with the session.
1
2 The different Effects in this specific Patch are
not audible initially. Their Smart Knobs are assigned to Key Switches so you can mix them in
by just clicking or playing a Key Switch. All
available Key Switches appear blue on the screen
keyboard. The currently activated Key Switch is
green. After activating a Key Switch, a short description is shown in the multi-purpose display.
A Key Switch does not trigger samples that are
mapped in the corresponding key range.
Click the second Key Switch C#0, or play the
corresponding key to add dirt to the kit’s sound.
3
Key Switches
426
4
Try out the other Key Switches.
5
The synth pad patch has Key Switches too.
Audio Plug-Ins Guide
To assign an Avid Virtual Instrument parameter to
a MIDI controller:
 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Assign and
choose a controller number from the pop-up
MIDI CC list.
-or Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Learn from
the menu and move the desired knob or controller on your MIDI keyboard or sequencer. The instrument plug-in will set this MIDI controller to
the parameter you have chosen.
Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
MIDI CC
Function
120
All Sound Off
121
Reset Controllers
123
All Notes Off
124
Omni Off (Not used in Plug-Ins)
125
Omni On (Not used in Plug-Ins)
126
Mono On (Not used in Plug-Ins)
127
Mono Off (Not used in Plug-Ins)
To remove a MIDI controller assignment:
Control-click or Right-click (Mac), or Rightclick (Windows) an assigned control and choose
Forget to remove its MIDI controller assignment .

All Avid Virtual Instrument plug-ins have
pre-defined parameter assignments for Avid
and supported third-party hardware control
surfaces.
Set Min/Max
These options let you scale incoming MIDI controller data so that the chosen control does not
go below or above a certain value.
This is useful, for example, when you want to
keep the Cutoff control on Vacuum from going
above a certain amount, but you don’t want to
have to pay close attention to how you move the
MIDI controller knob you’ve assigned to it.
To set the Min/Max level:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, choose Set Min or Set
Max, and select the desired lower or upper limit
for the current control.
Invert Range
This option lets you invert incoming MIDI controller data so that the chosen control reacts in
inverse proportion to the assigned MIDI controller.
This is useful, for example, when you’re assigning the drawbars in DB-33 to a set of MIDI fader
controls, but you want the action to be reversed,
so that the faders work like the drawbars on a
physical organ.
To invert a control’s response:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, and select Invert
Range.
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427
428
Audio Plug-Ins Guide
Chapter 82: TL Drum Rehab
TL Drum Rehab is an RTAS plug-in for
Pro Tools that provides engineers with a powerful tool for the precise drum replacement and
enhancement of drum tracks in real-time, regardless of performance, equipment, or recording limitations in the original track. Use
TL Drum Rehab to do everything from replacing
poor drum sounds to remixing drum performances with completely new and different
sounds.
TL Drum Rehab is a mono plug-in only. It
cannot be used on multi-channel tracks
(stereo or greater).
TL Drum Rehab Features
• Editable sample-accurate trigger locations
• Dynamic multi-sample support of up to 16
layers (Zones)
• Envelope and tone shaping controls
• Undo
• Powerful sample browser and converter
• Favorites
• Custom file format (DRP)
• Tracking, compression, and quantization
• Triggering sensitivity and filtering controls
• Random sample selection
• No Latency mode
TL Drum Rehab plug-in
Chapter 82: TL Drum Rehab
429
TL Drum Rehab Overview
TL Drum Rehab can be used to reinforce a drum
performance with sampled drum sounds or can
be used to replace the original drum sounds entirely with sampled drums.
Replacing a Kick Drum Sound
Using TL Drum Rehab
To replace a kick drum sound using TL Drum
Rehab:
1 Insert TL Drum Rehab on a mono audio track
of a kick drum recording.
For most applications of TL Drum Rehab you
only need to use the Trigger panel (see “TL
Drum Trigger Panel Display and Controls” on
page 435).
For more complicated drum parts, you may
want to use the Expert panel to commit or ignore
specific detected triggers, as well as quantize or
edit the location of committed triggers (see “TL
Drum Rehab Expert Panel Display and Controls”
on page 440).
TL Drum Rehab is a mono plug-in only. It
cannot be used on multi-channel tracks
(stereo or greater).
Make a short selection to set TL Drum Rehab’s
parameters. For example, make a two bar selection.
2
To edit sample layers and adjust the sound of
samples, use the Samples panel (“Samples Panel
Display and Controls” on page 444).
See the following workflow examples for using
TL Drum Rehab to replace drum sounds in a
track:
• The first example (“Replacing a Kick Drum
Sound Using TL Drum Rehab” on page 430)
uses TL Drum Rehab to replace the kick
drum sound on a mono kick drum track in
real-time.
In TL Drum Rehab’s Trigger panel (see “TL
Drum Trigger Panel Display and Controls” on
page 435), select Kick from the Detector Mode
pop-up menu (see “TL Drum Rehab Detector
Mode Menu” on page 436).
3
• The second example (“Replacing and Quantizing a High Hat Using TL Drum Rehab” on
page 433) describes a more complicated
procedure, using TL Drum Rehab’s Expert
panel to replace a high hat track and quantize the replacement samples.
Detector Mode pop-up menu (left) and Trigger Panel
button (right)
430
Audio Plug-Ins Guide
Enable Listen mode by clicking the Listen button. The Listen button, located in the bottom left
corner of the plug-in window, lights when Listen
mode is enabled.
4
In this example, there is some bleed from the
snare on the kick track and TL Drum Rehab detected a trigger on one of the snare hits. Adjust
the Minimum Threshold control so that only the
kick drum hits are detected (see “TL Drum Rehab Minimum and Maximum Threshold Controls” on page 439).
6
Minimum Threshold
Detected triggers
Listen Button
Create a Selection memory location for your
two-bar selection. This lets you quickly return to the original selection in case you
want to further adjust TL Drum Rehab’s settings.
Start playback in Pro Tools. As Pro Tools plays
back, TL Drum Rehab “listens” to the track, and
analyzes the audio for attack transients and
marks those sample locations with triggers.
These triggers play back the samples loaded into
TL Drum Rehab to replace or enhance the drum
sounds on the audio track.
5
7 After adjusting the Minimum Threshold, play
back the selection to re-detect triggers.
8 In TL Drum Rehab’s Library browser (see “TL
Drum Rehab Library Browser” on page 446), locate the drum sample or DRP file you want to
load. You can audition samples and DRP files by
enabling the Auto-Audition option and selecting
the sample or DRP file you want audition in the
browser.
DRP files are a collection of samples loaded into
TL Drum Rehab’s Zones and Clips that work together to create a realistic and dynamic drum
sound. For more information on DRP files, see
“TL Drum Rehab DRP Name Display” on
page 436.
Detected triggers
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431
9
Do one of the following:
• To load a DRP file into TL Drum Rehab,
double click the desired DRP file in the Library browser.
– or –
• To load a sample into TL Drum Rehab, double click the desired sample (WAV, AIF, or
SD2) in the Library browser. The sample is
loaded into the currently selected Zone (see
“TL Drum Rehab Velocity Map and Velocity
Zones” on page 437).
10 In the Trigger panel, decrease the Input slider
to lower the volume of the original kick sound,
and increase the Samples slider to increase the
volume of the replacement kick sample. This way
you can effectively augment or replace the original drum sound with the sampled drum sound.
You can also adjust the Dynamics control to have
the amplitude of the original drum sound affect
the playback amplitude (velocity) of the sampled
drum sound. (For more information, see “Playback Controls” on page 439.)
You can also use the Ducking control to
mask track’s audio with the triggered sample (see “Playback Controls” on page 439).
11 In the Pro Tools Transport window, press Return to Zero, and press Play to begin playback
from the beginning of the track. TL Drum Rehab
plays back the selected drum sample at every detected trigger in the original track, all in real
time.
During playback, you can further adjust
TL Drum Rehab’s playback controls as desired
to get just the right blend between the original
drum sound and the replacement drum sound.
Once you are satisfied with the result, do one
of the following:
12
Auto-Audition enabled
Selected sample
• Bus and record the output of TL Drum Rehab to a new audio track.
• Use Bounce to Disk to render the replacement track and import it back into the session. For more information on Bounce To
Disk, see the Pro Tools Reference Guide.
• Leave the plug-in inserted and continue to
use it during playback.
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Audio Plug-Ins Guide
Replacing and Quantizing a High
Hat Using TL Drum Rehab
Using the TL Drum Rehab Expert panel to replace
and quantize a high hat sound:
1 Insert TL Drum Rehab on a mono audio track
containing a high hat recording.
2
As in workflow example 1, do the following:
need to compensate for the delay inherent in
non-close miked recordings (such as overs for
the cymbals). Committed triggers are indicated
by a red arrow.
For more information on working with committed triggers, see “TL Drum Rehab Commit Button” on page 441.
• Load the desired DRP file, or load samples
(WAV, AIF, or SD2) into Zones.
• Make a Timeline selection.
• In the Trigger panel, select the appropriate
Detector Mode setting.
• Enable Listen mode.
• Play back the selection to detect triggers.
3
In the Expert panel, click Commit All.
Commit All button (left) and the Expert panel button
(right)
Committed triggers
4 If there are some committed triggers that you
do not want to play back, click either Uncommit
or Ignore.
Uncommitted triggers do not playback if Listen
mode is disabled, but do playback if it is enabled
(because they are re-detected in Listen mode, so
a new trigger is generated). Ignored triggers do
not playback regardless of whether or not Listen
mode is enabled. When working with committed
triggers, Listen mode is typically disabled so
that TL Drum Rehab doesn’t reanalyze the selection’s attack transients and generate new triggers after you have already edited any committed triggers.
Committed triggers play back regardless of
whether or not Listen mode is enabled. TL Drum
Rehab lets you edit the position of committed
triggers by clicking and dragging, which can be
useful if you are working with drum sounds that
do not have clear attack transients, or if you
Uncommitted trigger (left) and an ignored trigger
(right)
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433
5
Disable Listen mode.
For no latency on playback, enabled No Latency mode (see “Triggering Controls” on
page 439).
6
TL Drum Rehab Main Window Provides access to
four different control panels: Trigger, Expert,
Sample, and Preferences. See “TL Drum Rehab
Main Window” on page 434.
Select the desired quantize resolution from the
Quantize To pop-up menu (see “TL Drum Rehab
Quantize To Menu” on page 443).
7
Adjust the Quantize slider to achieve the desired amount of quantization. 100% hard quantizes committed triggers to the selected
Quantize To resolution (for example, sixteenth
notes).
8
Adjust TL Drum Rehab’s playback controls as
desired (see “Playback Controls” on page 439).
9
Once you are satisfied with the result, do one
of the following:
10
TL Drum Rehab Main window
TL Drum Rehab Library Browser Is to the right
of the Main window and lets you select samples
for playback, and also lets you manage your
sample library. See “TL Drum Rehab Library
Browser” on page 446.
• Bus and record the output of TL Drum Rehab to a new audio track.
• Use Bounce to Disk to render the replacement track and import it back into the session. For more information on Bounce To
Disk, see the Pro Tools Reference Guide.
• Leave the plug-in inserted and continue to
use it during playback.
TL Drum Rehab Main Library browser
TL Drum Rehab Main Window
TL Drum Rehab Controls and
Displays Overview
When using TL Drum Rehab, most operations
take place in one of two displays: the Main window and the Library Browser.
The TL Drum Rehab Main window lets you access four different panels: Trigger, Expert, Samples, and Preferences.
Trigger Panel Provides the most commonly used
controls for detecting triggers and playback
controls (see “TL Drum Trigger Panel Display
and Controls” on page 435).
Expert Panel Lets you precisely edit the place-
ment of triggers (see “TL Drum Rehab Expert
Panel Display and Controls” on page 440).
434
Audio Plug-Ins Guide
Samples Panel Lets you view and manage drum
samples loaded into TL Drum Rehab (“Samples
Panel Display and Controls” on page 444).
Preferences Panel Lets you edit TL Drum Re-
hab’s preferences (see “TL Drum Rehab Preferences Panel Display and Controls” on page 446).
A Note About TL Drum Rehab Control Sliders
TL Drum Trigger Panel
Display and Controls
The Trigger panel provides most of the controls
you need to use TL Drum Rehab. The Trigger
panel lets you identify triggers and set up Velocity Zones for sample playback. Additionally, the
trigger panel provides several playback controls.
TL Drum Rehab has several control sliders that
are global controls and are available in more
than one panel. For example, the A/B Blend control is available in the Trigger, Expert, and Samples panels. Adjusting a global control in one
panel view updates that control in all panel
views. These controls can be automated and are
displayed in a luminous blue.
To access the trigger panel:
A/B Blend slide, a global control
Trigger panel button
Other sliders are unique to a single panel, such
as the Quantize control in the Expert panel.
These controls cannot be automated and are displayed in a luminous gray.
TL Drum Rehab Waveform Display

Click the Trigger Panel button.
The Waveform display provides a graphic representation of the selected track’s audio, and also
displays detected triggers and velocities (amplitudes). Detected triggers are displayed as light
blue lines on the waveform.
Quantize slider, a unique control
Detected triggers
Not all sliders are active controls in every panel.
For example, the last slider in the Trigger panel
is grayed out.
Inactive slider
Detected amplitudes
Waveform display with detected triggers and
amplitudes
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435
If TL Drum Rehab detects unwanted triggers
(such as kick bleed through on the snare
track), refer to the detected amplitude for
the unwanted triggers and adjust the Minimum Threshold control accordingly (see
“TL Drum Rehab Minimum and Maximum
Threshold Controls” on page 439).
You can increase or decrease the vertical zoom
of the waveform in the Waveform display by
clicking on the waveform and dragging up or
down.
TL Drum Rehab Voicing Menu
Use the Voicing pop-up menu to select whether
the triggered sample plays back freely (the entire sample plays when triggered) or is choked
(the triggering of the next sample silences the
sounding sample). Typically, you would select
Free for cymbals, since they tend to ring, and
Choke for drums, like kicks and snares. However, you may find that you get some interesting
effects by trying something a little different,
such as selecting Choke for cymbals.
TL Drum Rehab Detector Mode
Menu
Use the Detector Mode pop-up menu to select
the algorithm for trigger detection. TL Drum
Rehab provides four detection algorithms: Snare
Mode 1, Snare Mode 2, Kick, and Tom.
Snare 1 Use Snare 1 for detecting flams and
rolls. Snare 1 is a more sensitive trigger for busier snare tracks.
Snare 2 Use Snare 2 for detecting snare hits and
cymbals. Snare 2 is a more general purpose detection setting.
Kick Use Kick for lower frequency sounds.
Tom Use Tom for mid-range sounds.
Depending on the type of material on the track,
experiment and try different settings to get the
results you want.
Selecting the voicing
TL Drum Rehab DRP Name
Display
The DRP Name display displays the name of the
currently loaded DRP file above the Waveform
display in the Trigger and Samples panels. DRP
files are a collection of samples loaded into TL
Drum Rehab’s Zones and Clips that work together to create a realistic and dynamic drum
sound. DRP files can contain a up to 16 Zones,
two positions (A and B), and four clips per position. TL Drum Rehab comes with a full library of
DRP files.
DRP display
To load a DRP file:
 In the Library browser, locate and doubleclick the DRP file you want to load. All samples
in the DRP file are loaded into their assigned
Zones and Clips.
Selecting the detection algorithm from the Detector
Mode pop-up menu
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Audio Plug-Ins Guide
TL Drum Rehab # of Zones
The # of Zones pop-up menu lets you select the
number of Velocity Zones into which you can
load samples. Use multiple Zones to load samples of different dynamics, but use only as many
Velocity Zones as necessary to layer dynamically
differentiated samples for play back at varying
velocities. For example, using four Zones, you
can load in, from left (quiet) to right (loud), a p
snare sample, an mf snare sample, a f snare sample, and an ff snare sample. During playback,
each Zone is triggered only by the corresponding amplitude of the detected transient so that a
soft hit on the original snare track triggers the p
snare sample and a loud snare hit triggers the f
or ff snare sample.
When using only one or a just a few Velocity
Zones, you may want to use the Dynamics
control to affect the playback velocity by the
detected amplitude on the original drum
track. The Dynamics slider controls the amplitude (velocity) of the triggered sample
relative to the original detected amplitude.
When a more natural sounding drum track
is desired, using multiple Velocity Zones
more closely models the sound of acoustic
drums at different dynamic levels. For more
information on the Dynamics control, see
“Playback Controls” on page 439.
TL Drum Rehab lets you have up to 16 Velocity
Zones, and up to 4 Clips (samples) per Zone. Using slightly different sounds on multiple Clips
per Zone adds a greater degree of realism by
adding variety to the sound (see “Clips” on
page 444).
Selecting the number of Velocity Zones
TL Drum Rehab Velocity Map
and Velocity Zones
The Velocity Map, below the Waveform display,
graphically represents playback amplitude of
the track audio against the specified Velocity
Zones. TL Drum Rehab translates the detected
amplitudes to MIDI velocity for sample playback. When the detected amplitude of trigger is
in the range of a particular Velocity Zone, the
sample loaded into that Zone is played back
(triggered).
Velocity Zones
(quiet to loud)
detected
Selected
amplitude Velocity Zone
(in dB)
Velocity Zones in the Velocity Map
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The Velocity Map displays the current velocity
(amplitude) on playback. The Velocity Zones are
depicted as colored bars in the Velocity Map.
The different colors from left to right (quiet to
loud) indicate the velocity range: darker colors
represent lower velocity ranges (for example,
1–32) and brighter represent higher velocity
ranges (for example, 95–127). Velocity Zones
trigger samples within the amplitude range of
the Minimum and Maximum Threshold settings
(see “TL Drum Rehab Minimum and Maximum
Threshold Controls” on page 439.
Use the Velocity Map to select a Zone for loading
a sample (see “Loading a Sample into a Zone” on
page 438) and also to adjust the crossfade between Zones (see “Adjusting the Crossfade Between Zones” on page 439). Using multiple Velocity Zones lets you layer samples by dynamics
for more realistic drum sample playback. Use
the left-most Zone for the quietest (pianissimo)
samples, use the right-most for the loudest (fortissimo). Up to four samples (Clips) can be
added to each Zone, to give playback a more human and natural quality. (For more information
on using multiple clips per Zone, see “Clips” on
page 444).
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Audio Plug-Ins Guide
Loading a Sample into a Zone
To load a sample into a Zone:
Click the Zone in the Velocity Map where you
want to load a sample. The selected Zone is indicated by a white triangle.
1
2 In the Library browser (located to right of the
Main window), navigate to the audio file you
want to load (a WAV, AIF, or SD2 file, not a DRP
file).
Double-click the audio file (WAV, AIF, or SD2)
you want to load into the selected Zone.
3
DRP files cannot be loaded into a Zone. DRP
files contain multiple sample with fixed
Zone and Clip assignments. Once you load
samples into Zones and Clips, you can save
them all together as a DRP file.
For a workflow example of loading samples
into Zones, see “Loading Samples and Saving Custom DRP Files in TL Drum Rehab”
on page 448.
In most simple TL Drum Rehab applications,
you may only need to load a single sample into a
single Zone. However, for nuanced and dynamic
sounds, you can use up to 16 Zones for dynamically layered samples.
Adjusting the Crossfade
Between Zones
The Minimum and Maximum Threshold controls also set the amplitude range within which
Velocity Zones trigger samples.
To adjust the crossfade between Zones:
To change the location of the crossfade between Zones, click the border between Zones
and drag it left or right. This determines the
range in which the detected amplitude of the
original track triggers (plays back) the sample
loaded into the Zone.

Adjusting the location of the crossfade between
Velocity Zones
 To change the range of the crossfade between
Zones, click the border between Zones and drag
it up or down. This determines the range of the
crossfade between samples loaded into adjacent
Zones.
Adjusting the range of the crossfade between Velocity
Zones
TL Drum Rehab Minimum and
Maximum Threshold Controls
Adjust the Minimum and Maximum Threshold
controls to determine the minimum and maximum amplitudes for detecting triggers. The
Minimum Threshold control is to the left of the
Velocity Map and the Maximum Threshold control is to the right. The Minimum Threshold
control is useful for filtering out bleed through
hits (like the snare bleed through on a kick
track) so that you only get the triggers you want.
Triggering Controls
Listen Enable the Listen button to “listen” for
triggers in TL Drum Rehab. When Listen is disabled, TL Drum Rehab only plays back Committed triggers (see “TL Drum Rehab Commit Button” on page 441). For most uses of
TL Drum Rehab, Listen is enabled.
No Latency Enable the No Latency button to
play back committed triggers with 0 samples of
latency. No Latency mode ensures sample accurate drum replacement. This is useful when Delay Compensation is disabled in Pro Tools ( Options > Delay Compensation), or for use with
Pro Tools or lower versions of Pro Tools that do
not provide Delay Compensation. When No Latency mode is enabled, only committed triggers
play back and Listen is deactivated.
Playback Controls
The Trigger panel provides global playback controls for input gain (track audio), sample playback gain, ducking, dynamics, and A/B blend.
All playback controls can be automated.
Input Controls the playback gain of the source
track audio. This is like a Dry Mix control. The
range of the Input control is between –40 dB and
+20 dB.
Samples Controls the playback gain of samples
loaded into Velocity Zones. This is like a Wet
Mix control. The range of the Samples control is
between –40 dB and +20 dB.
TL Drum Rehab Main window
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439
Ducking Controls the amount of gain reduction
applied to the input audio when a sample is triggered. This is like a balance control, letting you
adjust exactly how much the track’s audio is
suppressed by the samples triggered by TL
Drum Rehab. The range of the Ducking control
is between –40 dB and 0 dB.
Dynamics Controls the dynamic response of
sample playback and scales the playback velocity of the triggered sample to the detected amplitude of the audio on the track. The range of
the Dynamics control is between 1% and 100%.
When the Dynamics control is all the way to the
left, it is off and samples play back at their original amplitude with no gain scaling. The Dynamics control is especially useful if you are
triggering a single sample or only a few Zones,
but you want more dynamic response on playback than the number of Zones and loaded samples provide.
TL Drum Rehab Expert Panel
Display and Controls
The Expert panel lets you commit, uncommit, or
ignore specific triggers for sample playback, as
well as quantize committed triggers and edit the
location of committed triggers. Playback must
be stopped to commit, uncommit, ignore, or
otherwise edit triggers.
The Expert panel also provides some of the same
controls as the Trigger panel: Listen, No Latency, Minimum and Maximum Threshold, and
the Velocity Map and Velocity Zones.
To access the Expert panel:

Click the Expert Panel button.
A/B Blend Controls the mix between samples
loaded into Positions A and B in the Samples
panel (see “Position A/B” on page 444). For example, Position A could have one center hit
snare sample and position B could have another
center hit snare sample of a slightly different
color. Mixing between the A and B positions
helps give triggered samples a fuller sound by
blending alternate samples.
Expert panel button
Playback Controls
The Expert panel provides the same playback
controls as the Trigger panel: Input, Sample,
Ducking, Dynamics, and A/B Blend. See “Playback Controls” on page 439).
Waveform Display
The Waveform display in the Expert panel is the
same as in the Trigger panel (see “TL
Drum Rehab Waveform Display” on page 435).
It provides a graphic representation of the selected track’s audio, and also displays detected
triggers and velocities (amplitudes). Detected
triggers are displayed as light blue lines on the
waveform. If the Tempo Changes preference is
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Audio Plug-Ins Guide
enabled (see “Tempo Changes” on page 446), the
Waveform display in the Expert panel also
shows Pro Tools Tempo events as green lines
with the tempo indicated at the top of the display.
3
• Click Commit All to commit all detected
triggers.
– or –
• Click Commit and then click only the triggers you want to commit. Committed triggers are indicated by a red arrow.
Tempo events
Waveform display in Expert mode with detected
triggers and amplitudes, and Tempo events
TL Drum Rehab Commit Button
Commit lets you commit specific triggers for
sample playback. Committed triggers play back
regardless of whether or not Listen is enabled. If
Listen is enabled, all detected triggers play back.
If Listen is disabled, only committed triggers
play back. Committing triggers with Listen enabled is useful for making sure that specific triggers are always at the desired location—for example, with sounds that do not have clear attack
transients, you can commit and move the detected trigger to the desired location. Committing triggers with Listen disabled is useful for
playing back only the committed triggers—for
example, when using TL Drum Rehab on a track
with a recording of an entire drum kit, you may
want to only enhance the kick drum sound.
Do one of the following:
Committing specific detected triggers
To play back only committed triggers:
1
Deselect Listen.
2
Start playback.
To edit the position of a committed trigger:
In the Expert or Trigger panels, click and hold
the trigger you want to move. The waveform display zooms to the sample level centered around
the selected trigger.
1
Editing the location of a committed trigger
While still holding down the mouse, move the
trigger left or right until it is at the desired location.
2
To commit detected triggers:
Listen for triggers (see “Triggering Controls”
on page 439).
1
2
3
Release the mouse.
Select the Expert panel.
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441
If you have already selected replacement samples to be triggered, the waveform of the replacement sample is displayed in green over the
track audio waveform (which is white).
To uncommit triggers, do one of the following:
In the Expert panel, click Uncommit All to uncommit all triggers.

– or –
 Click Uncommit and click only the triggers you
want to uncommit.
Uncommit All
Clicking Uncommit All uncommits all detected
triggers in the Timeline selection.
Editing the location of a committed trigger,
replacement sample waveform displayed in green
TL Drum Rehab Ignore Button
To change the amplitude of a committed trigger:
When Listen is enabled, Ignore lets you specify
detected triggers to be ignored during playback.
Triggers do not have to be committed to be ignored.
 Control-click (Windows) or Command-click
(Mac) and drag the trigger left to lower its amplitude or right to increase its amplitude.
Commit All
Clicking Commit All commits all detected triggers in the Timeline selection.
TL Drum Rehab Uncommit
Button
Uncommit lets you uncommit triggers that are
currently committed. This can be useful for simplifying a recorded part (you can uncommit
triggers for a more sparse kick track), and in
cases when the Minimum and Maximum
Threshold controls aren’t able to filter out all
the undesired triggers. For example, if TL Drum
Rehab detects erroneous triggers from bleed
though, such as the floor tom sounding on the
kick track, you can Commit All triggers to be
sure you get all the kick drum hits, and then
manually Uncommit all the triggers generated
by the floor tom.
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Audio Plug-Ins Guide
To ignore specific triggers during playback when
Listen is enabled:
1
In the Expert panel, click Ignore.
2
Click only the triggers you want to ignore.
Triggers that are ignored are marked with a red
X.
Ignoring specific detected triggers
TL Drum Rehab Add Button
Clicking Add analyzes the amplitude of the audio signal at the sample location of the
Pro Tools playback cursor and adds a new trigger with a velocity based on that analysis at that
location. You can use the Add command to add a
trigger during playback or at the current playback cursor location when playback is stopped.
If you have a timeline (playback) selection, the
Add button is unavailable.
While the playback is stopped, use the
Pro Tools Tab To Transients feature to locate the desired trigger location, or zoom to
the sample level to place the cursor at the
precise sample location where you want to
add a trigger.
TL Drum Rehab Undo
If you clicked a trigger that you did not want to
commit, uncommit, or ignore, click Undo in the
Expert panel. TL Drum Rehab supports multiple
undo.
TL Drum Rehab Quantize To
Menu
Use the Quantize To pop-up menu to select the
quantize grid value. The Quantize To pop-up
menu lets you select a quantize grid of 1/2, 1/4,
1/8, 1/16, 1/32, or 1/64 notes.
Selecting Quantize To value
Accurate quantization requires an accurate
Tempo map and Bar|Beat grid. For more information on using the Tempo map and
Bar|Beat grid, see the Pro Tools Reference
Guide.
TL Drum Rehab Quantize
The Quantize slider adjust the amount (from 0%
to 100%) that committed triggers are quantized
to the selected Quantize To value. Quantizing
committed triggers is useful for tightening up a
sloppy performance, as well as an effect to get a
drum machine–like sound.
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443
Samples Panel Display and
Controls
The Samples panel lets you load, view, shape,
and organize samples for playback.
greater degree of realism by adding variety to
the sound. For example, you might want to load
samples of the same drum played with slightly
different stick positions into Clips 1–4 and have
TL Drum Rehab trigger them in random order
for a more realistic sounding “performance.”
To access the Samples panel:

Click the Samples Panel button.
To add a sample to a Clip:
1 In the Samples panel, select the Velocity Zone
to which you want to add a sample.
2 Click the desired Clip: 1, 2, 3, or 4. In order to
select a Clip, there must be a sample already
loaded into the preceding clip.
Samples panel button
Position A/B
3 In the Library browser (located to right of the
Main window), double-click the sample (WAV,
AIF, or SD2) you want to add. TL Drum Rehab
loads the sample into the selected Clip for the selected Zone.
4
The Position A and B button lets you store samples in two different sets of Zones and Clips. The
mix between Positions A and B can be controlled
during playback using the A/B Blend slider in
the Trigger, Expert, or Samples panels. For example, Position A could have a center hit snare
sample and position B could have an off-center
hit snare sample. Mixing between the A and B
positions helps give triggered samples a fuller
sound by blending between alternate samples.
The A/B Blend control can be automated to vary
the mix between Position A and Position B over
time.
Clips
In the Samples panel, TL Drum Rehab lets you
load up to four samples per Velocity Zone using
Clips 1, 2, 3, and 4. Use the Clip Playback Mode
pop-up menu to select whether the Clips are
triggered in sequential order (Cycle) or in random order (Random). Using slightly different
sounds on multiple Clips per Zone adds a
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Audio Plug-Ins Guide
Repeat steps 2–3 as desired.
From the Clip Playback Mode pop-up menu,
select Cycle or Random to determine whether
the clips playback in sequence or in random order.
5
Selecting the Clip Playback mode
DRP Name Display
The DRP Name display displays the name of currently loaded DRP file above the Waveform display in the Samples panel. This is the same as in
the Trigger panel (see “TL Drum Rehab DRP
Name Display” on page 436).
# of Zones
Velocity Map
The # of Zones pop-up menu lets you select the
number of Velocity Zones into which you can
load samples. TL Drum Rehab lets you have up
to 16 Velocity Zones. This is the same as in the
Trigger panel (see “TL Drum Rehab # of Zones”
on page 437).
In the Samples panel, the Velocity Map functions the same as in the Trigger panel (see “TL
Drum Rehab Velocity Map and Velocity Zones”
on page 437).
Play
In the Samples panel, click Play to audition the
currently loaded sample for the selected Zone
and Clip.
Clear
In the Samples panel, click Clear to clear the currently loaded sample for the selected Zone and
Clip.
Velocity Map
In the Samples panel, the Velocity Map functions the same as in the Trigger panel (see “TL
Drum Rehab Velocity Map and Velocity Zones”
on page 437).
Invert
Sample Name display
TL Drum Rehab Samples Panel
Display Waveshaping Controls
Use the envelope and EQ controls to shape the
sound for all clips in the currently selected position (A or B).
Attack Emphasizes or reduces the attack charac-
teristics of all clips in the currently selected position (A or B). The Attack slider has a range of
–100% to +100%.
Sustain Emphasizes or reduces the sustain characteristics of all clips in the currently selected
position (A or B). The Sustain slider has a range
of –100% to +100%.
In the Samples panel, click Invert to invert the
phase of all Clips in the currently selected position (A or B). Invert can be useful for ensuring
phase alignment with other drum tracks in the
session. It can also be used for shaping the tone
of drum sounds—a classic analog technique.
EQ Gain Applies a peaking or dipping EQ to all
clips in the current position (A or B). The EQ
Gain slider has a range of –15 dB to +15 dB.
Sample Name Display
Q Adjusts the Q of the EQ for all clips in the current position (A or B). The Q slider has a range
of 0.1 to 6.0.
The Sample Name display displays the name of
the sample currently loaded into the selected
Zone and Clip is displayed right above the Clear
button.
Freq Adjusts the frequency of the EQ for all clips
in the current position (A or B). The EQ Gain
slider has a range of 10 Hz to 15 kHz.
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445
TL Drum Rehab Preferences
Panel Display and Controls
The Preferences panel lets you set the preferences for TL Drum Rehab. In most cases the default preference settings do not need to be
changed.
Tempo Changes
When the Tempo Changes preference is set to
Show, TL Drum Rehab shows Pro Tools Tempo
events as green lines with the tempo indicated at
the top of the Waveform display in the Expert
panel (see “Waveform Display” on page 440).
This preference is set to Hide by default.
To access the Preferences panel:

Click the Preferences Panel button.
TL Drum Rehab Library
Browser
TL Drum Rehab provides a Library browser for
finding and organizing your library of DRP files
and drum samples. TL Drum Rehab includes a
library of professionally recorded DRP files
(drum samples) tailored specifically for use with
TL Drum Rehab.
Preferences Panel button
Timeline Buffer Size
The Timeline Buffer Size determines the amount
of RAM allocated for the Waveform display. If
you are using TL Drum Rehab on large selections, you may want to increase the Timeline
Buffer Size.
Auto-Scroll Time
When there is no Timeline selection in
Pro Tools, the Auto-Scroll Time preference sets
the amount of time displayed in TL Drum Rehab’s Waveform display during playback. During playback, the Waveform display scrolls incrementally by the amount of time specified in
the Auto-Scroll Time preference.
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Audio Plug-Ins Guide
Library browser
In addition to using the samples that come with
TL Drum Rehab, you can also import your own
samples and save your own custom DRP files
(see “Loading Samples and Saving Custom DRP
Files in TL Drum Rehab” on page 448).
Library
Edit
Click the Library button to view TL Drum Rehab’s Library of DRP files. To navigate through
multiple directories, double-click folders and
use the Up arrow to go up one directory level.
You can also use the disclosure triangles to show
or hide the contents of a folder.
Use the Edit pop-up menu to Add or Remove Favorites, and organize your Favorites in folders.
All of the files available to the TL Drum Rehab
library are stored in the following locations:
Windows <system drive letter>:\Documents and
Settings\<user name>\Application Data
\Trillium Lane\TL Drum Rehab\Samples
Mac /Library/Application Support
/Trillium Lane/TL Drum Rehab/Samples
Favorites
Click the Favorites button to show your favorite
drum samples and folders of drum samples. For
information on Favorites, see “Edit” on
page 447.
Add To Favorites Adds the currently selected
DRP file or folder to the Favorites folder.
Remove From Favorites Removes the currently
selected DRP file or folder from the Favorites
folder.
New Favorites Folder Creates a new folder in the
Favorites folder.
Rename Favorites Folder Lets you rename the
selected Favorites folder.
Auto-Audition
Enable Auto-Audition to hear drum samples in
the Library browser automatically when you
click them. Use the slider to adjust the audition
volume.
File
Use the File pop-up menu to navigate to directories and files, and to save DRP files.
Save New DRP File Saves all audio files cur-
Auto-Audition
rently loaded into Clips and Zones as a new DRP
file.
Help
Save DRP File Saves any edits to the currently
The Help button at the top of the Main window
turns TL Drum Rehab Help Balloons on or off.
loaded DRP file.
Show All Volumes Displays all volumes (drives)
in the Library browser. The Show All Volumes
command retains the last finder view and location.
Refresh All Volumes Searches for newly
mounted volumes (such as sample CDs). It also
clears the most recent finder search location,
and returns the browser to the root level view.
TL Drum Rehab Help Balloons
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447
Loading Samples and Saving
Custom DRP Files in
TL Drum Rehab
To audition a file before importing it, enable
Auto-Audition and click the sample name in the
Library browser.
4
In addition to using the DRP files that come with
TL Drum Rehab, TL Drum Rehab lets you load
your own samples and save custom DRP files.
While you can load samples in both the Trigger
and Expert panels, the Samples panel provides
the most extensive features for loading samples
and saving custom DRP files. The following example describes loading several snare samples
layered by dynamics and then saving them as a
custom DRP file.
Loading samples and saving a custom DRP file:
1
Insert TL Drum Rehab on a mono audio track.
2 In the Library browser, select File > Show All
Volumes. The Library browser displays the root
level of your computer.
Select the Samples panel (see “Samples Panel
Display and Controls” on page 444).
5
Select the desired number of Zones from the #
Of Zones pop-up menu. This example uses 6
Zones for 6 samples of a snare hit all recorded at
different dynamics from p to fff. (See “# of
Zones” on page 445.)
6
Navigate to the directory where the snare samples are located. Double-click a volume or directory to open it in the Library browser, or click
the disclosure triangle to the left of the volume
or directory name to reveal its contents.
3
Select the Zone into which you want to load
the first sample. In this example the samples will
be loaded from soft to loud, so select the leftmost Zone first. (See “TL Drum Rehab Velocity
Map and Velocity Zones” on page 437.)
7
If you want to import samples from a CD,
and you don’t see the CD you may have just
inserted, select File > Refresh All Volumes.
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Audio Plug-Ins Guide
In the Library browser, double-click the desired audio file (WAV, AIF, or SD2) to load it into
the selected Zone.
8
12 Select File > Save New DRP File. The new DRP
file appears highlighted at the top of the browser
list as “Drum Samples.drp.”
9 Repeat steps 7 and 8 for each new sample until
all the samples have been loaded into the corresponding Velocity Zones.
For more variety of sound, you can load
more samples into as many as four Clips per
Zone. (See “Clips” on page 444.)
Click and rename the file to something identifiable. In this example, the samples were recordings of a Noble and Cooley snare, so it is
named “NC Snare 1.”
13
Press Enter (if you do not press Enter, the new
DRP will not be saved). The new DRP file appears in the current directory.
14
Click the Play button to audition the sample
loaded into the currently selected Zone and adjust the Waveshaping controls and other Samples
panel parameters until you get the sound you
want.
10
11 In the Library browser, navigate to the directory where you want to save the loaded samples
as a new DRP file.
TL Drum Rehab provides a User DRPs directory in the Library for storing your custom
DRP files.
Select the new DRP file in the Library
browser and choose Edit > Add To Favorites to
readily access to the new DRP file in the future.
15
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Audio Plug-Ins Guide
Chapter 83: TL Metro
TL Metro is an RTAS metronome plug-in designed to provide you with the convenience of a
traditional metronome, as well as providing advanced functionality for sophisticated timekeeping requirements.
To configure Pro Tools versions 6.9 or earlier for
use with TL Metro:
1
Select MIDI > Click Options.
In the Click Options dialog, ensure that the velocity for the accented note is higher than that of
the unaccented note. By default, they should be
127 and 100 respectively.
2
3
Click OK.
4
Ensure that the MIDI > Click is enabled.
To configure Pro Tools versions 6.1 or earlier for
use with TL Metro, you must also do the following:
TL Metro plug-in
Configuring Pro Tools for Use
with TL Metro
1
Select MIDI > MIDI Beat Clock.
2
Enable MIDI Beat Clock.
3
Select TL Metro as an output.
4
Click OK.
Create a Pro Tools session as a template
with this MIDI setup and use the template as
a basis for future Pro Tools sessions with TL
Metro.
For TL Metro to work in conjunction with the
Pro Tools transport in “linked” mode, it must
receive MIDI from Pro Tools. This is configured
in each Pro Tools session.
To configure Pro Tools versions 7.x or higher for
use with TL Metro:
1
Create a new Pro Tools session.
Create a new audio, Auxiliary Input, or Instrument track.
2
Factory Presets
TL Metro provides a number of factory presets
that provide a range of sounds.
To audition a preset:
3
Insert TL Metro on the new track.
Select the desired preset from the Plug-In Librarian menu.
4
Ensure that Options > Click is enabled.
2
1
Click Play in TL Metro.
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451
TL Metro Controls and
Displays
Volume Sliders
The volume of each individual note can be adjusted using the five Volume sliders. If the volume slider for the accented whole note is reduced to zero, the quarter note will be played
instead of the whole note.
Tempo Controls
Tempo can be specified by manually entering
the tempo, or using the provided slider. Tempo
controls are disabled when TL Metro is linked to
Transport and Tempo.
Tempo controls
Link Status
TL Metro can be linked to the Pro Tools Transport or to the Pro Tools Transport and Tempo
track. For more information, see “Synchronizing TL Metro to Pro Tools” on page 453.
Beats Per Measure Selector
Volume sliders
Sample Selectors
Select the desired audio sample played for each
of the five different notes from the corresponding Sample selector. A sample can be selected
from any of up to 50 sample slots.
Select the number of beats per measure using
the Beats Per Measure selector. If Link Status is
set to Transport+Tempo, TL Metro uses the
Pro Tools session’s Meter track and the Beats
Per Measure selector is unavailable.
Selecting the number of beats per measure
Sample selectors
Master Volume
The Master Volume slider controls the overall
volume of the metronome audio signal.
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Audio Plug-Ins Guide
Sound Library
The Sound Library menu lets you import custom
samples for specific beats. For more information, see “Importing Custom Samples to TL
Metro” on page 454.
Play Button
Using TL Metro and Control
Surfaces
TL Metro parameters can be assigned to a control surface, such as D-Command, Command|8,
Control|24, or Pro Control. The abbreviated
name for each of the beats when displayed on a
control surface as follows.
The Play button activates the metronome. In
linked modes, the Play button is disabled and
the metronome is activated when the Pro Tools
transport is engaged.
• Accented Quarter Note = Beat 1
Tap Button
• Triplet = Beat 5
The Tap button provides a tap tempo function.
Click the tap button in time with the beat to determine the beast. The detected tempo is displayed in the Tempo field and in the LCD display.
TL Metro Information Display
• Quarter Note = Beat 2
• Eighth Note = Beat 3
• Sixteenth Note = Beat 4
Synchronizing TL Metro to
Pro Tools
TL Metro can be synchronized to the Pro Tools
Transport and Tempo using the Link Status selector.
The LCD style information display in TL Metro
displays the following:
• The current tempo in beats per minute
(bpm)
• The current beat of the measure
• Link status
The MIDI name of this instantiation of the
TL Metro plug-in also appears in the display beneath the tempo. This is typically shown as “TL
Metro 1,” “TL Metro 2,” or similar. This enables
multiple instantiations of TL Metro to be easily
identified when routing MIDI.
If a flashing question mark appears in the information display, this indicates TL Metro has encountered an error. For example, MIDI Beat
Clock may not be configured correctly. Click on
the question mark for a dialog window with additional information.
Selecting TL Metro Link Status
Unlinked
When the Link Status is set to None, the TL
Metro can be started and stopped independently
of the Pro Tools Transport and Tempo. This is
useful for recording when you only need the
metronome for a few bars.
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453
Linked to Transport
When the Link Status is set to Transport, the
metronome will start and stop automatically
when the Pro Tools Transport is engaged or disenganged.
When using TL Metro linked to Transport, three
points should be kept in mind:
• Ensure that MIDI is correctly configured
for TL Metro in Pro Tools (see “Configuring
Pro Tools for Use with TL Metro” on
page 451).
• The tempo in TL Metro must be set manually.
• TL Metro assumes you are starting from the
beginning of each bar when you start the
Transport.
Linked to Transport and Tempo
TL Metro can also be linked to both the
Pro Tools Transport and Tempo. In this mode,
TL Metro automatically follows the tempo of the
Pro Tools session in addition to following the
Transport.
Ensure that MIDI is correctly configured for TL
Metro in Pro Tools (see “Configuring Pro Tools
for Use with TL Metro” on page 451).
Customizing TL Metro
Presets
TL Metro provides a selection of factory presets,
including commonly used click sounds. These
presets can be selected from the Plug-In Librarian menu.
User created presets can also be stored using the
Plug-In Settings menu.
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To make any preset the default when TL Metro is
instantiated:
From the Plug-In Librarian menu, select the
desired preset.
1
2
From the Plug-In Settings menu, select Set As
User Default.
From the Plug-In Settings menu, select Settings Preferences > Set Plug-In Default To > User
Setting.
3
For more information on using plug-in presets in Pro Tools, see the Pro Tools Reference Guide .
Importing Custom Samples to
TL Metro
TL Metro supports up to 50 different samples for
metronome click sounds. TL Metro includes factory samples in the first 40 slots, the remaining
slots are marked as “<Unassigned>.”
TL Metro supports import of WAV and AIFF
sound files for specific beat sounds. Sounds can
be loaded into any one of the 50 available slots.
Typically, user samples are loaded into the unassigned slots in order to avoid overwriting the
factory samples. However, any of the 50 slots
can be replaced by user imported samples if desired.
For best results, imported sounds should have
the following characteristics.
• The sound should start in the very first
sample of the file, and have a sharp attack
to ensure proper timing.
• The sample should be normalized before
importing.
• Sound length should be limited to approximately one second to avoid playback problems.
To import a sound:
Click the Sound Library button to display the
sample menu.
1
2
Select an unassigned slot.
3 In the resulting File dialog, select the WAV or
AIFF file you want to import.
4
Click OK.
The name of the selected file is displayed in each
sample menu. To use the imported sample, select it from the sample menu for the appropriate
beat.
Factory and imported samples are stored in a
preferences file named “TL Metro Plug-In” located in your system preferences folder. On
Windows, it’s located in <system drive letter>:\Documents and Settings\<user
name>\Application Data\Trillium Lane\TL
Metro PlugIn.rsr. On Macintosh, it’s located in
Users\<user name>\Library\Preferences\TL
Metro Plug-In.
If you want to use the particular samples you imported into TL Metro on a different Pro Tools
system, copy this preferences file between systems. If the TL Metro preferences file is deleted,
all factory and user samples will be deleted. To
restore TL Metro to the factory samples only,
quit Pro Tools and delete this preferences file.
The next time you use TL Metro, it will recreate
the preferences file with only the factory samples.
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Audio Plug-Ins Guide
Chapter 84: Vacuum
Vacuum is a virtual analog monophonic synthesizer plug-in, with a focus on creating rich timbres
with a lot of sonic control. Employing a new Vacuum Tube Synthesis method, extensive modulation
routing, and a unique age-simulation section, Vacuum invites comparison to classic synths and has a
character all its own. Vacuum is an RTAS plug-in that is part of the Avid Virtual Instrument collection of plug-ins.
Vacuum plug-in window, main controls and sections
Chapter 84: Vacuum
457
Vacuum Controls
Vacuum’s is styled after classic mono synths,
with one control per parameter, and no menus.
By getting a feel for the various sections within
the interface, you’ll soon be creating innovative
new sounds.
Vacuum VTO One and Two
Controls
Vacuum features two VTOs (Vacuum Tube Oscillators). These modules are where Vacuum’s
sound originates from, before it goes through
the rest of the processing chain.
Each VTO has its own set of controls, labelled
“VTO One” and “VTO Two.”
VTO controls
Range Sets the octave at which the current VTO
plays. This is helpful when creating sounds
where the two oscillators must play an octave or
more apart, and also for easily changing the
range a sequence is playing in after the MIDI
note data has already been recorded.
Each Range knob also has a special setting. The
“Wide” setting for VTO 1 changes its Fine knob
into a wide-ranging pitch control that is continuously variable up or down as many as 5 octaves.
The “Lo” setting changes VTO 2 into an LFO
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(low-frequency oscillator). In this mode, its
pitch is too low to be heard, but instead, it can
be routed using the Modulation Routing section
to modulate other parameters in the synth.
Fine Continuously varies the current VTO pitch
up or down as much as 7 semitones. Subtle
changes can create thick, detuned sounds.
Larger amounts can create intervallic splits between the two VTOs, for chordal effects.
Shape Continuously morphs the current VTO
oscillation between several types of wave
shapes.
Wave Shape
Description
Tri
Generates a Triangle wave, with
a mellow, yet slightly edgy sound.
This is the first option for VTO 1
Shape control.
Noise
Generates random white noise.
This is the first option for VTO 2
Shape control.
Saw
Generates a Sawtooth wave,
which is brighter than Tri, and rich
in even harmonics.
PW50–PW0
Generates a Pulse wave, which
can be swept through a continuum between a standard, 50%
on, 50% off wave and a thinner,
more modulated type. Pulse
wave sounds are rich in odd harmonics, with a “reedy” character.
Env 1 to Shape Controls the modulation of the
current VTO wave shape by Envelope 1.
As one of the Env knobs is moved to the right,
more and more modulation occurs, offsetting
the value of the Shape control upward when a
MIDI note is received, then down, following the
envelope over time.
As the control is moved left of center, the same
occurs, only the modulation is negative instead
of positive, so the effect is inverted.
Clicking the missing Drive knob will create a
new patch at random.
Vacuum Mixer Controls
The Vacuum Mixer is where the signals from the
two oscillators are mixed together, their levels
balanced relative to one another. Also, an effect
called Ring Modulation can be added, and Drive
can be applied to the sum of both signals.
Vacuum Filter Controls
Vacuum features two separate filters, one a
high pass filter (HPF), the other a low pass filter
(LPF). The sound of each filter is affected by volume of incoming oscillator signals. Lower mixer
levels give the filters a clean response and a
sharper resonant peak. Increasing mixer gain
can overdrive the filters, adding character and
de-emphasizing resonance.
Each filter has its own set of controls.
=
HPF controls
Mixer section
VTO 1 and VTO 2 Sets the relative volume of the
two oscillators. One (or even both) oscillators
can be reduced to silence, if needed.
Drive Adds a variable amount of distortion to
the mixed signal.
Ringmod Adds a variable amount of the VTO 1
and VTO 2 signals, multiplied together. This is
called Ring Modulation, and can create interesting metallic or abrasive effects.
Cutoff Sets the frequency at which the given filter begins to “cut off” part of the signal’s frequency spectrum. In the HPF, frequencies below
the chosen frequency are affected. In the LPF,
frequencies above the chosen frequency are affected.
Slope Sets the curve of the filter slope. At higher
settings, the slope is steeper, and more of the
spectrum is cut off. At lower Settings, the slope
is more shallow, and more of the spectrum is allowed to pass.
Reso Affects the filter resonance, which is the
amount of signal fed back into the filter circuit
around the chosen frequency. At higher values,
a pronounced peak is created, which can range
from a subtle “edge,” all the way to a sine-wavelike tone. At lower values, the filter simply cuts
off the specified frequencies.
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459
Env 1 Controls the amount that the filter cutoff
frequency is modulated by Envelope 1. At its
center, no modulation occurs.
Vacuum’s modulation envelopes have four main
controls, A (Attack), D (Decay), S (Sustain) and
R (Release).
As the control is moved to the right, more and
more modulation occurs, moving the cutoff frequency up when a MIDI note is received, then
down, following the envelope’s movement over
time.
Example Modulation Envelope
When the control is moved left of center, the
same occurs, only the modulation is negative instead of positive, so the envelope’s effect is inverted.
Key Trk Sets the amount that the currently play-
ing MIDI note’s pitch affects the filter’s cutoff
frequency. At zero, there is no effect. At 100%,
the frequency moves in direct relationship with
the keys played.
This is most apparent with high Res values, as
the tone created may be made to move in tandem, harmonically, with the notes that are
played, thus acting almost as an additional oscillator, with interesting sonic possibilities.
Sat Adds saturation to the resonant feedback
loop, changing the tonal quality of the current
filter from soft to aggressive and distorted.
lope’s modulation to reach its highest point
when a MIDI note is received.
Decay The amount of time it takes for the envelope’s modulation to move from the top of the
Attack phase to the level set by the Sustain control.
Sustain The level at which the envelope stops
while the current MIDI note is held. At zero, the
envelope drops to zero by the end of the decay
period, whether the note is held or not. At 100%,
the envelope holds at its highest point until the
note is released.
Release The amount of time it takes for the envelope’s modulation to drop back to zero after a
note is released. This control has no effect when
sustain is at zero.
Vacuum Envelope Controls
Vel Varies the effect that incoming MIDI note
Vacuum has two modulation envelopes. By default, Env One modulates each filters’ cutoff frequency over time, and Env Two is used to do the
same to the amplitude of Vacuum’s output. The
envelopes can modulate other parameters, as
well. See “Vacuum Modulation Routing Controls” on page 461. for more information.
As the Vel control is moved to the right, more
and more modulation occurs relative to incoming note velocity.
Env One controls
460
Attack The amount of time it takes for the enve-
Audio Plug-Ins Guide
velocity has on the envelope's destination(s) (by
default Filter Cutoff for Env One and overall volume for Env Two). All the way to the left, no
change in modulation occurs.
Vacuum Modulation Routing
Controls
The Modulation Routing section gives you the
ability to go beyond the default modulation
routings, and get deeper into designing sounds.
There are two modulation paths, each with three
controls.
Destination
Description
Pitch
Pitch of both oscillators
VTO 1 Wave
Oscillator 2wave shape
VTO 2 Pitch
Oscillator 2 pitch parameter
VT HPF
High pass filter cutoff frequency
VT LPF
Low pass filter cutoff frequency
Depth Sets the amount of modulation that oc-
curs. At its center, no modulation will occur. To
the right, modulation increases, and to the left,
modulation also increases, but with reversed polarity.
Vacuum Age Controls
Modulation Routing controls
Source Sets what signal is used to modulate the
chosen parameter. The choices are:
Source
Description
Env 1
Envelope 1 modulation signal
Env 2
Envelope 2modulation signal
VTO 1
Oscillator 1 signal
VTO 2
Oscillator 2 signal
LFO
The LFO controlled in the Pitch
and Mod Wheel section’s signal
LFO X MW
The LFO controlled in the Pitch
and Mod Wheel section’s signal,
attenuated by the Mod wheel
MW
The position of the Mod wheel
AT
The amount of Aftertouch, if supported by your MIDI controller
Destination Sets which parameter is modulated.
The choices are:
The Age section lets you explore the tonal effects
of aging internal circuitry and years of dust and
dirt.
Age controls
Drift Adds a variable amount of pitch drift to the
oscillators. At mild settings, the sound is thickened slightly. At more extreme settings, the
sound becomes more detuned and unpredictable, like a poorly-maintained analog synthesizer.
Dust Adds glitches and noise to the signal, emulating the worn and dusty contacts often found
on older synths.
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461
Vacuum VTA Controls
The VTA (Vacuum Tube Amplifier) section acts
as the master volume control for Vacuum, and is
the final place where saturation and distortion
can be introduced to the signal.
Speed Sets the speed of the arpeggio in rhythmic
values that are synchronized to the session
tempo.
Speed
Description
1/4
Quarter notes
1/8
Eighth notes
1/16
Sixteenth notes
1/32
Thirty-second notes
In between each setting, there are two unlabeled
settings for triplet and dotted rhythms. Experiment with different settings until you find the
rhythm you want.
VTA controls
Vol Sets the overall volume.
Mode Sets the direction of the arpeggiator.
Shape Adds a variable amount of tube satura-
Mode
Description
tion to the final output signal.
Up
The arpeggio moves up from the
lowest note held. Once all notes
have been played, the pattern
repeats.
Down
The arpeggio moves down from
the highest note held. Once all
notes have been played, the pattern repeats.
U&D
The arpeggio moves up from the
lowest note held. When the highest note is reached, the arpeggiator runs in reverse, moving down.
Once the lowest note is reached,
the pattern repeats.
RND
The arpeggiator will play through
the held notes, randomly.
Vacuum Arp Controls
The Arp section controls the Arpeggiator: a feature that creates rhythmic arpeggios when one
or more MIDI notes are played and held down.
Arpeggiator controls
On/Off Turns the arpeggiator on and off.
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Audio Plug-Ins Guide
Vacuum Pitch and Mod Wheel
Controls
Pitch and Modulation wheels are the most common controllers on almost any electronic keyboard. The pitch wheel shifts the pitch up or
down a specified amount, for pitch-bending effects.
The modulation wheel is traditionally used as an
expressive tool. In most cases, it controls the
modulation of one or more parameters using an
LFO (low frequency oscillator).
Vacuum Setup Page
Setup button
Click the Setup button to view the Setup page.
The Setup page provides three controls that affect Vacuum’s behavior.
Glide
The Time control sets the amount of slewing (or
Portamento) applied to the pitch of the VTOs.
When set to 0s, the VTOs will play as normal.
When set to a higher setting, the VTOs will take
the number of seconds chosen to glide up or
down to the next note played.
The Mode menu provides the following options:
Off No Glide.
Pitch and Mod Wheel controls
Held Only apply glide when more than one note
Pch and Mod Onscreen wheels move along with
is held at once.
incoming pitch bend and modulation MIDI messages. They can also be clicked and dragged like
other controls.
Dest Sets what parameter is modulated when the
Mod wheel is moved upward.
On apply glide to every note.
Pitch Bend Range
This sets the range of the Pitch wheel, in semitones.
Dest
Description
Envelope Retrigger
Off
No modulation occurs
Vib
Pitch is modulated, creating a
vibrato effect
Wah
LPF cutoff frequency is modulated, creating a wah-wah effect
When set to On, each note played in a legato
phrase will retrigger the actions of Vacuum’s envelopes. When set to Off, legato notes will not retrigger the envelopes until all notes are released
and a new note is struck.
Trem
Overall volume is modulated, creating a tremolo effect
Rate Sets the modulation speed from
0.01–20 Hz.
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463
Inserting Vacuum on a Track
To use one of the Avid Virtual Instruments to its
best advantage, insert it on a stereo Instrument
track in your Pro Tools session.
To insert an instrument plug-in on an Instrument
track:
Create a new stereo Instrument track (recommended) in your Pro Tools session:
1
• Choose Track > New.
• Select 1 new Stereo Instrument track in
Ticks.
• Click Create.
Click the Pro Tools Track Insert selector and
select an Avid Virtual Instruments.
In addition to pre-assigned MIDI controllers
(such as Sustain Pedal and Volume), you can assign MIDI controllers to parameters within
Pro Tools Creative Collection’s instrument
plug-ins for automation or real-time control
from a MIDI keyboard or control surface. MIDI
assignments are saved with the session.
To assign a Pro Tools Creative Collection
Instrument parameter to a MIDI controller:
 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Assign and
choose a controller number from the pop-up
MIDI CC list.
-or-
2
If needed, you can now record-enable the instrument track to enable the use of a MIDI controller to play the instrument and/or help in
creating MIDI sequences within the sequencer
in Pro Tools.
3
See the Pro Tools Reference Guide for instructions on how to use the MIDI sequencer
in Pro Tools.
Using the MIDI Learn
Function on Avid Virtual
Instruments
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Audio Plug-Ins Guide
 Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Learn from
the menu and move the desired knob or controller on your MIDI keyboard or sequencer. The instrument plug-in will set this MIDI controller to
the parameter you have chosen.
Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
MIDI CC
Function
120
All Sound Off
121
Reset Controllers
123
All Notes Off
124
Omni Off (Not used in Plug-Ins)
125
Omni On (Not used in Plug-Ins)
126
Mono On (Not used in Plug-Ins)
127
Mono Off (Not used in Plug-Ins)
To remove a MIDI controller assignment:
 Control-click or Right-click (Mac), or Rightclick (Windows) an assigned control and choose
Forget to remove its MIDI controller assignment .
All Pro Tools Creative Collection plug-ins
have pre-defined parameter assignments for
Avid and supported third-party hardware
control surfaces.
Set Min/Max
These options let you scale incoming MIDI controller data so that the chosen control does not
go below or above a certain value.
This is useful, for example, when you want to
keep the Cutoff control on Vacuum from going
above a certain amount, but you don’t want to
have to pay close attention to how you move the
MIDI controller knob you’ve assigned to it.
To set the Min/Max level:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, choose Set Min or Set
Max, and select the desired lower or upper limit
for the current control.
Invert Range
This option lets you invert incoming MIDI controller data so that the chosen control reacts in
inverse proportion to the assigned MIDI controller.
To invert a control’s response:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, and select Invert
Range.
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Chapter 85: Xpand!2
Xpand! 2 is a virtual workstation synthesizer featuring a broad range of sound generation possibilities
including multi-sampled instruments as well as FM, wavetable, and virtual analog synthesis. Xpand! 2
is an RTAS plug-in that is part of the Avid Virtual Instrument collection of plug-ins.
Xpand!2 plug-in window
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467
Xpand!2 Controls
Getting started with Xpand! 2 is easy, especially
if you are already familiar with virtual instruments or hardware workstations.
Xpand! 2 is multi-timbral. It provides four synthesizer slots, each with individual MIDI channel, Mix, Arpeggiator, Modulation and Effects
settings. A slot can hold one of 1200 synthesizer
presets, called Parts.
The settings of all four slots and their respective
Parts can be saved as a single Patch. Xpand! 2
comes with a set of over 2300 Patches, created by
renowned sound designers. Browse through
these Patches to get an impression of the versatility of Xpand! 2 .
Patch is another name for the plug-in settings. Refer to the Pro Tools Reference Guide
for information on working with RTAS plugins.
The Part selector switches (A, B, C, or D) give
access to the Smart Knob parameters for the selected part.
The Easy button switches the Smart Knobs to
Easy mode. In Easy mode, the Smart Knobs can
address a group of Parts that are all assigned to
a single MIDI channel. Specify the chosen MIDI
channel in the pop-up menu that appears to the
right of the part selectors.
The assigned parameter is displayed in a light
green field below each knob.
Xpand2 Level Control (Master Volume)
The Xpand! 2 Level control affects the master
volume level for the Xpand! 2 plug-in. The level
meter to the right of the level knob shows the
overall output level.
Xpand!2 Global Controls
Xpand!2 Smart Knobs
Level control
The upper section of Xpand! 2 provides 6 controls called Smart Knobs. These are intended for
adapting a preset Part or Patch to your session
in terms of feel, timbre, envelope, and other settings.
Xpand2 Smart Display
The Smart display is a context-sensitive text display. When you select a Patch or Part, it displays
descriptive text about the selected item.
Smart knobs
The Smart Knobs are intelligently pre-assigned
to important parameters by professional sound
designers to make working with Xpand! 2 as easy
as possible.
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Audio Plug-Ins Guide
Smart display
Info Display
MIDI Channel Selector
At the bottom of the Xpand! 2 plug-in window,
an Info display shows the setting of the currently selected control.
To choose the MIDI channel that the current
part responds to, click the MIDI Channel Selector and select the channel from the pop-up
menu.
Info display
Xpand!2 Part Controls
Each Part has a set of controls that address loading patches, the Part’s place in the mix, and its
MIDI channel.
On the right, there is a display that can show
three sets of Patch Edit parameters, including
advanced MIDI settings, Arpeggiator controls,
and Modulation controls.
MIDI Channel selector
For details on how to assign MIDI controllers, see “Using the MIDI Learn Function on
Avid Virtual Instruments” on page 475.
Category Selector
To view Parts organized by categories, click the
Category selector.
On/Off
Activate or deactivate the Part by clicking its
On/Off button. When the Part is activated the
Part character in its center is lit.
Category selector
Part Name
To load a Part into the slot, click the Part Name
field and select a Part from the pop-up menu.
Part On/Off button
Part Selector
Part Name
Click the Part selector to select the Part, so that
its Smart Knobs are displayed.
Level
Part selector
Move the slider to set the Part volume level, increasing volume to the right and decreasing to
the left. The meter above shows the slot’s audio
output.
Level slider
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469
Pan (Panning)
On/Off
Move to the right or left to set the Part’s position
in the stereo field.
Click the FX1 and FX2 buttons to activate or deactivate the effects. The effects are activated
when the buttons are lit.
Pan control
FX1 & FX2
The FX1 and FX2 knobs control the current
Part’s send amount to the effects processors FX1
and FX2.
FX 1 On/Off button
Type
Click the FX type display to select an effect from
the pop-up menu.
F/X controls
Xpand!2 FX Parameter Controls
Xpand! 2 provides two FX (effects) engines. Send
controls for each Part are located on the Mix and
FX pages.
F/X parameter
FX 1 type display
FX1 and FX2 Parameters
Edit the selected effect by adjusting the available
FX parameter knobs. The available parameter
knobs vary depending on the type of effect selected.
FX1 (left) and FX2 (right) parameters
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To FX2 (FX2 Send to FX1)
This control lets you send a percentage of the
FX2 output signal into FX1, instead of directly
to the output. At 0%, no signal is sent to FX1. At
100%, all of the FX2 output signal is sent to FX1.
This is useful for cascading a delay effect into a
reverb for a more ambient effect, for example.
Xpand!2 Play Patch Edit
Controls
The Play controls let you set basic parameters
for the current part, including pitch transposition, keyboard splits, voicing behavior, and
pitch bend range.
Play parameters
FX1 to FX2 knob
Xpand!2
Patch Edit Controls
Overview
The Patch Edit parameter buttons provide access to additional sets of controls where you can
edit the current patch in more detail. Click one
of the three following buttons so that it is lit to
edit its associated parameters:
Button
Controls
Play
Mixer, Panning, FX Sends, MIDI
Arp
Arpeggiator Settings
Mod
Modulation Settings
Tr/Fine
The Tr/Fine (Transpose/Fine Tune) section includes two different controls for transposing incoming MIDI notes. The Semitone control (the
upper control) transposes incoming notes up or
down in semitones. For finer control, use the
Cents control (the lower control), which transposes notes up or down in cents.
Click the control and drag up or down to increase or decrease its value.
Hi/Lo Key
Use the Hi/Lo Key controls to assign Parts to different keyboard ranges. This can be useful for
splitting your keyboard across different Parts.
For example, Part A holding a bass sound could
be assigned C-1 to B2 and Part 2 your synth lead
assigned C3 to G8.
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471
To assign a Part to a certain key range do the
following:
Xpand!2 Arp Controls
 Click the Upper/Lower key range limit control
and drag up or down to increase or decrease its
value.
-or
Do the following:
• Right-click the control and choose Learn.
• Then press the appropriate key on your
MIDI keyboard.
Voice Mode
The Voice Mode section controls the voice behavior of each Part. The Mono/Poly selector (the
upper control) chooses between Monophonic
(one note playable at a time) and Polyphonic
(more than one note playable at a time) modes.
The lower control’s function is different in each
mode. In Mono mode, it selects the key priority
(Last, First, High, Low), which defines which
note is played when more than one note is
played at once. In Poly mode, it selects how
many notes of polyphony are available (1–64).
PB Range
Use the PB Range control to select how many
semitones the given Part can be bent up or down
by pitch bend controller data.
Arp parameters
The arpeggiator automatically triggers the notes
that are played simultaneously in pre-defined
rhythmical patterns. Each Part has its own Arpeggiator.
Some Parts, such as Action Pads and Loops, automatically switch on the Arpeggiator as it
forms an integral part of their sound.
On
Click this button to activate or deactivate the
Arpeggiator. The Arpeggiator will trigger the input notes in the selected pattern as long as the
notes are held. When the Arpeggiator is activated the button is lit.
Latch
Click the Latch button to activate Latch mode
playback. In this mode, the Part’s Arpeggiator
will continue to play after releasing keys until
playback is stopped. Released keys are only removed from the arpeggio when new keys are
pressed. When Latch mode is activated, the button is lit.
When the Arpeggiator is switched on, the
Sustain pedal acts as a temporary Latch
switch, overriding the displayed setting.
Mode
Click the Mode display to select an Arpeggiator
mode from the appearing pop-up menu. An Arpeggiator Mode is a pre-defined rhythmic pattern that the Arpeggiator uses to trigger held
notes.
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Audio Plug-Ins Guide
Rate
Click the Rate display to select the Arpeggiator's
Rate (or speed) from the list. For example “1”
stands for a whole note and “32” stands for a
32nd note. Dotted and triplet timing are indicated by an asterisk (*) or “T” respectively.
dom.” If the pop-up is set to Const the movements of the modulation wheel will directly
modulate the destination without a time varying
waveform.
Xpand!2 Mod Patch Edit
Controls
Mod Wheel Shape button
The Mod (modulation) controls let you easily
create sophisticated modulation settings for
shaping a Part. Modulation wheel and pressure
(aftertouch) can be used as modulation sources.
Click the Destination button (the lower Mod
Wheel button) to select a destination for the
modulation from the pop-up menu.
Mod parameters
Normally, the modulation wheel provides a periodically repeating modulation such as vibrato,
and aftertouch provides a static offset to the selected destination such as volume or filter
swells.
Many Xpand! 2 Patches and Parts have pre-assigned settings for modulation wheel and aftertouch. With the following controls you can
adapt them or create your own.
Mod Wheel Destination
Mod Wheel Destination button
The following destinations for mod wheel action
are provided:
Mod Wheel
Destination
Description
Pitch
Affects the Part’s pitch.
Wave
Changes the sound in different
ways, depending on the
selected Part. For example,
shaping waveforms, FM modulation depth, sample start point
offset, detuning.
Filter
Affects the Part’s filter cutoff frequency.
Volume
Affects the Part’s volume level.
Pan
Affects the Part’s position in the
stereo field.
Xpand!2 Mod Wheel Controls
Mod Wheel Shape
Click the Shape button (the upper Mod Wheel
button) to select the waveform shape for the
modulation from the pop-up menu—an LFO
waveform used to modulate the selected destination. For most waveforms there is a choice of
a freely adjustable and a tempo-synchronized
setting (Sync), except for “Const” and “Ran-
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473
Xpand!2 Pressure Controls
Rate
Move this knob to set the speed or rate of the
modulation wheel’s modulation. When using a
synchronized shape (such as Saw Sync), the Rate
control sets the speed in fixed, tempo synchronized steps. When using other shapes (such as
Sine, Tri, and Saw), the LFO speed is freely adjustable.
Many MIDI keyboards provide pressure (also
called aftertouch) to generate a MIDI control
signal which depends on how hard you press
down held keys after the initial “note on.”
With Xpand! 2 you can use this control signal to
modulate a number of useful controls.
Pressure Destination
Select a destination for the modulation using
pressure (aftertouch) from the pop-up menu.
Mod Wheel Rate knob
Depth
This knob sets the strength or amount of how
much the signal is affected by the modulation.
Depth is a bipolar control, which means that it
can be set to positive or negative values.
Pressure Destination button
The following destinations for pressure action
are provided:
Pressure
Destination
Description
Pitch
Affects the Part’s pitch.
Wave
Changes the sound in different
ways, depending on the
selected Part. For example,
shaping waveforms, FM modulation depth, sample start point
offset, detuning.
Filter
Affects the Part’s filter cutoff frequency.
Volume
Affects the Part’s volume level.
Mod Wheel Depth knob
For example, with the modulation wheel’s
shape set to Const and destination to Pan,
moving the mod wheel up makes the signal
go to the left (negative Depth value) or to the
right (positive Depth value).
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Depth
This knob sets how much the signal is affected
by the pressure control signal. Depth is a bipolar
control, which means that it can be set to positive and negative values.
If needed, you can now record-enable the instrument track to enable the use of a MIDI controller to play the instrument and/or help in
creating MIDI sequences within the sequencer
in Pro Tools.
3
See the Pro Tools Reference Guide for instructions on how to use the MIDI sequencer
in Pro Tools.
Pressure Depth control
Using the MIDI Learn
Function on Avid Virtual
Instruments
For example, with destination set to Filter,
applying aftertouch increases (positive
Depth value) or decreases (negative Depth
value) the filter cutoff frequency.
Inserting Xpand!2 on a Track
To use one of the Avid Virtual Instruments to its
best advantage, insert it on a stereo Instrument
track in your Pro Tools session.
To insert an instrument plug-in on an Instrument
track:
Create a new stereo Instrument track (recommended) in your Pro Tools session:
1
In addition to pre-assigned MIDI controllers
(such as Sustain Pedal and Volume), you can assign MIDI controllers to parameters within
Pro Tools Creative Collection’s instrument
plug-ins for automation or real-time control
from a MIDI keyboard or control surface. MIDI
assignments are saved with the session.
• Choose Track > New.
• Select 1 new Stereo Instrument track in
Ticks.
• Click Create.
Click the Pro Tools Track Insert selector and
select an Avid Virtual Instruments.
2
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475
To assign a Pro Tools Creative Collection
Instrument parameter to a MIDI controller:
Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Assign and
choose a controller number from the pop-up
MIDI CC list.

– or –
Control-click or Right-click (Mac) or Rightclick (Windows) the control, select Learn from
the menu and move the desired knob or controller on your MIDI keyboard or sequencer. The instrument plug-in will set this MIDI controller to
the parameter you have chosen.

Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
MIDI CC
Set Min/Max
These options let you scale incoming MIDI controller data so that the chosen control does not
go below or above a certain value.
To set the Min/Max level:
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, choose Set Min or Set
Max, and select the desired lower or upper limit
for the current control.
Invert Range
This option lets you invert incoming MIDI controller data so that the chosen control reacts in
inverse proportion to the assigned MIDI controller.
Function
To invert a control’s response:
120
All Sound Off
121
Reset Controllers
123
All Notes Off
124
Omni Off (Not used in Plug-Ins)
125
Omni On (Not used in Plug-Ins)
126
Mono On (Not used in Plug-Ins)
127
Mono Off (Not used in Plug-Ins)
To remove a MIDI controller assignment:
 Control-click or Right-click (Mac), or Rightclick (Windows) an assigned control and choose
Forget to remove its MIDI controller assignment .
All Pro Tools Creative Collection plug-ins
have pre-defined parameter assignments for
Avid and supported third-party hardware
control surfaces.
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Audio Plug-Ins Guide
 Control-click or Right-click (Mac) or Rightclick (Windows) a control, and select Invert
Range.
Chapter 86: ReWire
Pro Tools supports ReWire version 2.0 technology developed by Propellerheads Software. ReWire is available in Pro Tools using the ReWire
RTAS plug-in.
ReWire provides real-time audio and MIDI
streaming between applications, with sampleaccurate synchronization and common transport functionality.
Once the outputs of your software synthesizers
and samplers are routed to Pro Tools, you can:
• Process incoming audio signals with plugins
• Automate volume, pan, and plug-in controls
• Bounce To Disk
• Take advantage of the audio outputs of your
Pro Tools audio interfaces
Pro Tools does not support sending audio to
ReWire client applications.
ReWire RTAS plug-in
Using ReWire, Pro Tools can send and receive
MIDI to and from a ReWire client application,
such as a software synthesizer, and receive audio
back from the ReWire client. Pro Tools applies
MIDI time stamping to all incoming MIDI.
Compatible ReWire client applications are automatically detected by Pro Tools and are available in the RTAS Plug-Ins Insert menus in
Pro Tools. Selecting a ReWire client application
within Pro Tools automatically launches that
application (if the client application supports
this feature). Any corresponding MIDI nodes
for that application are available in any Instrument track’s MIDI Output selector (Instrument
view) and any MIDI track’s Output selector.
Not all ReWire client applications support
automatic launch from a ReWire-mixer application. For these applications, launch the
ReWire client app separately, and then select it as a plug-in insert in Pro Tools.
Exchange of additional metadata such as
controller and note names between
Pro Tools and ReWire clients is not supported.
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477
MIDI from Pro Tools to ReWire client (Reason)
Audio from ReWire client (Reason) to Pro Tools
MIDI from ReWire client (Reason) to Pro Tools
Audio and MIDI signal flow between Pro Tools and a ReWire client application (Reason shown)
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ReWire Requirements
To use the ReWire plug-in, you will need:
• An Avid-qualified Pro Tools system
• ReWire-compatible client software (such as
Reason from Propellerheads Software)
Client software must support the same sample rate as the session using ReWire. For example, third-party client software that does
not support sample rates above 48 kHz cannot be used in a 96 kHz Pro Tools session.
ReWire support is also under development for
other third-party companies. For availability,
check with the manufacturer or visit the Avid
website (www.avid.com).
Track Count with Pro Tools HD
With Pro Tools HD, the ReWire RTAS plug-in
can be inserted on any kind of track. Each channel of audio transmitted through ReWire then
uses the same amount of resources as the audio
track on which it is inserted.
Consequently, you can only use a total combination of audio tracks and ReWire audio streams
that does not exceed the maximum number of
possible voices for your system. For example, if
you are playing 40 audio tracks in a 48 kHz/24bit session on a 128-voice Pro Tools|HD 2 system, that will leave 88 channels of audio (88
mono, or 44 stereo) available for use with ReWire. (However, ReWire only supports a maximum of 64 audio streams per application.)
Using ReWire at higher sample rates will increase the load on the CPU. For example, CPU
load at 96 kHz is double the load at 48 kHz. You
can monitor Pro Tools CPU usage in the System
Usage window, making sure to not overtax your
system.
With Pro Tools HD, the standard Hardware
Buffer size of 512 samples is strongly recommended for using ReWire in sessions with
sample rates above 48 kHz.
Track Count with Pro Tools Host-based
Systems
With Pro Tools host-based systems, performance is determined by several factors, including host CPU speed, available memory, and buffer settings. Avid cannot guarantee 64
simultaneous audio channel outputs with ReWire on all computer configurations.
For the latest information on recommended
CPUs and system configurations, visit the Avid
website (www.avid.com).
Using ReWire
The ReWire plug-in is installed when you install
Pro Tools. All inter-application communications between Pro Tools and ReWire client software is handled automatically.
To use a ReWire client application with Pro Tools:
Make sure that the ReWire client application is
installed properly and that you have restarted
your computer.
1
2 In Pro Tools, choose Track > New and specify
one Instrument track (or audio or Auxiliary Input track), and click Create.
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479
3 In the Mix window, click the Insert selector on
the track and assign the ReWire RTAS client
plug-in to the track insert.
The ReWire client application launches automatically in the background (if the client applications supports auto-launch).
If the client application does not support
auto-launch, launch it manually. Some ReWire client applications may need to be
launched and configured before launching
Pro Tools (such as Cycling 74’s MAX/MSP).
Others may need to be launched after
Pro Tools is launched (such as Ableton
Live). For more information, consult the
manufacturer’s documentation for your ReWire client application.
Configure the ReWire client application to
play the sounds you want.
4
5 In Pro Tools, set the output of the client application in the ReWire plug-in window. This is the
audio output of the ReWire client to Pro Tools.
In the Mix window, click the track’s MIDI Output selector a and select the ReWire client application. Some ReWire clients (such as Reason)
may list multiple devices. If so, choose the device that you want.
6
Selecting the ReWire client device to receive MIDI
from Pro Tools (Instrument track shown)
Choose Options > MIDI Thru and record enable
the MIDI track. Play some notes on your MIDI
controller to trigger the client application. The
selected ReWire device responds to MIDI sent
from Pro Tools and plays back audio through the
assigned Pro Tools track (Instrument, Auxiliary
Input, or audio track).
7
If your ReWire client application is a sequencer
and you want to begin synchronized playback
with Pro Tools, press the Spacebar or click the
Play button on the Pro Tools Transport.
Selecting the audio output from a ReWire client
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Audio Plug-Ins Guide
If you experience system performance problems while using Pro Tools with ReWire client applications, you may need to increase
the Pro Tools CPU Usage Limit. See the
Pro Tools Reference Guide for instructions.
MIDI Automation with ReWire
To record MIDI from a ReWire client application in
Pro Tools:
You can use Pro Tools MIDI tracks to record
MIDI continuous controller (CC) data from a
ReWire client application, and then play back
MIDI from Pro Tools to send the recorded MIDI
CC data back to the ReWire client application.
In this way, you can adjust parameters in the ReWire client application (using the mouse or an
external MIDI controller) and record those
changes in Pro Tools.
1
In Pro Tools, create a new MIDI track.
From the track’s MIDI Input selector, select the
ReWire device that you want to record.
2
Recording MIDI Continuous
Controller Data Over ReWire
The first step in automating a ReWire client application’s parameters is to record the CC data
to a MIDI track in Pro Tools.
Selecting the ReWire client device to record MIDI CC
data in Pro Tools
You must select the ReWire device from
which you want to record MIDI controller
data. Leaving the track’s MIDI Input set to
All does not record any MIDI data over ReWire.
3
Record enable the MIDI track.
4
Start recording in Pro Tools.
5
Switch to the ReWire client application.
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481
6 Adjust the parameter for which you want to record MIDI CC data. Parameter changes are recorded to the Pro Tools MIDI track as CC data.
Playing Back MIDI Continuous
Controller Data Over ReWire
Once you have recorded MIDI CC data from the
ReWire client application to a MIDI track, configure the MIDI track to play the ReWire client
application. You can also edit the MIDI CC data
in Pro Tools until you achieve the best results.
To play back MIDI CC data over ReWire:
From the MIDI track’s MIDI Output selector,
select the ReWire client application device you
want to control (the same device from which you
recorded the MIDI CC data).
1
Adjusting a parameter in a ReWire client application
(Reason’s SubTractor shown)
2
If your external MIDI controller is correctly
mapped to the corresponding ReWire client
application’s parameters, and it is correctly
routed through Pro Tools, use your MIDI
controller to adjust the parameter you want
to record.
When you are done adjusting the parameter,
return to Pro Tools and stop recording.
7
8
Record disable the MIDI track.
From the MIDI Track View selector in the Edit
window, select the view for the CC data you just
recorded.
Start playback in Pro Tools.
Switch to the ReWire client application. Notice
that the corresponding parameter changes according the MIDI CC data from Pro Tools.
3
Quitting ReWire Client
Applications
When quitting Pro Tools sessions that integrate
ReWire client applications, quit the client application first, then quit Pro Tools.
9
MIDI CC data recorded from a ReWire client
application
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Audio Plug-Ins Guide
If you quit Pro Tools before quitting ReWire
client applications, a warning dialog may
appear stating that “one or more ReWire
applications did not terminate.” To avoid
this, quit all ReWire client applications before quitting Pro Tools.
Session Tempo and Meter
Changes and ReWire
Looping Playback with
ReWire
Pro Tools transmits both Tempo and Meter data
to ReWire client applications, allowing ReWirecompatible sequencers to follow any tempo and
meter changes in a Pro Tools session.
Because Pro Tools does not offer separate loop
markers as found in other third-party applications such as Reason, if you want to loop playback, do one of the following:
With the Pro Tools Conductor button selected,
Pro Tools always acts as the Tempo master, using the tempo map defined in its Tempo Ruler.
To loop playback in Pro Tools:
With the Pro Tools Conductor button deselected, the ReWire client acts as the Tempo master. In both cases, playback can be started or
stopped in either application.
Pro Tools supports tempo values from
30–300 bpm. When slaved to a ReWire client application, Pro Tools playback will be
restricted to this range even if the client application’s tempo is outside this range. Additionally, some ReWire client applications
(such as Reason) may misinterpret
Pro Tools meter changes, resulting in mismatched locate points and other unexpected
behavior. To prevent this, avoid using meter
changes in Pro Tools when using Reason as
a ReWire client.
In the Pro Tools Timeline, select the time
range that you want to loop.
1
Begin playback by pressing the Spacebar or
clicking the Play button in the Transport.
2
To loop playback within a ReWire client sequencer
 With playback stopped, specify the loop
within the ReWire client application and begin
playback.
If you create a playback loop by making a
selection in the Pro Tools Timeline, once
playback is started, any changes made to
loop or playback markers within the ReWire
client application will deselect the Pro Tools
Timeline selection and remove the loop.
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483
Automating Input Switching
with ReWire
ReWire supports automation for switching inputs during playback.
To automate switching inputs during playback:
1
Set the track’s automation to write.
2
Do one of the following:
• Change the input link pop-up menu manually.
– or –
• Draw the automation in the Edit window.
For information on drawing automation,
see the Pro Tools Reference Guide.
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Part XIII: Other Plug-Ins
Chapter 87: BF Essentials Plug-Ins
BF Essentials is a set of utility plug-ins that are
available in RTAS and AudioSuite formats.
BF Essential Clip Remover
(AudioSuite)
This chapter describes the following plug-ins in
the BF Essential series:
• Clip Remover (see “BF Essential Clip Remover” on page 487)
• Correlation Meter (see “BF Essential Correlation Meter” on page 488)
• Meter Bridge (see “BF Essential Meter
Bridge” on page 488)
The BF Essential Clip Remover repairs clipped
audio recordings. That red light no longer
means a blown take! You’ll be amazed how
quickly this essential tool can repair clipped recordings. Best of all, it’s much quicker and more
accurate than using the Pencil tool. Set your levels very carefully. But when your signal gets too
excited, try the BF Essential Clip Remover.
• Noise Meter (see “BF Essential Noise Meter” on page 488)
BF Essential Clip Remover
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487
BF Essential Correlation
Meter
(RTAS)
Solve tracking and mix problems, and troubleshoot phase coherency with the BF Essential
Correlation Meter. It works on stereo tracks or
stereo submixes. Use it to stop phase problems
before they start.
BF Essential Noise Meter
(RTAS)
The BF Essential Noise Meter is three meters in
one:
• Set to “A,” it’s an A-weighted noise meter
(A-weighting is the most commonly used of
a family of curves relating to the measurement of sound pressure level, as opposed to
actual sound pressure).
• Set to “R-D,” it’s a Robinson-Dadson equalloudness meter (An equal-loudness contour
is a measure of sound pressure, over the frequency spectrum, for which a listener perceives a constant loudness).
BF Essential Correlation Meter
BF Essential Meter Bridge
• Set to “None,” it’s a VU meter with 100 DB
of visual range (Volume Unit metering averages out peaks and troughs of short duration to reflect the overall perceived
loudness).
(RTAS)
The BF Essential Meter Bridge provides best-ofbreed VU metering on any channel while using
minimal DSP resources. Enjoy the ease of use afforded by a needle, a big meter, and a faithful
emulation of the decades-old standard for meter
ballistics. Select RMS or Peak metering, and calibrate instantly for useful viewing at any signal
level, just like a pro tape machine.
BF Essential Meter Bridge
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Audio Plug-Ins Guide
BF Essential Noise Meter
Chapter 88: Signal Generator
Signal Generator is a test tone generator plug-in
that is available in TDM, RTAS, and AudioSuite
formats.
The Signal Generator plug-in produces audio
test tones in a variety of frequencies, waveforms, and amplitudes. It is particularly useful
for generating reference signals with which to
calibrate Pro Tools|HD interfaces and other elements of your studio.
Signal Generator Controls
The Signal Generator plug-in provides the following controls:
Frequency Sets the frequency of the signal in
hertz. Values range from a low of 20 Hz to a high
of 20 kHz in a 44.1 kHz session. The upper limit
of the frequency range for this setting will increase to match the Nyquist frequency (half the
sample rate) in 96 kHz and 192 kHz sessions
(HD-series systems only).
Level Sets the amplitude of the signal in deci-
bels. Values range from a low of –95 dB to a high
of 0.0 dB.
Signal Generator plug-in
Refer to the guide that came with your audio
interface for instructions on using Signal
Generator to calibrate the interface.
The TDM Signal Generator produces a tone
as soon as it is inserted on a track. To mute
the Signal Generator, use the Bypass button.
When using the RTAS version of Signal Generator, start playback to generate.
Signal These buttons select the waveform.
Choices are sine, square, sawtooth, triangle,
white noise, and pink noise.
The Signal Generator plug-in is not intended for rigorous test purposes; it is a simple level calibration tool.
Peak Generates signal at the maximum possible
level without clipping.
RMS Generates signal at levels consistent with
the RMS (Root-Mean-Square) value, or the effective average level of the signal.
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489
AudioSuite Processing with
Signal Generator
To create an audio clip using the Signal Generator
plug-in:
1
Make a selection in the Edit window.
2
Choose AudioSuite > Signal Generator.
3 Enter values for the Frequency, Level, and Signal controls.
4
Click Render in the Signal Generator plug-in.
Select the Create Continuous File option for
greater flexibility in making audio selections for use with the Signal Generator plugin.
You can use the AudioSuite Signal Generator plug-in for musical purposes as well as
for testing purposes. For example, you might
want to add a little color to a kick drum
track by doubling it with a 50 Hz tone, using
the kick track as the key input signal gating
the tone track.
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Audio Plug-Ins Guide
Chapter 89: SoundReplacer
SoundReplacer is an AudioSuite plug-in designed to replace audio elements such as drums,
percussion, and sound effects in Pro Tools
tracks with alternate sounds. SoundReplacer
can quickly and intelligently match the timing
and dynamics of original performance material,
making it ideal for both music and audio post
production.
SoundReplacer features:
• Sound replacement with phase-accurate
peak alignment
• Intelligent tracking of source audio dynamics for matching the feel of the original performance
• Three separate amplitude zones per audio
event for triggering different replacement
samples according to performance dynamics
• Zoomable waveform display for precision
threshold/amplitude zone adjustment
• Crossfading or hard-switching of replacement audio in different amplitude zones for
optimum realism and flexibility
• Online help
SoundReplacer
Audio Replacement
Techniques
Replacing audio elements during the course of a
recording session is a fairly common scenario.
In music production it is often done in order to
replace or augment an element that lacks punch.
In film or video post-production it is typically
done to improve or vary a specific sound cue or
effect.
In the past, engineers and producers had to rely
on sampling audio delay lines or MIDI triggered
audio samplers—methods that had distinct disadvantages. Delay lines, for example, support
only a single replacement sample, and while
they can track the amplitude of the source
events, the replacement sample itself remains
the same at different amplitude levels.
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491
The result is static and unnatural. In addition to
these drawbacks, sample triggers are notoriously difficult to set up for accurate timing.
Similarly, with MIDI triggered samplers, MIDI
timing and event triggering are inconsistent, resulting in problems with phase and frequency
response when the original audio is mixed with
the triggered replacement sounds.
The SoundReplacer Solution
SoundReplacer solves these timing problems by
matching the original timing and dynamics of
the source audio while providing three separate
amplitude zones per audio event. This lets you
trigger different replacement samples according
to performance dynamics.
Each replacement sample is assigned its own adjustable amplitude zone. Variations in amplitude within the performance determine which
sample is triggered at a specific time. For example, you could assign a soft snare hit to a low
trigger threshold, a standard snare to a medium
trigger threshold, and a rim shot snare to trigger
only at the highest trigger threshold.
Replacement samples that are triggered in rapid
succession or in close proximity to each other
will overlap naturally—avoiding the abrupt
sound truncation that occurs on many samplers.
In addition to its usefulness in music projects,
SoundReplacer is also an extremely powerful
tool for sound design and post production. Morphing gun shots, changing door slams, or adding
a Doppler effect can now be accomplished in
seconds rather than minutes—with sample-level
precision.
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Audio Plug-Ins Guide
Replacement audio events can be written to a
new audio track, or mixed and re-written to the
source audio track. Sample thresholds can be
amplitude-switched between the replacement
samples, or amplitude crossfaded for seamless
transitions.
SoundReplacer Controls
SoundReplacer Waveform
Display
The waveform display shows the audio that you
have selected for replacement. When you select
audio on the source track, then open SoundReplacer, the audio waveform will automatically
be displayed here.
Waveform display with trigger markers shown
Once the audio selection is displayed, you can
load the desired replacement samples and adjust
their trigger thresholds while viewing the waveform peaks. Trigger markers then appear in the
waveform, indicating the points at which the
samples will be triggered.
The color of each marker indicates which
threshold/replacement sample will be triggered.
The blue Trigger Envelope shows the waveform
slope that determines the trigger points. The
Zoomer lets you increase or decrease waveform
magnification here to help accurately set trigger
thresholds.
If you change the audio selection on the source
track, click Update to update the waveform display. If Auto Update is selected, SoundReplacer
automatically updates the waveform display
each time you make a new selection or begin
playback.
If you frequently change selections or start
and stop playback, turn off Auto Update to
prevent too-frequent redraws.
SoundReplacerTrigger
Threshold
The color of the Trigger markers correspond to
the matching Threshold slider. This lets you see
at a glance which replacement samples will be
triggered and where they will be triggered.
If you zoom the waveform display below a
specific Trigger Threshold slider’s amplitude zone, the slider will be temporarily unavailable. To access the slider again, zoom
back out to an appropriate magnification
level.
SoundReplacer Load/Unload
Sound Buttons
Load/Unload Sound
Threshold controls
The color-coded Trigger Threshold sliders set a
total of three amplitude zones (one for each replacement audio file) for triggering replacement
samples:
• The yellow slider represents amplitude zone 1,
the lowest-level trigger.
• The red slider represents amplitude zone 2,
the middle-level trigger.
• The blue slider represents amplitude zone 3,
the highest-level trigger.
Clicking the Load/Unload Sound icons loads or
unloads replacement samples for each of the
three trigger threshold amplitude zones. Clicking the Floppy Disk icon loads a new sample (or
replaces the current sample). Clicking the Trash
Can icon unloads the current sample.
SoundReplacer does not perform a sample
rate conversion before loading replacement
samples if they are at a different sample rate
from the session. Replacement samples
should be at the same sample rate as the session, otherwise they will playback at the
wrong speed and pitch.
With a replacement sample loaded, drag the
Threshold slider to the desired amplitude level.
Color-coded trigger markers will appear in the
Waveform at points where the source audio signal exceeds the threshold set for that amplitude
zone. The replacement sample will be triggered
at these points.
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493
To audition a replacement sample before loading it into SoundReplacer, use the Import Audio
command in Pro Tools. Once you have located
and previewed the desired audio file, you can
then load it into SoundReplacer using the
Load/Unload Sound icons.
SoundReplacer does not load clips that are
part of larger audio files. To use a clip as a
replacement sample, you must first save it
as an individual audio file.
SoundReplacer Zoomer
SoundReplacer Crossfade
When Crossfade is selected, SoundReplacer
crossfades between replacement audio files in
different amplitude zones. This helps smooth
the transition between them.
When Crossfade is deselected, SoundReplacer
hard switches between replacement audio files
in different amplitude zones.
Crossfading is particularly useful for adding a
sense of realism to drum replacement. Crossfading between a straight snare hit and a rim shot,
for example, results in a much more “live” feel
than simply hard switching between the two
samples.
SoundReplacer Peak Align
Zoomer
The Zoomer increases or decreases magnification of the waveform data currently visible in
the center of the waveform display so that you
can more accurately set sample trigger thresholds.
• To zoom in on amplitude, click the Up Arrow.
• To zoom out on amplitude, click the
Down Arrow.
• To zoom in on time, click the Right Arrow.
When Peak Align is on, SoundReplacer aligns
the peak of the replacement file with the peak of
the source file in a way that best maintains
phase coherency. When Peak Align is off,
SoundReplacer aligns the beginning of the replacement file with the trigger threshold point.
Depending on the characteristics of your source
and replacement audio files, using Peak Align
can significantly affect the timing of audio
events in the replacement file. It is essential that
you choose the option most appropriate to the
material that you are replacing.
• To zoom out on time, click the Left Arrow.
If you zoom the waveform display below a
specific Threshold slider’s amplitude zone,
the slider will be temporarily unavailable.
To access the slider again, zoom back out to
an appropriate magnification level.
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For more information on using Peak
Align, see “Getting Optimum Results with
SoundReplacer” on page 497.
SoundReplacer Update
SoundReplacer Dynamics
When you click Update, the waveform display is
redrawn, based on the audio currently selected
on the source track. Each time you make a new
selection on a source track, you must click Update for SoundReplacer to draw the waveform of
the selection.
Dynamics controls how closely the audio events
in the replacement file track the dynamics of the
source file:
SoundReplacer Auto Update
When Auto Update is selected, SoundReplacer
automatically updates the waveform display
each time you make a new selection on a source
track. If you frequently change selections or
start and stop playback, you may want to deselect Auto Update to prevent frequent redraws.
SoundReplacer Mix
Mix adjusts the mix of the replacement audio
file with the original source file. Higher percentage values weight the mix toward the replacement audio. Lower percentage values weight the
mix toward the original source audio.
The Mix button toggles the Mix control on and
off. When Mix is toggled off, the balance is instantly set to 100% replacement audio.
Setting Mix to 50% and clicking Preview lets
you audition source audio and replacement
audio together to check the accuracy of replacement triggering timing.
 Setting the ratio to 1.00 matches the dynamics
of the source file.
 Increasing the ratio above 1.00 expands the
dynamic range so that softer hits are softer, and
louder hits are louder. This is useful if the source
material lacks variation in its dynamic range.
 Decreasing the ratio below 1.00 compresses
the dynamic range so that there is less variation
between loud and soft hits. This is useful if the
dynamics of the source material are too extreme.
The Dynamics button provides a quick means of
toggling on and off the Dynamics control. When
Dynamics is toggled off, SoundReplacer will not
track changes in the source audio file’s dynamics. Audio events in the resulting replacement
audio file will uniformly be at the amplitude of
the replacement samples themselves, with no
variation in dynamics.
SoundReplacer Online Help
Online help
To use online help, click the name of any control
or parameter and an explanation will appear.
Clicking the Online Help button provides further details on using this feature.
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Using SoundReplacer
Following are basic guidelines for using
SoundReplacer effectively. Also see “Getting
Optimum Results with SoundReplacer” on
page 497.
To use SoundReplacer:
On the source track, select the audio you want
to replace. Only selected audio will be replaced.
1
11 Adjust the Mix slider to get the desired balance between replacement audio and source audio.
Adjust the AudioSuite File controls. These
settings will determine how the file is rendered
and what effect the rendering will have on the
original clips. Here are some guidelines:
12
 Decide where the selected clip should be rendered:
• To render the selected clip only in the track
in which it appears, choose Playlist from the
Selection Reference pop-up.
Choose SoundReplacer from the AudioSuite
menu.
2
Click the Load Sound icon (the icon beneath
the yellow slider) to import the replacement
sound for amplitude zone 1.
– or –
3
4
Locate the desired audio file and click Open.
5
Adjust the amplitude zone slider.
• To render the selected clip in the Audio Clip
List only, choose Clip List from this pop-up.
 Decide if you want to update every occurrence
of the selection clip:
• To render and update every occurrence of
the selected clip throughout your session,
enable Use In Playlist (and also choose Clip
List from the Selection Reference pop-up).
Repeat steps 3–5 to load replacement sounds
into amplitude zones 2 and 3.
6
If you use only a single replacement sample,
you should still set all three amplitude zones
for optimum results. This will ensure accurate triggering. For details, See “Mapping
The Same Sample Into Multiple Amplitude
Zones with SoundReplacer” on page 498.
To align the amplitude peak in the replacement
file(s) to threshold trigger markers in the source
audio, enable Peak Align.
7
8 Click Preview to audition the replacement audio.
– or –
• If you do not want to update every occurrence of the selected clip, disable Use In
Playlist.
 If you have selected multiple clips for rendering and want to create a new file that connects
and consolidates all of these clips together,
choose Create Continuous File from the File
mode pop-up menu.
Because SoundReplacer does not allow destructive rendering, the AudioSuite Overwrite Files option is not available.
9 Adjust the Threshold sliders to fine tune audio
replacement triggering.
Adjust the Dynamics slider to fine tune how
SoundReplacer tracks and matches changes in
the source audio’s dynamics.
10
From the Destination Track pop-up, choose
the destination for the replacement audio.
13
14
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Audio Plug-Ins Guide
Click Render.
Getting Optimum Results
with SoundReplacer
To illustrate why Peak Align makes a difference,
look at the following two illustrations.
Getting optimum results with SoundReplacer
generally means making sure that the audio
events in the replacement audio file have accurate timing in relation to the source audio. The
techniques given here help ensure this.
Using Peak Align in
SoundReplacer
A fast-peaking kick drum
Proper use of the Peak Align feature can significantly improve the results of sound replacement. Since turning Peak Align on or off controls how SoundReplacer aligns the replacement
audio with the source audio, it will significantly
affect the timing of audio events in the replacement file.
A slower-peaking kick drum
In general:
Turn on Peak Align if you are replacing drum
or percussion sounds whose peak level occurs at
the initial attack.

 Turn off Peak Align if you are replacing
sounds whose peak level occurs somewhere after
the initial attack. Peak Align should also be
turned off if the sounds you are replacing are not
drum or percussion sounds.
The first figure shows a fast-peaking kick drum
whose peak level occurs at its initial attack.
The second figure shows a slower-peaking kick
drum whose peak level occurs after its initial attack.
If you turn on Peak Align and attempt to replace
the fast-peaking kick with the slow-peaking kick
(or vice-versa), SoundReplacer will align their
peaks—which occur at different points in the
sound. The audible result would be that the replacement audio file (slow-peaking kick) would
trigger too early.
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Mapping The Same Sample Into
Multiple Amplitude Zones with
SoundReplacer
If you are performing drum replacement and intend to use just a single replacement sample,
mapping it into multiple amplitude zones will
ensure more accurate triggering. Here is why:
If you use a single amplitude threshold to trigger
the replacement sample, you have to set the
threshold low enough to trigger at the soft hits.
The problem occurs at the loud hits: The threshold is now set so low that the pre-hit portion of
the loud hits can exceed the threshold—triggering the replacement sample too early. This results in a replacement track with faulty timing.
Imagine that you are replacing a kick drum part.
If you look at the waveform of a kick drum, you
will often see a “pre-hit” portion of the sound
that occurs as soon as the ball of the kick pedal
hits the drum. This is rapidly followed by the
denser attack portion of the sound, where most
of sound’s weight is.
A single low threshold causes the second, louder kick
to trigger too early, as evidenced by the trigger marker
at the very start of the waveform.
A kick drum with a pre-hit preceding a denser attack
The best way to avoid this problem is to set multiple threshold zones for the same sample using
a higher threshold for the louder hit. Soft hits
will trigger threshold 1 and louder hits will trigger threshold 2.
With a sound like this, using a single amplitude
threshold presents a problem because typically,
in pop music, kick drum parts consist of loud accent hits and softer off-beat hits that are often
6 dB or more lower in level.
Using a second, higher threshold for the louder kick
will make it trigger properly, as shown by the now
properly-aligned trigger marker.
To set the precise threshold for louder hits, you
may need to zoom in carefully to examine the
waveform for trigger points (indicated by colorcoded trigger markers) and then Commanddrag the Threshold slider for more precise adjustment.
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Audio Plug-Ins Guide
If there is a great deal of variation in the dynamics of the source audio, you may need to use all
three Trigger Thresholds/Amplitude Zones for
optimum results.
If only one replacement sample is loaded
into SoundReplacer and it is loaded into
Trigger threshold/amplitude zone 1 (yellow), SoundReplacer will let you use the red
and blue Trigger Threshold sliders to set
Amplitude Zones 2 and 3—without having
to load the same sample again.
Using the Audio Files Folder
for Frequently Used
SoundReplacer Files
If it is not there, SoundReplacer looks in a folder
named Audio Files within SoundReplacer’s Root
Plug-In Settings folder (Plug-In Settings/SoundReplacer/Audio Files).
If SoundReplacer finds the replacement audio
file there, the Settings file will load with the associated audio.
By always putting replacement audio files in this
special folder, you can freely exchange SoundReplacer settings—and the audio files associated with them—with other users.
Do not create subfolders within SoundReplacer’s Audio Files folder. Files located
within subfolders are not recognized.
If you often use the same settings and replacement sounds in different sessions, SoundReplacer provides a convenient way to keep the
replacement audio files and settings linked together.
When you choose a preset from the Plug-In Librarian menu, SoundReplacer looks for the replacement audio files associated with that preset. Sound-Replacer first looks in the audio file’s
original hard disk location (at the time you
saved the setting).
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Audio Plug-Ins Guide
Chapter 90: Time
Compression/Expansion
Time Compression/Expansion is an AudioSuite
time-processing plug-in.
The Time Compression/Expansion plug-in adjusts the duration of selected clips, increasing or
decreasing their length without changing pitch.
Time Compression/
Expansion Controls
The Time Compression/Expansion plug-in provides the following controls:
Source and Destination The Source fields dis-
play the length of the current selection before
processing in each of the listed formats. All
fields are always active; a change made to one
value is immediately reflected in the others.
The Destination fields both display and control
the final length of the selection after processing.
Enter the length of the Destination file by double-clicking the appropriate field in the Destination column.
Use the Ratio, Crossfade, Min Pitch, and Accuracy controls to fine-tune the Time Compression/Expansion process.
Time Compression/Expansion plug-in
It is especially useful in audio post production
for adjusting audio to specific time or SMPTE
durations for synchronization purposes. Time
Compression/Expansion is nondestructive.
Normalizing a selection before using Time
Compressing/Expansion may produce better
results.
Ratio Sets the destination length in relation to
the source length. Moving the slider to the right
increases the length of the destination file, while
moving the slider to the left decreases its length.
Crossfade Adjusts the crossfade length in milli-
seconds, optimizing performance of the Time
Compression/Expansion according to the type
of audio material being processed. (This plug-in
achieves length modification by replicating or
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501
subtracting very small portions of audio material and very quickly crossfading between these
alterations in the waveform of the audio material.)
Crossfade length affects the amount of smoothing performed on audio material. This prevents
audio artifacts such as clicks from occurring.
Long crossfade times may over-smooth a signal
and its transients. This may not be desirable on
drums and other material with sharp transients.
Use the Crossfade slider to manually adjust and
optimize crossfade times if necessary. For audio
material with sharper attack transients, use
smaller crossfade times. For audio material with
softer attack transients, use longer crossfade
times.
Min Pitch Sets the minimum, or lowest, pitch
that will be used in the plug-in’s calculations
during the Time Compression/Expansion process. The control has a range of 40 Hz to 1000
Hz.
This control should be set lower when processing bass guitar or audio material with a low frequency range. Set this control higher when processing higher frequency range audio material.
Accuracy Prioritizes the processing resources
allocated to audio quality (Sound) or timing
(Rhythm). Moving the slider towards “Sound”
generally results in better sonic quality and
fewer audio artifacts. Moving the slider towards
“Rhythm” puts the emphasis on keeping the
tempo consistent.
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Audio Plug-Ins Guide
When you are working with audio loops, listen
carefully and adjust the Accuracy slider until
you find a setting that keeps timing solid within
the clip. If you don’t, start and end times may be
precise, but the beats in rhythmic material may
appear to be shuffled if too little priority is given
to Rhythm.
Time Compression & Expansion settings
created in version 4.x and later of Pro Tools
for Windows are not compatible with later
versions.
Chapter 91: TL InTune
TL InTune is a professional instrument tuner
plug-in that is available in TDM and RTAS formats. It offers the features and performance of a
rack mounted digital tuner in the convenience
of a plug-in. TL InTune provides accurate and
rapid tuning for a wide range of musical instruments, saving valuable studio time and adding a
level of unprecedented convenience for musicians and audio engineers.
When TL InTune detects an audio signal from
the track, the meter lights up and displays the
relative pitch of the incoming signal. With
stringed instruments, this will vary during the
attack and decay of the note.
To use TL InTune with Pro Tools, simply create
a new mono audio or Auxiliary Input track in
Pro Tools, and select TL InTune from the plugin menu for that track.
TL InTune provides a number of factory presets
for stringed instruments in alternate tunings.
Each factory preset is programmed with the specific notes for each string of the instrument in
order to speed the tuning process, as well as
making it easier for engineers to generate test
tones for musicians to tune with.
By default, TL InTune loads the Chromatic tuner
preset. This displays all notes in the scale and
automatically displays the required octave.
TL InTune plug-in
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503
TL InTune Controls and
Displays
To hear a test tone:
1 Select Sine, Triangle, or Audible from the Test
Tone selector.
TL InTune Auto Button
Click the Auto button to toggle Automatic Mode
on and off. When Automatic mode is active, TL
InTune will detect the note played and automatically show the pitch for that note.
To enable Automatic mode:
Selecting a test tone
2
Click the Note button for the desired note.
 Click the Auto button to enable Automatic
mode. The Auto button highlights.
To tune to a single note and turn off Automatic
mode:

Click the button for the desired note.
Selecting a test tone note
3
Adjust the Tone Volume slider as desired.
When a test tone is playing, “Tone Playing” appears in the information display.
Selecting a note
This turns off automatic mode. TL InTune will
now display pitch relative to the selected note
only.
TL InTune Test Tone Menu
Selector
TL InTune will generate both sine wave and triangle wave test tones as shown in the tone menu.
The “Audible” tuning tone modulates the input
signal against the reference tone.
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Audio Plug-Ins Guide
TL InTune Edit Button
Clicking the Edit button displays the Tuner Programming screen, where you can create customized tuning presets that display note selections
for specific instruments and tunings. See “Creating TL InTune Tuning Presets” on page 507.
TL InTune Meter Selector
The Meter selector lets you use a standard needle style meter or a strobe style display.
TL InTune Reference Frequency
Control
To select the Meter display:
Select Needle or Strobe from the Meter selector.

TL InTune, Reference Frequency
You can adjust the tuning reference frequency
using the arrows inside the information display.
By default, reference frequency is A=440 Hertz.
TL InTune Note Buttons
The Note buttons provide two functions:
Selecting Meter display, Strobe
Strobe Display
• When in automatic mode, clicking on a
note button will turn off automatic mode
and TL InTune will now display pitch relative to the selected note only.
• When a tone is selected in the test tone
menu, clicking on a note button will play a
test tone for that note. Click the note button
again to turn off the test tone.
TL InTune, Strobe display
The number of note buttons will depend on the
preset selected. The default chromatic preset
will display all twelve notes. A preset for a six
string guitar will only display six notes.
The Strobe display scrolls to the left when the
tuned note is flat, and to the right when the
tuned note is sharp. When the tuned note is
close to the target note, the strobe slows to a
stop. The information display shows the exact
number of cents sharp or flat from the target
note.
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505
Octave Buttons
TL InTune Presets
TL InTune provides a selection of factory presets for stringed instruments. These presets can
be selected from the Plug-In Librarian menu.
Down Octave button
Up Octave button
Octave buttons
The octave range of 0–6 displayed in TL InTune
is based on middle C being equal to C4. In chromatic presets, you can select the desired tuning
octave from 0–6 using the arrows at each end of
the note display.
Selecting a TL InTune preset
TL InTune Tone Volume
To make any preset the default when TL InTune is
instantiated:
The Tone Volume slider controls the volume of
the test tone audio signal.
1
TL InTune Information Display
2
The LCD style information display in TL InTune
displays the following:
3 From the Plug-In Settings menu, select Settings Preferences > Set Plug-In Default To > User
Setting.
• The reference frequency
• The current note to which TL InTune is tuning
• The number of cents sharp or flat from the
current note
• The status of any test tones playing
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Audio Plug-Ins Guide
From the Plug-In Librarian menu, select the
desired preset.
From the Plug-In Settings menu, select Set As
User Default.
For more information on using plug-in presets in Pro Tools, see the Pro Tools Reference Guide .
Creating TL InTune Tuning
Presets
TL InTune lets you create customized tuning
presets that display note selections for specific
instruments and tunings. Once created, these
tuning presets can be saved as part of a standard
Pro Tools plug-in preset.
From the main TL InTune screen, click the Edit
button to display the Tuner Programming
screen.
Single Octave Mode is typically used for instruments which generate harmonics in multiple octaves, such as bass guitars. Because of the low
frequency waveform generated by a bass guitar,
it is easier for TL InTune to tune to a higher harmonic of the note instead.
Display Flat Semitones
TL InTune will display all semitones entered
into note fields as sharp by default. For example,
a guitar tuned to E-flat is usually represented by
the following.
Eb2, Ab2, Db3, Gb3, Bb3, Eb4
By default, if these notes are entered in the Edit
screen, TL InTune will display these same notes
in the following way.
D#2, G#2, C#3, F#3, A#3, D#4
Tuner Programming
Chromatic Mode
When selected, Chromatic Mode overrides any
custom note selections and displays a 12-note
chromatic scale. The note entry fields are disabled when Chromatic Mode is selected.
Single Octave Mode
When selected, Single Octave Mode disables the
display of octave information with each note on
the main TL InTune screen. When tuning in this
mode, TL InTune ignores the octave of the note
being tuned. The octave information entered in
the Edit screen is used only for generating test
tones.
The Display Flat Semitones option overrides the
default behavior and displays semitones as flats,
not sharps. It is not possible to display both
sharp and flat semitones in the same tuning preset.
Note Entry Fields
The twelve note entry fields allow entry of individual notes from A0 to G7. Flat semitones are
entered with a “b” (for example, Ab2), and sharp
semitones are entered with a hash or pound
character (for example, A#2). To clear an entry,
enter “– –.”
Note fields are committed by pressing Return
(Macintosh) or Enter (Windows). If you do not
press Return or Enter, the note field will return
to the previous value entered. TL InTune will
automatically justify the note buttons as needed
so they fit in the correct area on the main screen.
The Note Entry fields are not available in Chromatic mode.
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507
Exit
In the Tuner Programming screen, click the Exit
button to return to the main TL InTune screen.
Using TL InTune
When TL InTune detects a signal, the meter
lights up and displays the relative pitch of the
incoming signal. With stringed instruments,
this will vary during the attack and decay of the
note.
In Automatic mode, TL InTune estimates the
note to which you are trying to tune. If the correct note is not lit in automatic mode, click on
the note to which you are trying to tune for
greater accuracy. This will lock TL InTune to the
specified note.
The meter will display the frequency of the note
detected, and the accuracy is displayed on a
scale of plus/minus 50 cents. In addition, the information display will display the note and the
number of cents from perfect tuning.
When loading factory presets, stringed instruments are laid out from the highest numbered
string (usually the lowest tone) to the highest,
from left to right. For example, a six string guitar in standard tuning is shown as E2, A2, D3,
G3, B3, E4, which are the notes and octaves for
the sixth string through to the first string respectively.
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Audio Plug-Ins Guide
For best tuning results with guitars, do the following:
• Use headphones, as loud monitors can
modulate the guitar string.
• Switch your guitar to its rhythm (neck)
pickup, if it has one.
• Roll your guitar’s tone knobs all the way off
to remove all the highs.
• Pluck the open string right over the twelfth
fret, not over the pickup.
To produce convenient test tones, select the appropriate preset from the Librarian menu and
select an appropriate test tone from the Test
Tone menu. Click on the desired Note button to
produce the appropriate test tone. Test tones
can be routed to headphones as required for musicians during session.
Chapter 92: TL MasterMeter
TL MasterMeter is an oversampling meter plug-in that is designed for critical mixing and mastering
applications. TL MasterMeter is available in TDM and RTAS formats.
TL MasterMeter plug-in
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509
TL Master Meter Overview
This section provides an overview of metering
and mastering, and how TL MasterMeter can
help you produce great sounding mixes.
Understanding Digital Distortion
Clients in the music industry regularly demand
the loudest possible mixes. In the process of
achieving such a “hot mix,” unwanted distortion
can be introduced. Intersample peaks that exceed 0 dB may play without distortion in a studio environment, but when the same mix is
played through a consumer CD player, the digital to analog conversion and oversampling process can reproduce a distorted mix.
A complex waveform
Waveform sampled
Digital Audio Theory
A key observation in digital audio theory is that
the entire waveform is represented by the sampling points, but a reconstruction process still
needs to occur in order to recreate the waveform
represented. One cannot simply “connect the
dots” between sample points and yield the original waveform.
Sampling
Waveform as reconstructed at the D/A
A waveform can be represented in multiple ways
during the process of sampling, display and reconstruction.
The process of recreating the original waveform
from the sampled waveform involves a filter
called a reconstruction filter. This filter removes all content above the Nyquist frequency
(half the sample rate). The range below the Nyquist frequency defines the “legal” range of allowed frequencies as frequencies in this range
The following four figures show how the same
complex waveform shown in the previous figure
can be represented in the digital domain.
510
Waveform as represented in DAW
Audio Plug-Ins Guide
can be accurately reproduced. All frequencies
above the Nyquist frequency do not adhere to
Nyquist or Shannon’s theorems regarding allowable frequencies, cannot be reproduced and
are therefore considered “illegal” frequencies.
Because of mathematical realities observed by
Fourier in the 1800’s and subsequently by Shannon in 1948, when a waveform has all frequencies removed above the Nyquist frequency, the
resulting waveform will be the original waveform that was sampled.
This process is significantly more involved than
simply “connecting the dots” between sample
points. Today it involves extremely sophisticated means of reconstructing the waveform,
using filters that are highly complex mathematical systems utilizing “oversampling,” “upsampling,” “linear phase, equiripple FIR” designs
and much more.
Oversampling creates a more accurate digital
representation of an analog signal by sampling
some number of times per second (frequency)
and converting into digital form. Oversampling
requires at least twice the bandwidth of the frequency being sampled. For example, a consumer
CD player using 2x oversampling is processing
information at 88.2 kHz.
The result is that today’s digital to analog converters get closer to the original than ever before, making music played on systems today as
accurate as possible. Even today’s inexpensive
components such as off-the-shelf CD players
have drastically improved filters and thus better
reconstruction abilities than in years past.
Application
Most contemporary audio recording is done
with Digital Audio Workstations (DAWs), although digital mixing systems in the form of
outboard digital mixers are also very popular.
To the user, these digital systems appear similar
to traditional audio tools and are designed order
to emulate the operation of a conventional analog recording system.
One familiar analog tool that has been carried
over to the digital realm is a “peak meter” that
tells the amplitude of the waveform’s peaks. In
the analog realm, peak signal was an indicator
that would alert the audio engineer when the
peak signal level was getting too high. A peak
signal in analog recording would cause the tape
to saturate, creating distortion. In an analog
system however, this type of distortion was often deliberately engineered into tracks in order
to achieve a certain sound.
In the digital realm this type of meter is important and more vital, because if the amplitude of
a waveform exceeds the top of the measurable
scale (full scale, or “full code”), the signal will
“clip” causing unwanted and unpleasant distortion rather than the traditional distorted sound
of analog. This digital clipping occurs because
the waveform is “lopped off” and the data is
changed. When the waveform is reconstructed it
cannot be accurately done in order to represent
the original waveform. Instead, it has a significant amount of inharmonic distortion caused by
aliasing. For this reason, digital recording has a
maximum level at which signals can be recorded. Anything exceeding this level (full
scale) has undesirable consequences.
The method used for computing the peak value
inside the system however is not particularly accurate. DAW systems typically take the amplitude of the samples and use these as the basis for
the peak meter. The problem with this approach
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511
is easily identified: the samples themselves do
not represent the peak value of the waveform.
The waveform is only complete after the reconstruction process. Until this process has been
completed, the waveform is inaccurately represented by the samples. This is the reason that in
most DAWs the waveform is represented on the
screen as a “dot to dot” connection between
sample points. They do not undergo the reconstruction process inside the system, so all that
can be represented is the sample points and for
the sake of visual ease, they connect the dots between them with straight lines. They save the reconstruction process for the digital to analog
converters.
Intersample peaks
The consequence of the way in which DAWs
treat waveforms is that the meter inside the
DAW or other digital mixers inevitably shows
inaccurate information. It is virtually a mathematical certainty that the waveform will exceed
the amplitude of the samples in any sampling
system. The samples themselves only represent
a waveform. It is important to understand that
the amplitude of the waveform will invariably
exceed the sample values.
pressed even more so with multi-band compressors, limited, normalized, and maximized to get
the audio to play as loud as possible out of a consumer’s system. Hence, it is very common for
popular music CDs to be full of digital samples
that are at, or nearly at full scale.
The problem is realized in that while going
through these digital gyrations and utilizing
digital tools to amplify the signal as much as
possible, both during mixing and during mastering, the “peak value” of the sample points is
closely watched to ensure that it does not get to
full scale. Since the peak meters in said DAW
and digital mixing systems are inaccurate, and
do not actually indicate the peak values of the
resulting waveform, the result is that while the
samples themselves do not exceed full scale and
are carefully monitored to ensure this, the resulting waveforms represented by the samples
may exceed full scale throughout any standard
CD!
While the digital mixing system is not clipping
the music or distorting the music, the digital to
analog converters that have the task of recreating the audio through digital reconstruction filters are clipping repeatedly throughout most
CDs on the market. The result is that most CDs
and DVDs end up distorting with regularity
when they are asked to reconstruct and play
back audio that appears to be completely “legal”
because not a single sample actually clipped.
Manifestation
Today’s recording environment demands that
sessions are mixed and mastered as “hot” as is
possible, pushing the levels up to the highest tolerable amount, supposedly just short of clipping. Sophisticated digital tools allow music to
be highly compressed, then recompressed, com512
Audio Plug-Ins Guide
D/A converter range
In a recent paper [Nielsen 2003], seven consumer CD players were subjected to tests designed to analyze their ability to reproduce and
reconstruct signal levels above full scale
(0 dBFS). All of the players experienced difficultly dealing with signal levels this high, further showing that, while all of the samples can
be legal, the level can still be hotter than is legal.
The result is that a CD player can be unable to
reproduce the audio accurately. In some cases,
the reconstruction sounds “perfect” to the mastering engineer, because the engineer’s equipment can actually reproduce the waveforms
properly.
The Red Book format for CDs and the DVD specs
both allow for this illegal content and the mastering engineer is still allowed to put out releases that meet the spec while allowing consumers’ players to distort. With an oversampled
peak meter, the engineer will be able to know
that the music is clipping, by how much and
where. With this knowledge the engineer can
then decide with complete information whether
or not to accommodate the legal range of digital
audio on a PCM sampled system.
The goal of TL MasterMeter is to allow an engineer to use a DSP model of the reconstruction
process to monitor the reconstructed waveform
for potential clipping at the final mix and mastering stages. Using TL MasterMeter, engineers
can compare regular and intersample peaks over
time and make appropriate adjustments without
sacrificing overall level or dynamic range. Utilizing an oversampled peak meter in the digital
audio studio that represents the reconstruction
filters in digital to analog converters is the first
step toward an improvement in audio quality in
music releases.
TL MasterMeter References and
Further Reading
Aldrich, Nika. Digital Audio Explained For the
Audio Engineer. San Francisco: Backbeat Books,
2004.
Banquer, Dan, Dick Pierce, Herbie Robinson, et
al. “Intersample Peaking.” Pro Audio Mailing
List. 21 December, 2002 - 31 December, 2002.
Nielsen, Soren and Thomas Lund. “Level Control in Digital Mastering.” Preprint 5019, 107th
AES Convention. Denmark, 1999.
Nielsen, Soren and Thomas Lund. “0 dBFS+
Levels in Digital Mastering.” TC Electronic: Risskov, Denmark. 17 July, 2003. http://www.tcelectronic.com/media/
Level_paper_AES109.pdf
Nyquist, Henry. “Certain Topics in Telegraph
Transmission Theory.” Transactions of the
AIEE. Vol. 47 (April 1928): 617-644.
Shannon, Claude E. “Communication in the
Presence of Noise.” Proceedings of the IRE. Vol.
37 (January 1949): 10-21.
Using TL MasterMeter
TL MasterMeter uses the DSP power of
Pro Tools to model the conversion process
found in typical consumer devices. In technical
terms, the TL MasterMeter algorithm uses a 31tap Blackman-Harris windowed sync conversion
with oversampling ratios from 2x to 8x depending on the session sample rate. The output of
this DSP algorithm is then displayed visually.
This assists engineers in highlighting potential
distortion which may be introduced on playback
of mixes, especially mixes which have been processed to be particularly loud or “hot.”
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TL MasterMeter can be used in two different
ways during a session: Real-Time Metering or
Historical Metering.
TL MasterMeter Controls and
Displays
Real-Time Metering
TL Master Meter Browsers
TL MasterMeter can be used to monitor live signal levels, even if the Pro Tools transport is
stopped. This can be useful in quickly determining the appropriate level for mixing and mastering.
Signal Clip Events Browser
When used in real time, the timecode information displayed in the browsers should be ignored.
Historical Metering
To gain an overall picture of the levels in an entire session, TL MasterMeter can be inserted on
a Master Fader track and the entire session
played from beginning to end. This is typically
done during final mix and mastering.
When session playback is complete, TL MasterMeter shows historical peak and event information for the entire session, as well as a historical
list of events in the browsers for both signal
clips and oversampled clips. You can then manually examine the relevant parts of the session
using the timecode listed in the browsers to determine any appropriate corrective actions.
Signal Clip Events browser
The Signal Clip Events browser displays historical clip events from the current session. The columns displayed show the relevant timecode for
the beginning and ending of a clip event. When
used in a stereo track, the first column shows L
or R to indicate if the left or right channel has
clipped. The Min and Max values in this browser
will always be zero, unless the Clip level is set
below zero. The contents of this browser can be
sorted in ascending and descending order by
any column simply by clicking on the desired
column one or more times.
The time information displayed in this browser
is relative to where the transport started. The
Offset field can be used to adjust the timecode
values if TL MasterMeter is being used for historical metering but the session was started
from a point other than the beginning. If TL
MasterMeter is being used in real time, the timecode information in this browser can be ignored.
At the bottom of the browser, the Peak field displays the highest dB value of the audio signal received so far. The Events field shows the historical total of clip events in the audio signal. Once
TL MasterMeter reaches 2,000 clip events, it
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ceases to record additional events. Although the
meters remain active and the Peak field continues to be updated, new events will not be added
to the browsers. The Events field flashes “2000”
to indicate this condition.
The information in this browser is cleared using
the Clear button, or is cleared automatically
whenever the Pro Tools transport is started.
Oversampled Clip Events Browser
At the bottom of the browser, the Peak field displays the highest dB value of the oversampled
audio received so far. The Events field shows the
historical total of clip events in the oversampled
audio signal. Once TL MasterMeter reaches 2000
clip events, it ceases to record additional events.
Although the meters remain active and the Peak
field continues to be updated, new events will
not be added to the browsers. The Events field
flashes ‘2000’ to indicate this condition.
The Oversampling field displays the current
oversampling factor in use by the DSP processing. This will vary between 2x, 4x and 8x oversampling depending on the session sample rate.
The information in this browser is cleared using
the Clear button, or is cleared automatically
whenever the Pro Tools transport is started.
Oversampled Clip Events browser
The Oversampled Clip Events browser displays
historical clip events from the DSP oversampling of the session audio. The amount of potential clipping in excess of 0 dB is also displayed.
The columns displayed show the relevant timecode for the beginning and ending of a clip
event, as well as the minimum and maximum
clip values created after passing through the
DSP processing. When used in a stereo track, the
first column shows L or R to indicate if the left
or right channel has clipped. The contents of
this browser can be sorted in ascending and descending order by any column simply by clicking on the desired column one or more times.
The time information displayed in this browser
is relative to where the transport started. The
Offset field can be used to adjust the timecode
values if TL MasterMeter is being used for historical metering but the session was started
from a point other than the beginning. If TL
MasterMeter is being used in real time, the timecode information in this column can be ignored.
TL MasterMeter Meters
Signal Level Meters
The Signal Level meter shows the instantaneous
signal level of the current audio signal. The clip
light at the top of the meter can be cleared by
clicking on it, or by using the Clear button.
Oversampled Level Meter
The Oversampled Level meter shows the instantaneous signal level of the current audio signal
after it has been oversampled. As the oversampling process can create levels above 0 dB, this
meter shows an expanded scale from –6 dB to
0 dB and from 0 dB to +6 dB.
The clip light at the top of the meter can be
cleared by clicking on it, or using the Clear button.
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TL MasterMeter Clear Button
TL MasterMeter Clip Field
The Clear button clears all of the historical information displayed in Signal Clip Events
browser and the Oversampled Clip Events
browser. It also click the clip lights at the top of
the Signal Level and Oversampled Level meters.
This information is also cleared when the
Pro Tools transport is activated by pressing Play
or Record.
The Clip field can be used to set the clip threshold at a lower point. For example, if a session
must not exceed –10 dB, the Clip field can be set
to –10 dB and TL MasterMeter will treat that as
the clip threshold for both signal and oversampled clip events. When the Clip field is set to a
non-zero value, the Min and Max values of the
Signal Clip browser are used to indicate the clip
range.
TL MasterMeter Export Button
The Export button exports all of the information
displayed in the two browsers to the clipboard as
tab delimited text. It can then be pasted into any
text or spreadsheet application.
TL MasterMeter View Time
Menu
The View Time menu lets you select the way in
which timing information is displayed, in either
minutes and seconds format, or in samples format. This affects the timecode display in both
the data browsers and the Offset field.
TL MasterMeter Offset Field
The Offset field offsets the values displayed in
both the browsers by the value entered. This is
useful for historical metering but the session
was started from a point other than the beginning. The Enter key must be used after a new offset is typed for it to become active. The information shown in the browsers is updated
immediately when the new Offset is entered.
For example, if the session was started from the
point 1:03.901 (1 minute 3.901 seconds), this
value should be entered into the Offset to ensure
the timecode displayed in both of the browsers
matches that of the Pro Tools session.
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Chapter 93: TL Metro
TL Metro is an RTAS metronome plug-in designed to provide you with the convenience of a
traditional metronome, as well as providing advanced functionality for sophisticated timekeeping requirements.
To configure Pro Tools versions 6.9 or earlier for
use with TL Metro:
1
Select MIDI > Click Options.
In the Click Options dialog, ensure that the velocity for the accented note is higher than that of
the unaccented note. By default, they should be
127 and 100 respectively.
2
3
Click OK.
4
Ensure that the MIDI > Click is enabled.
To configure Pro Tools versions 6.1 or earlier for
use with TL Metro, you must also do the following:
TL Metro plug-in
Configuring Pro Tools for Use
with TL Metro
1
Select MIDI > MIDI Beat Clock.
2
Enable MIDI Beat Clock.
3
Select TL Metro as an output.
4
Click OK.
Create a Pro Tools session as a template
with this MIDI setup and use the template as
a basis for future Pro Tools sessions with TL
Metro.
For TL Metro to work in conjunction with the
Pro Tools transport in “linked” mode, it must
receive MIDI from Pro Tools. This is configured
in each Pro Tools session.
To configure Pro Tools versions 7.x or higher for
use with TL Metro:
1
Create a new Pro Tools session.
Create a new audio, Auxiliary Input, or Instrument track.
2
Factory Presets
TL Metro provides a number of factory presets
that provide a range of sounds.
To audition a preset:
3
Insert TL Metro on the new track.
Select the desired preset from the Plug-In Librarian menu.
4
Ensure that Options > Click is enabled.
2
1
Click Play in TL Metro.
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517
TL Metro Controls and
Displays
Volume Sliders
The volume of each individual note can be adjusted using the five Volume sliders. If the volume slider for the accented whole note is reduced to zero, the quarter note will be played
instead of the whole note.
Tempo Controls
Tempo can be specified by manually entering
the tempo, or using the provided slider. Tempo
controls are disabled when TL Metro is linked to
Transport and Tempo.
Tempo controls
Link Status
TL Metro can be linked to the Pro Tools Transport or to the Pro Tools Transport and Tempo
track. For more information, see “Synchronizing TL Metro to Pro Tools” on page 519.
Beats Per Measure Selector
Volume sliders
Sample Selectors
Select the desired audio sample played for each
of the five different notes from the corresponding Sample selector. A sample can be selected
from any of up to 50 sample slots.
Select the number of beats per measure using
the Beats Per Measure selector. If Link Status is
set to Transport+Tempo, TL Metro uses the
Pro Tools session’s Meter track and the Beats
Per Measure selector is unavailable.
Selecting the number of beats per measure
Sample selectors
Master Volume
The Master Volume slider controls the overall
volume of the metronome audio signal.
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Sound Library
The Sound Library menu lets you import custom
samples for specific beats. For more information, see “Importing Custom Samples to TL
Metro” on page 521.
Play Button
Using TL Metro and Control
Surfaces
TL Metro parameters can be assigned to a control surface, such as D-Command, Command|8,
Control|24, or Pro Control. The abbreviated
name for each of the beats when displayed on a
control surface as follows.
The Play button activates the metronome. In
linked modes, the Play button is disabled and
the metronome is activated when the Pro Tools
transport is engaged.
• Accented Quarter Note = Beat 1
Tap Button
• Triplet = Beat 5
The Tap button provides a tap tempo function.
Click the tap button in time with the beat to determine the beast. The detected tempo is displayed in the Tempo field and in the LCD display.
TL Metro Information Display
• Quarter Note = Beat 2
• Eighth Note = Beat 3
• Sixteenth Note = Beat 4
Synchronizing TL Metro to
Pro Tools
TL Metro can be synchronized to the Pro Tools
Transport and Tempo using the Link Status selector.
The LCD style information display in TL Metro
displays the following:
• The current tempo in beats per minute
(bpm)
• The current beat of the measure
• Link status
The MIDI name of this instantiation of the
TL Metro plug-in also appears in the display beneath the tempo. This is typically shown as “TL
Metro 1,” “TL Metro 2,” or similar. This enables
multiple instantiations of TL Metro to be easily
identified when routing MIDI.
If a flashing question mark appears in the information display, this indicates TL Metro has encountered an error. For example, MIDI Beat
Clock may not be configured correctly. Click on
the question mark for a dialog window with additional information.
Selecting TL Metro Link Status
Unlinked
When the Link Status is set to None, the TL
Metro can be started and stopped independently
of the Pro Tools Transport and Tempo. This is
useful for recording when you only need the
metronome for a few bars.
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519
Linked to Transport
When the Link Status is set to Transport, the
metronome will start and stop automatically
when the Pro Tools Transport is engaged or disenganged.
When using TL Metro linked to Transport, three
points should be kept in mind:
• Ensure that MIDI is correctly configured
for TL Metro in Pro Tools (see “Configuring
Pro Tools for Use with TL Metro” on
page 517).
• The tempo in TL Metro must be set manually.
• TL Metro assumes you are starting from the
beginning of each bar when you start the
Transport.
Linked to Transport and Tempo
TL Metro can also be linked to both the
Pro Tools Transport and Tempo. In this mode,
TL Metro automatically follows the tempo of the
Pro Tools session in addition to following the
Transport.
Ensure that MIDI is correctly configured for TL
Metro in Pro Tools (see “Configuring Pro Tools
for Use with TL Metro” on page 517).
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Audio Plug-Ins Guide
Customizing TL Metro
Presets
TL Metro provides a selection of factory presets,
including commonly used click sounds. These
presets can be selected from the Plug-In Librarian menu.
User created presets can also be stored using the
Plug-In Settings menu.
To make any preset the default when TL Metro is
instantiated:
From the Plug-In Librarian menu, select the
desired preset.
1
2
From the Plug-In Settings menu, select Set As
User Default.
From the Plug-In Settings menu, select Settings Preferences > Set Plug-In Default To > User
Setting.
3
For more information on using plug-in presets in Pro Tools, see the Pro Tools Reference Guide .
Importing Custom Samples to
TL Metro
TL Metro supports up to 50 different samples for
metronome click sounds. TL Metro includes factory samples in the first 40 slots, the remaining
slots are marked as “<Unassigned>.”
TL Metro supports import of WAV and AIFF
sound files for specific beat sounds. Sounds can
be loaded into any one of the 50 available slots.
Typically, user samples are loaded into the unassigned slots in order to avoid overwriting the
factory samples. However, any of the 50 slots
can be replaced by user imported samples if desired.
For best results, imported sounds should have
the following characteristics.
• The sound should start in the very first
sample of the file, and have a sharp attack
to ensure proper timing.
Factory and imported samples are stored in a
preferences file named “TL Metro Plug-In” located in your system preferences folder. On
Windows, it’s located in <system drive letter>:\Documents and Settings\<user
name>\Application Data\Trillium Lane\TL
Metro PlugIn.rsr. On Macintosh, it’s located in
Users\<user name>\Library\Preferences\TL
Metro Plug-In.
If you want to use the particular samples you imported into TL Metro on a different Pro Tools
system, copy this preferences file between systems. If the TL Metro preferences file is deleted,
all factory and user samples will be deleted. To
restore TL Metro to the factory samples only,
quit Pro Tools and delete this preferences file.
The next time you use TL Metro, it will recreate
the preferences file with only the factory samples.
• The sample should be normalized before
importing.
• Sound length should be limited to approximately one second to avoid playback problems.
To import a sound:
Click the Sound Library button to display the
sample menu.
1
2
Select an unassigned slot.
3 In the resulting File dialog, select the WAV or
AIFF file you want to import.
4
Click OK.
The name of the selected file is displayed in each
sample menu. To use the imported sample, select it from the sample menu for the appropriate
beat.
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Audio Plug-Ins Guide
Chapter 94: Trim
The Trim plug-in is available in TDM and RTAS
formats and can be used to attenuate an audio
signal from – (Infinity) dB to +6 dB or –
(Infinity) dB to +12 dB. For example, using a
multi-mono Trim plug-in on a multi-channel
track provides simple, DSP-efficient muting
control over the individual channels of the
track.
This capability is useful, since Track Mute buttons mute all channels of a multi-channel track
and do not allow muting of individual channels
within the track.
Trim Controls
The Trim plug-in provides the following controls:
Phase Invert Inverts the phase (polarity) of the
input signal to change the frequency response
characteristics between multi-miked sources or
to correct for miswired microphone cables.
Gain Provides – dB to +6 dB or +12 dB of gain
adjustment, depending whether the Gain toggle
is set to +6 or +12.
+6/+12 Gain Toggle Switches the maximum level
of attenuation between – dB to +6 dB and
– dB to +12 dB.
Trim plug-in
Alt-click (Windows) or Option-click (Mac)
the Trim selector to open a Plug-In window
for each channel of a multi-channel track.
Automation data adjusts to reflect the current Gain setting. When working with automation data from an older version of the
Trim plug-in, ensure the Gain setting is set
at +6 dB.
Output Meter Indicates the output level, including any gain compensation added using the Gain
control.
Mute Mutes the signal output.
Chapter 94: Trim
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Chapter 95: Other AudioSuite Plug-In
Utilities
The following AudioSuite-only utility plug-ins
are installed when you install Pro Tools:
• DC Offset Removal
• Duplicate
• Gain
To remove DC offset from an audio clip:
1
Select the clip with DC offset.
2
Choose AudioSuite > Other > DC Offset Re-
moval.
3
Ensure that Use In Playlist is enabled.
4
Click Render.
• Invert
• Normalize
• Reverse
Duplicate
DC Offset Removal
The DC Offset Removal plug-in removes DC offset from audio files. DC offset is a type of audio
artifact (typically caused by miscalibrated analog-to-digital convertors) that can cause pops
and clicks in edited material.
To check for DC offset, find a silent section in
the audio material. If DC offset is present, a
near-vertical fade-in with a constant or steadystate offset from zero will appear in the waveform. Use the DC Offset Removal plug-in to remove it.
The Duplicate plug-in duplicates the selected
audio in place. Depending on how its controls
are configured, the new clip will appear in either
the Clip List or playlist. You can use this to flatten or consolidate an entire track consisting of
multiple clips into one continuous audio file
that resides in the same place as the original individual clips.
Duplicate plug-in
DC Offset Removal plug-in
The audio is unaffected by Pro Tools volume or
pan automation, or by any real-time plug-ins
that may be in use on the track as inserts. The
original audio file clips are merely rewritten in
place to a single duplicate file.
Chapter 95: Other AudioSuite Plug-In Utilities
525
The Duplicate plug-in works nondestructively.
You cannot choose to overwrite files.
To change the gain of an audio clip:
1
Select the clip whose gain you want to change.
To duplicate an audio selection:
2
Choose AudioSuite > Other > Gain.
1
Select the audio you want to duplicate.
3
Adjust the Gain slider as desired.
2
Choose AudioSuite > Other > Duplicate.
4
Click Preview to audition your changes.
3
Ensure that Use In Playlist is enabled.
5
Ensure that Use In Playlist is enabled.
4
Click Render.
6
Click Render.
Gain
Invert
The Gain plug-in boosts or lowers a selected
clip’s amplitude by a specific amount. Use it to
smooth out undesired peaks and other dynamic
inconsistencies in audio material.
The Invert plug-in reverses the polarity of selected audio. Positive sample amplitude values
are made negative, and all negative amplitudes
are made positive.
This process is useful for altering the phase or
polarity relationship of tracks. The Invert plugin is useful during mixing for modifying frequency response between source tracks recorded with multiple microphones. You can also
use it to correct audio recorded out of phase
with an incorrectly wired cable.
Gain plug-in
Gain Specifies the desired gain level. Set this
value by manually adjusting the Gain slider, or
by entering a numeric decibel value, or by entering a percentage.
526
Invert plug-in
Find Level When clicked, displays the peak amplitude value of the current selection.
To invert the phase an audio clip (or selection):
1
Select the clip whose phase you want to invert.
RMS/Peak Toggle Switches the calibration of
gain adjustment between Peak or RMS modes.
Peak mode adjusts the gain of the input signal to
the maximum possible level without clipping.
RMS mode adjusts the input signal to a level
consistent with the RMS (Root-Mean-Square)
value, or the effective average level of the selected clip.
2
Choose AudioSuite > Other > Invert.
3
Ensure that Use In Playlist is enabled.
4
Click Render.
Audio Plug-Ins Guide
Normalize
The Normalize plug-in optimizes the volume
level of an audio selection. Use it on material recorded with too little amplitude, or on material
whose volume levels are inconsistent (as in a
poorly recorded narration).
Unlike compression and limiting, which modify
the dynamics of audio material, normalization
preserves dynamics by uniformly increasing (or
decreasing) amplitude.
To prevent clipping during sample rate conversion, Normalize to no greater than the
range between –2 dB to –0.5 dB. Optimum
settings will vary with your program material and your Conversion Quality setting (in
the Editing tab of the Preferences dialog).
RMS/Peak Toggle Switches the calibration of
normalizing between Peak or RMS modes. Peak
mode normalizes the input signal at the maximum possible level without clipping. RMS mode
normalizes the input signal at a level consistent
with the RMS (Root-Mean-Square) value, or the
effective average level of the selected clip.
Normalizing Multiple Clips Across Tracks
When multiple clips are selected across multiple
tracks, the Normalize plug-in can search for
peaks in two different modes:
Peak On Each Chan/Track Searches for the peak
level on a channel-by-channel or track-by-track
basis.
Peak On All Chans/Tracks Searches for the peak
level of the entire selection. If ten tracks are selected, for example, the Normalize function will
find the peak value from all ten.
To normalize an audio clip (or selection):
1
Select the clip you want to normalize.
2
Choose AudioSuite > Other > Normalize.
3
Adjust the Level slider as desired.
4
Ensure that Use In Playlist is enabled.
5
Click Render.
Normalize plug-in
Max Peak At Specifies how close to maximum
level (clipping threshold) the peak level of a selection is boosted. Set this value by adjusting the
Max Peak At slider, by entering a numeric decibel value below the clipping threshold, or by entering a percentage of the clipping threshold.
You can normalize stereo pairs together so that
two sides of a stereo signal are processed relative to each other.
Chapter 95: Other AudioSuite Plug-In Utilities
527
Reverse
The Reverse plug-in replaces the audio with a
reversed version of the selection. This is useful
for creating reverse envelope effects for foley,
special effects, or musical effects.
Reverse plug-in
To reverse an audio clip (or selection):
528
1
Select the clip you want to reverse.
2
Choose AudioSuite > Other > Reverse.
3
Ensure that Use In Playlist is enabled.
4
Click Render.
Audio Plug-Ins Guide
Part XIV: Eleven
Chapter 96: Eleven and Eleven Free
Eleven is a guitar amplifier plug-in that is available in TDM, RTAS, and AudioSuite formats.
Eleven gives you stunning guitar amplifier, cabinet, and microphone models of the “best of the
best” vintage and contemporary gear.
Eleven Free is a free version of Eleven that
comes with every Pro Tools system, with a reduced feature set. Eleven Free comes in RTAS
and AudioSuite formats only.
Eleven Plug-In Features
• Classic amp models that faithfully recreate the
sound and dynamic response of the original
amps.
• Highly accurate speaker cabinet models with
variable speaker breakup (cone distortion).
• Support of up to 8 channel (7.1) operation, in
mono or multi-mono plug-in only.
Eleven Free Plug-In Features
• Two custom amp models from Avid.
• Two speaker cabinet models.
• Amps and cabinets can be mixed and matched.
• Noise Gate to control any unwanted noise.
• Settings files (presets) to store and recall factory and custom tones.
• Support of any compatible work surface or
MIDI controller. MIDI Learn provides effortless mapping to any continuous controller
(CC)–capable MIDI device.
• Selectable mics, with on- and off-axis options.
• Support of sample rates of 96 kHz, 88.2 kHz,
48 kHz, and 44.1 kHz.
• Amps, cabs, and mics can be mixed and
matched into nearly limitless combinations.
• Support of up to 8 channel (7.1) operation, in
mono or multi-mono plug-in only.
• Amps and cabs can be bypassed separately.
• All controls can be automated.
• Noise Gate to control any unwanted noise.
.
Eleven can share preset data with the
Eleven Rack guitar processor/audio interface from Avid. For more information, see
the Eleven Rack User Guide.
• Settings files (presets) to store and recall factory and custom tones.
• Support of any compatible worksurface or
MIDI controller. MIDI Learn provides effortless mapping to any continuous controller
(CC)–capable MIDI device.
• Support of sample rates of 96 kHz, 88.2 kHz,
48 kHz, and 44.1 kHz.
Chapter 96: Eleven and Eleven Free
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Chapter 97: Eleven Input Calibration and
QuickStart
This section shows you how to get connected, calibrated, and cranking through Eleven as quickly as
possible.
Before You Begin with Eleven
Eleven was designed to model the essential aspects of each amplifier including characteristics of the
input stage. Providing an appropriate level of signal delivers the most accurate response from the
plug-in.
• If you are working with pre-recorded guitar tracks, see “Using Eleven with Pre-Recorded Tracks”
on page 537.
• If you are working with a live guitar signal, follow the steps on the next few pages for optimal input level calibration. Input calibration takes only a couple of minutes, and helps ensure the best
results with Eleven, its amps, and its factory presets.
Source
Hardware
Pro Tools
Eleven
Input LED
(Should be
yellow or
orange)
Vol at max
Hardware input gain
Pro Tools level
Basic gain stages to calibrate live guitar input for Eleven
Chapter 97: Eleven Input Calibration and QuickStart
533
Connect your Guitar and
Configure Source Input
If your setup includes pedals or other gear, it
helps to know whether the final output device is
providing an instrument- or line-level signal.
Choose and configure your input and source settings accordingly. (Check the Setup Guide that
came with your system for more information.)
To connect your guitar to a Pro Tools host-based
system:
Do one of the following, depending on your
hardware configuration:
1
• If you are using an interface that has a DI
input (such as an Mbox Pro), plug your guitar into an available DI input.
– or –
s
To connect your guitar to a Pro Tools|HD system:
Make sure you have a pre-amp (such as an Avid
PRE ®) or similar unit connected to a
Pro Tools|HD audio interface. (Note that
HD OMNI provides built-in preamps.)
1
Plug your guitar into an available pre-amp input and set its source, impedance, and other settings as needed for your setup.
2
If you use a direct box to convert your guitar’s hi-impedance output to a low-impedance signal, set the Line/Inst 1 input to Line
source or the equivalent on your particular
pre-amp.
For example, if using a Avid PRE you can plug
your guitar directly into the front panel
Line/Inst 1 input, then set its source to Inst.
• If you are using your computer’s built-in inputs, plug your guitar into an available input.
If you use a direct box to convert your guitar’s hi-impedance output to a low-impedance signal, connect the direct box to a mic
or line input instead of the DI input.
Make sure to use the correct input on your interface. For example, on Mbox Pro, plug your
guitar into front-panel DI Inputs 1 or 2.
2
PRE (or other pre-amp)
Pro Tools|HD audio interface
Guitar into Avid PRE into a 192 I/O™
Set Hardware and Levels
After plugging in, do the following to set your
primary gain and configure your Pro Tools
hardware by watching its input indicators (meters). This sets the first stage of your gain structure for Eleven.
Mbox Pro DI Inputs 1 and 2
Mbox Pro back-panel 1/4” inputs are
line-level only and should not be used
with a guitar.
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Audio Plug-Ins Guide
To prepare your guitar and Pro Tools host-based
hardware for input calibration:
Set Up a Pro Tools Track
In Pro Tools, choose Setup > Playback Engine
and set your Hardware Buffer to a low enough
setting to reduce monitor latency.
In this step, you’ll create and configure an audio
track to use for the final stage of input calibration.
1
On your guitar, select the highest output
pickup or position and set the volume and tone
controls to 10 (maximum).
2
Strum full chords (your loudest expected playing) while watching the Input indicators on your
audio hardware.
To set up and check Track level (all systems):
Choose Tracks > New, and create one mono Audio track.
1
3
Adjust the Input Gain on your audio interface
high enough to indicate a strong signal on the
hardware Input LED (but not overloading the input).
4
2 In the Mix window, click the track Input selector and choose your guitar input.
3
Click the track Insert selector and select
Eleven.
Eleven
Guitar input
Input 1 Gain on Mbox Pro
To prepare your guitar and Pro Tools|HD hardware
for input calibration:
Track meter
On your guitar, select the highest output
pickup or position and set all volume and tone
controls to the maximum.
1
Strum full chords (your loudest expected playing) while watching the Input indicators on your
audio hardware.
2
Adjust your pre-amp input gain until you see a
strong signal on your audio interface Input meters (but not overloading the input).
3
One audio track for input calibration on Pro Tools
Record enable the audio track, or enable its
TrackInput monitoring button (Pro Tools|HD
only).
4
Chapter 97: Eleven Input Calibration and QuickStart
535
Set Up Eleven
Use Eleven’s Input LED to make final gain adjustments and complete the input calibration process.
To calibrate your input signal to the Eleven plug-in:
1
Open the Eleven plug-in window by clicking its insert slot. Leave it at its default settings.
Eleven’s Input LED (left) and Clip LED (right)
2 Strum as hard as you can a few more times and
watch Eleven’s Input LED to see where your level
registers. The Input LED lights green, yellow, orange, or red to indicate the following level
ranges:
3 Leaving the Input control on the plug-in at its
default setting of 0 (12:00 position), set the signal level going to the plug-in by adjusting the input gain control on your hardware until Eleven’s
Input LED shows yellow or orange.
Green (Off to –8)
4
Indicates signal is present, but too low.
Yellow (–8 to –4)
Indicates the best level for low output sources,
such as single coil pickups.
Orange (–4 to 0)
Indicates the best level for higher output
sources, such as humbucker pickups.
Red (0 and above)
Indicates that you have clipped the plug-in
input. Click the Input LED to clear the clip
indicator.
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After calibrating, strum as you normally
would and/or back down your guitar volume
from the maximum setting used for input calibration. Don’t worry about the Input LED showing yellow or orange when playing normally. As
long as the plug-in isn’t indicating clipping, your
gain staging should be established.
5 Adjust the Output knob in Eleven’s Master section to raise or lower the plug-in output signal.
Proper input calibration of live guitar does
not require any adjustment of Eleven’s Input
control. To learn how this control was designed to work with the amp models, see “Input” on page 542.
Using Eleven with Pre-Recorded
Tracks
If the pre-recorded tracks were not calibrated
with the Eleven plug-in using the method previously described, you can use the Input control in
Eleven to adjust the signal level feeding the input stage of the amp model.
Use your ears as a guide and adjust to taste.
Since the Input LED measures the signal level
entering the plug-in and precedes the input control, you will not see any changes to the Input
LED as you make adjustments.
Getting Started Playing
Music with Eleven
To get started playing music with Eleven:
Make sure you already calibrated your input
signal as explained in the previous sections of
this chapter.
1
Click the plug-in’s Librarian menu and choose
a factory preset, then play guitar. Take your time
to explore — the Presets let you hear all of
Eleven’s different amps and combos.
2
See “Processing Pre-Recorded Tracks
Through Eleven” on page 551 for more information.
Librarian menu (left) and the Settings menu (right)
Pick any amp and cabinet from the available
types (see “Pairing Amps and Cabinets” on
page 546.)
3
Turn to Chapter 98, “Using Eleven” for specific
details on Eleven’s main controls, and for suggested track setups for recording, jamming, and
mixing.
4
Use the Settings menu to save, copy, paste,
and manage plug-in settings files. To save a
setting, see “Eleven Settings (Presets)” on
page 541.
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Chapter 98: Using Eleven
The following sections introduce you to the
main sections and controls in Eleven and show
you how to use them. You’ll also find suggested
track setups and signal routing tips to help you
get the most out of Eleven.
Inserting Eleven on Tracks
Eleven can be inserted on Pro Tools audio, Auxiliary Input, Master Fader, or Instrument tracks.
To insert Eleven on a track:
 Click an Insert selector on the track and
choose Eleven or Eleven LE.
Channel Formats
Eleven is available as a mono or multi-mono
plug-in only. For use in stereo or greater formats choose the multi-mono version.
Sample Rates
Eleven supports 96 kHz, 88.2 kHz, 48 kHz and
44.1 kHz sample rates.
Category and Manufacturer
When Pro Tools plug-ins are organized by Category or Manufacturer, Eleven is listed as follows:
Adjusting Eleven’s
Parameters
This section tells you how to adjust parameters
using your mouse or a Pro Tools worksurface.
For information on MIDI control, see “Using
MIDI and MIDI Learn with Eleven” on page 540.
Editing Parameters Using a Mouse
You can adjust Eleven’s rotary controls by dragging horizontally or vertically. Parameter values
increase as you drag upward or to the right, and
decrease as you drag downward or to the left.
Keyboard Shortcuts
 For finer adjustments, Command-drag (Mac)
or Control-drag (Windows) the control.
 To return a control to its default value, Option-click (Mac) or Alt-click (Windows) the control.
Navigating the Amp, Cab, and Mic Type
Selectors
You can click on the name of the current Amp
Type, Cab Type, or Mic Type to display their
pop-up menus and select an item.
You can also click the Previous/Next arrows to
step through Amp, Cabinet, and Mic choices one
at a time.
Category Harmonic
Manufacturer Digidesign
Chapter 98: Using Eleven
539
About Unused Controls and Worksurfaces
Previous arrow (top) and Next arrow (bottom) (Amp
Type shown)
You can control the Amp, Cab, and Mic
Type selectors with MIDI. See “Using MIDI
and MIDI Learn with Eleven” on page 540
Enabling Switches
To enable or disable a switch or button, such as
Amp Bypass, click it to toggle its setting.
Some amps that have relatively few controls
(such as the Tweed Lux) will display controls on
a worksurface that are not actually available
with that particular amp model. Even though
you can adjust these unused encoders or
switches, only those controls seen on-screen for
any amp can be adjusted from a worksurface.
Changing an unused control does nothing to the
current amp, but does alter the value of that unused control. If you switch to a different amp
that does include that (previously unused) control, the new amp inherits the altered setting
which can lead to sudden jumps in gain or other
settings.
Groups and Linked Plug-In Controls
Eleven’s parameters can follow Pro Tools
Groups (Mix, Edit, or Mix/Edit) for linked control of multiple inserts. For more information,
see the Pro Tools Reference Guide.
Using Automation
All of Eleven’s parameters can be automated.
When a parameter has been enabled for automation, an LED appears lit near that control.
See the Pro Tools Reference Guide for more
information on plug-in automation.
Using a Pro Tools
Worksurface with Eleven
Eleven can be controlled directly from any compatible Pro Tools worksurface. Eleven appears
along with other plug-ins and can be assigned,
edited, bypassed and automated using the Insert
section as available on the particular worksurface being used.
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Audio Plug-Ins Guide
Using MIDI and MIDI Learn
with Eleven
Eleven supports MIDI Control Change (CC)
messages, meaning that the Master section,
amp, cabinet and mic parameters can be controlled remotely by any CC-capable MIDI device. This includes MIDI controllers, mixers,
and instruments, as well as the 003 ® (in MIDI
Mode).
MIDI Learn lets you quickly map plug-in controls to a MIDI foot pedal, switch, fader, knob,
or other CC-compatible trigger. You can also
manually assign controls to specific MIDI CC
values.
It’s a Session Thing
MIDI control assignments are saved and restored with the Pro Tools session in which they
are defined. Settings files (presets) for Eleven do
not store or recall MIDI Learn assignments.
To map a MIDI controller to a parameter:
Make sure your external MIDI device is connected to your system, and recognized by your
MIDI Studio Setup (Windows) or Audio MIDI
Setup (Mac).
1
2
Right-click on any control in Eleven.
Eleven Settings (Presets)
You can pick a preset from the plug-in Librarian
menu.
To load a preset:
 Click the Librarian menu and select an available Settings file.
Librarian menu
Settings menu
Plug-In controls for Eleven Settings files
Right-clicking for MIDI Learn
If your Mac does not have a two-button
mouse, Control-click an Eleven parameter
to show the MIDI Learn menu. Note that you
won’t be able to use the Control key modifier
to “clutch” a Grouped control.
3
Do either of the following:
• Click Learn, then move the desired control
on your MIDI controller. Pro Tools maps
whichever control you touch to that plug-in
parameter.
– or –
• If you know the MIDI CC value of your foot
controller or other device, select it from the
Assign menu.
To clear a MIDI assignment:

Right-click the control and choose Forget.
You can save, import, copy, paste, and manage
settings using the Settings menu.
To save your settings as an Eleven preset:
1
Configure Eleven for the desired tone.
Click the Settings menu and choose Save Settings. Name the preset, choose a location, and
2
click Save.
You can scroll through and select preconfigured
Eleven Settings files (presets) using the plug-in
Librarian menu, and the +/– buttons.
For more information on Settings files and
folders, see the Pro Tools Reference Guide.
Master Section
The Master section includes plug-in I/O (input/output) and noise gate controls, the
Amp Type selector and the Cab Type selector.
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541
The Master section doesn’t change when you
switch amps. Master section settings are stored
and recalled with plug-in presets.
Input
Gate
Amp Type
Cab Type
Output
Output
The Output control sets the output gain after
processing, letting you make up gain or prevent
clipping on the channel where the plug-in is being used. Output range is –60 dB to +18 dB.
When you want to adjust Eleven’s output
level, use the Output knob. For tone/distortion, use the amp Master volume.
Master section
Input LED
The Input LED shows green, yellow, orange, or
red to indicate whether you are under- or overdriving the plug-in. The Input LED is before the
Input section of the Master section. To learn
more about the Input LED within the Eleven signal chain, see “Eleven Signal Flow Notes” on
page 559.
Amp Type selects which amplifier model to use
(see “Amp Types” on page 543).
Cab Type
This selector lets you select which speaker cabinet model to use (see “Eleven Cabinet Types” on
page 546).
Input
Gate
The Input knob provides input trim/boost, for
tone and distortion control. The Input range is
–18 dB to +18 dB.
Noise Gate Threshold
The Input knob provides a great way to increase
or decrease gain with amp models that don't
have a separate preamp control. It also provides
a way to trim or boost the level of pre-recorded
tracks you want to treat with Eleven
It is important to note that the setting of the Input knob is saved and restored with Settings
files (presets).
To learn more about the Input control, see
“Eleven Signal Flow Notes” on page 559
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The Noise Gate Threshold control sets the level
at which the Noise Gate opens or closes. At minimum Threshold setting, the Noise Gate has no
effect. At higher Threshold settings, only louder
signals will open the Gate and pass sound.
Threshold range is from Off (–90 dB) to –20 dB.
Noise Gate Release
The Noise Gate Release control sets the length of
time the Noise Gate remains open and passing
audio. Adjust the Release to find the best setting
for the current task (not too fast to avoid cutting
off notes, and not too slow to avoid unwanted
noise). Release range is from 10 ms to 3000 ms.
For suggested gate applications, see “Using
the Noise Gate” on page 543. For details on
where it derives its key (trigger) and applies
its gate, see “Eleven Signal Flow Notes” on
page 559.
Amp Types
The Amp Type selector lets you choose an amp.
Using the Noise Gate
You can use the Noise Gate to silence unwanted
signal noise or hum, or just for an effect.
Choosing an amp from the Amp Type selector
To use the Noise Gate to clean up unwanted, low
level noise:
Connect and calibrate your guitar as explained
in “1: Connect your Guitar and Configure Source
Input” on page 534.
1
For the next steps, hold your guitar but don’t
play it (and be sure to leave its volume up). You
should hear only the noise that we’ll soon get rid
of.
2
Available Amp Types in Eleven include the following:
• ’59 Tweed Lux *
• ’59 Tweed Bass *
• ’64 Black Panel Lux Vibrato *
• ’64 Black Panel Lux Normal *
• ’66 AC Hi Boost *
• ’67 Black Panel Duo *
To make it easier to hear the effect, begin by
setting the Release to its middle (12 o’clock) position.
• ’69 Plexiglas – 100W *
Now raise the Threshold control to its highest
setting, fully clockwise, so that the Gate fully
closes (you shouldn’t hear anything coming
through Eleven).
• ’89 SL-100 Drive *
3
4
Slowly lower the Threshold control until the
Gate opens again to find the cutoff (or, threshold) of the noise.
5
Raise the Threshold control again slightly, increasing it only enough to once again silence the
noise (hold Command (Mac) or Ctrl (Win) while
adjusting to be able to fine-tune the setting in
tenths of a dB). Now you’re in the ballpark.
6
7 If you lowered the Release setting as suggested
in step 3, make sure to return it to its maximum
setting (fully clockwise) before continuing.
• ’82 Lead 800 – 100W *
• ’85 M-2 Lead *
• ’89 SL-100 Crunch *
• ’89 SL-100 Clean *
• ’92 Treadplate Modern *
• ’92 Treadplate Vintage *
• DC Modern Overdrive
• DC Vintage Crunch
* These models only appear in the full version of
Eleven.
Eleven is not affiliated with, or sponsored or
endorsed by, the makers of the amplifiers
emulated in the product.
Chapter 98: Using Eleven
543
Eleven Amp Controls
Each Eleven amp provides a set of controls similar to (and in some cases identical to) those on the actual amp it models. The following sections give a general overview of amp controls.
Bypass
(Amp) Bright
Bass
Gain 1
Mid
Tone
Speed Depth
Treble
Presence
Tremolo Master Volume
Amp controls in the default Amp Type
Amp Bypass
The Amp Bypass switch (or lamp) lets you bypass just the amp model, leaving the cab and mic
settings in effect. The default setting is On.
When set to Bypass, only the amp is bypassed;
Master section, cabinet and microphone settings remain active.
All Eleven controls provide identical ranges
as the original amps, but some numbers
have been adjusted for consistency.
Bright
Gain 2
The Bright switch provides extra high frequency
response to the input signal, and alters the timbre of the distortion. On some amp models, the
effect is most apparent at lower volume settings.
Gain 2 is a second Gain knob used with some
amp models that determines the amount of
overdrive in the pre-amp stage. Gain 2 (also
known as “Presence” on some amps) allows for
more harmonic subtleties in character of the
amp tone. The default is 5.0. Gain 2 range is
from 0 to 10.
Gain 1
Gain 1 determines the overall gain amount and
sensitivity of the amp. When Gain 1 is low it allows for cleaner, brighter sounds with enhanced
dynamic response. When set high, the entire
personality of the amp changes, becoming fatter
and overdriven. Gain 1 responds differently
544
with each amp model and is designed to have a
musical response that closely matches that of its
original amp, at all settings. The default setting
is 5.0. Gain 1 range is from 0 to 10.
Audio Plug-Ins Guide
Parallel or Series The Gain 2 control on the
Tweed Lux, AC Hi Boost and Plexiglass is in parallel (“jumped”) with the Gain 1 control. The
M-2 Lead is in series, meaning the signal goes in
and out of Gain 1, then into Gain 2.
Tone
Presence
Tone controls let you shape the highs, mids and
lows of the amp sound. Electric guitar pickups
tend to have a strong low-mid emphasis and little high frequency response, often producing a
mid-range heavy sound that requires some treble boost. The response and interaction of the
tone controls are unique to each amp.
The Presence control provides a small amount of
boost at frequencies above the treble control.
Presence is applied at the end of each amp model
pre-amp stage, acting as a global brightness control that is independent of other tone controls.
The default setting is 3.0. The Presence range is
from 0 to 10.
Bass
Master
The Bass control determines the amount of low
end in the amp tone. The response of this control in some models is linked to the setting of the
Treble control. The default setting is 5.0. Bass
range is from 0 to 10.
Middle
The Middle control determines the mid-range
strength in lower gain sounds. With high gain
amp models, the Middle control has a more dramatic effect and can noticeably shape the sound
of the amp at both the minimum and extreme
settings. The default setting is 5.0. The Middle
range is from 0 to 10.
Treble
In most amp models, the Treble control is the
strongest of the three tone controls. Its setting
determines the blend and strength of the Bass
and Middle controls. When Treble is set to
higher values, it becomes the dominant tone
control, minimizing the effect of Bass and Middle controls. When Treble is set to lower values,
the Bass and Middle have more effect, making
for a darker amp tone. The default setting is 5.0.
The Treble range is from 0 to 10.
The Master control sets the output volume of the
pre-amp, acting as a gain control for the power
amplifier. In a standard master-volume guitar
amp, as the Master volume is increased more
power tube distortion is produced. The default
setting is 5.0. Master range is from 0 to 10.
Some might assume a Master volume knob
capable of silencing the amp completely. Not
so. Use the Output knob (in the Master section) to silence the output of the plug-in. Use
Master volume for tone and distortion.
Tremolo
Tremolo is achieved through the use of amplitude modulation, multiplying the amplitude of
the pre-amp output by a waveform of lower frequency. Tremolo is not available on all amps.
Tremolo Speed The Speed control sets the rate
of the Tremolo effect. The Tremolo Speed LED
pulses at the rate of Tremolo Speed. The default
setting is 5.0.
Eleven does not support Tempo Sync.
Tremolo Depth The Depth controls the amount
of the Tremolo effect. The default setting for
this control is 0.0, which is equivalent to off.
Some amp models call the Tremolo Depth control Intensity.
Chapter 98: Using Eleven
545
Eleven Cabinet Types
The Cab Type selector lets you pick a cabinet to
use with the current amp. The selected cabinet
and its controls are displayed directly below the
amp controls.
Cabinet Type selector in the Master section
Available cabinets include the following:
• 1x12 Black Panel Lux *
• 1x12 Tweed Lux *
• 2x12 AC Blue *
• 2x12 Black Panel Duo *
• 4x10 Tweed Bass *
• 4x12 Classic 30
• 4x12 Green 25W
* These models only appear in the full version of
Eleven.
Cabinets are listed by their number and diameter of their speakers. For example, “1x12” means
a cabinet has a single 12-inch speaker.
Eleven is not affiliated with, or sponsored or
endorsed by, the makers of the loudspeakers
and cabinets that are emulated in the product.
Visit the Avid website (www.avid.com) to
learn about each of the cabinets used to create Eleven.
Pairing Amps and Cabinets
Eleven lets you combine amps and cabinets in
traditional pairings (such as the combo amps
through their default cabinets) and non-traditional match ups.
Some of the amps modeled in Eleven are
“combo” amps. Combo amps have both their
amp and speaker housed in the same physical
box, meaning there is one and only one cabinet
associated with the signature sound of a combo
amp. The Tweed Lux and AC Hi Boost are both
examples of combo amps.
Other amps are amps-only (heads), and were designed to be run through a speaker cabinet.
Many amp/cab pairings have become standards.
Using Settings for Realistic and Classic
Pairings
You can use Eleven’s factory Settings files (presets) for combo amps and classic combinations.
Settings files store and recall all controls, (including Amp and Cabinet Type).
For combo amps and default combinations:
 Choose a factory Settings file for that amp
from Eleven’s Settings menu.
Using the Amp Type and Cabinet Type
Selectors for Unlinked Pairing
You can use the Amp Type and Cabinet Type selectors to try your own, unique combinations.
If you want to combine amps and cabs (unlinked):
 Click and choose from the Amp Type and Cabinet Type selectors to create new pairings.
Use the Settings menu to save new combinations and build your own custom library
(see “Eleven Settings (Presets)” on
page 541).
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Eleven Cabinet Controls
All cabinets provide Bypass, Speaker Breakup, Mic Type, and Position controls.
Cabinet Bypass
Speaker Breakup
Mic Type
Off/On Axis
Cabinet controls
Cabinet Bypass
The Bypass switch in the Cabinet section lets
you bypass cabinet and microphone processing.
When in the Bypass position, no cabinet or microphone processing is applied to the signal.
When in the On position, cabinet and microphone settings are applied.
When enabled, Speaker Breakup draws additional CPU resources.
Mic Type
The Mic Type selector lets you choose the microphone to use with the selected cabinet.
Speaker Breakup
(Full version, TDM Only)
The Speaker Breakup slider lets you specify how
much distortion is produced by the current
speaker model. Increasing the Speaker Breakup
setting adds distortion that is a combination of
cone breakup and other types of speaker distortion (emulated by the speaker cabinet model).
Range is from 1 to 10.
Below certain frequencies, the speaker cone vibrates as one piece. Above those frequencies
(typically between 1 kHz and 4 kHz), the cone
vibrates in sections. By the time a wave travels
from the apex at the voice coil out to the edge of
the speaker cone, a new wave has started at the
voice coil. The result is comb filtering and other
anomalies that contribute to the texture of the
overall sound.
Mic Type selector in the Cabinet section
Available Mic Types include the following:
• Dynamic 7
• Dynamic 57
• Dynamic 409
• Dynamic 421
• Condenser 67
• Condenser 87
• Condenser 414
• Ribbon 121
Eleven is not affiliated with, or sponsored or
endorsed by, the makers of the microphones
that are emulated in the product.
Chapter 98: Using Eleven
547
Mic Axis
When capturing the sound of a speaker cabinet
in a studio, sound engineers select different microphones and arrange them in different placements to get a particular sound. For example, a
mic can be pointed straight at a speaker or angled slightly off-center, in order to emphasize
(or de-emphasize) certain frequencies that the
mic picks up.
Tracks and Signal Routing for
Guitar
The way you set up Pro Tools tracks and signal
routing can vary depending on what you want to
do while recording and mixing with Eleven. This
section gives you a few specific examples of
some of the many different ways you can choose
to work:
• “Recording Dry: Monitor Through Eleven”
on page 548.
On-axis, for most microphones, is a line in the
same direction as the long dimension of the microphone and will produce a noticeable difference in coloration when compared to the same
microphone in the off-axis position.
• “Recording Wet: Record Eleven-Processed
Track to Disk” on page 549.
• “Recording Dry and Eleven Simultaneously” on page 550.
In Eleven, the Axis switch lets you toggle between on- and off-axis setting of the currently
selected microphone model. The default position for Mic position is On Axis.
• “Processing Pre-Recorded Tracks Through
Eleven” on page 551
• “Blending Eleven Cabinets and Amps” on
page 552.
Recording Dry: Monitor Through
Eleven
Mic Axis switch in the Cabinet section
About Mic Placement
All Eleven cabinets and mics were close mic’d
(whether on- or off-axis). This provides the purest tones possible, of any room tone or ambience
specific to the Eleven recording environment.
This workflow lets you record dry (clean) while
the recorded signal is processed through Eleven,
letting you hear it but without committing the
track to that tone forever.
The flexibility to audition and compare different settings and combinations of amps, cabinets
and microphones is a very creative and powerful
tool for mixing and arranging.
To record dry and monitor through Eleven:
Choose Track > New and configure the New
Track to create one mono Audio Track.
1
2 Set the track input to the audio interface input
your guitar is plugged in to (such as In 1 (Mono)).
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Audio Plug-Ins Guide
Insert Eleven on the track (see “Inserting
Eleven on Tracks” on page 539).
3
Eleven
Guitar input
Recording Wet: Record ElevenProcessed Track to Disk
In this workflow, the audio output of Eleven is
recorded to disk while tracking. Usually, no additional dry track is recorded.
This method commits your track to the original
Eleven tone used while tracking. It requires two
tracks (an Auxiliary Input and an audio track),
but after tracking, the plug-in can be deactivated or removed to up processing resources.
To record guitar with Eleven while playing:
1
Choose Track > New.
2
Configure the New Tracks dialog as follows:
• Create one mono Auxiliary Input track.
• Click the Add Row button (+).
• Create one mono audio track.
• Click Create.
In the Mix (or Edit) window, configure the Aux
Input by doing the following:
3
Audio track for recording dry while hearing Eleven
Choose a Settings file (preset), or adjust
Eleven’s parameters to get your tone (see “Eleven
Settings (Presets)” on page 541).
4
• Click the Input selector and choose your
guitar input (the audio interface input your
guitar is plugged in to).
• Click the Output selector and choose Bus 1.
• Click the Insert selector and select Eleven.
5 Record enable the track, or enable TrackInput
monitoring (Pro Tools HD only) and check your
levels.
When you’re ready, arm the Pro Tools Transport and press Record to record your part.
6
The audio that is recorded is the dry (unprocessed) signal only, while playback of the recording is processed through Eleven and any
other plug-ins inserted on the track.
Chapter 98: Using Eleven
549
4 Configure the audio track by doing the following:
• Click the Input selector and choose Bus 1.
– and –
• Record enable the audio track.
Eleven
Recording Dry and Eleven
Simultaneously
You can record a dry, unprocessed track and an
Eleven-processed track simultaneously.
This method gets the best of both worlds by
tracking dry (to experiment with tones later)
and at the same time recording the tone used on
the original tracking session. It requires two audio track, as follows:
To record guitar dry and with Eleven live:
1
Guitar input
Bus output
Bus input
Choose Track > New.
Configure the New Tracks dialog to create two
mono audio tracks, then click Create.
2
In the Mix (or Edit) window, configure the
first (left-most) new audio track by doing the
following:
3
• Click the Input selector and choose your
guitar input (the audio interface input your
guitar is plugged in to).
• Click the Output selector and choose Bus 1.
• Click the Insert selector and select Eleven.
Aux Input
Audio Track
Recording Eleven (printing its output)
• Record enable the audio track.
Configure the second audio track by doing the
following:
4
5 Make sure you are not overloading your input
signal by checking levels in all tracks and
Eleven's Input LED.
When you’re ready, arm Pro Tools and begin
recording.
• Click the Input selector and choose Bus 1.
– and –
• Record enable the audio track.
6
The output from Eleven is recorded to disk. If
you need to conserve DSP or RTAS processing
resources, you can remove or deactivate Eleven
after recording.
5 Make sure you are not overloading your input
signal by checking levels in all tracks and
Eleven's Input LED.
When you’re ready, arm Pro Tools and begin
recording.
6
The dry guitar is recorded to the first audio
track, processed through Eleven, then bussed to
the second audio track and recorded to disk.
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Processing Pre-Recorded
Tracks Through Eleven
You can process pre-recorded guitar tracks, or
almost any material, through Eleven.
To listen to pre-recorded tracks through Eleven
(without re-recording):
To process and re-record tracks through Eleven:
Import and place your audio in a Pro Tools audio track.
1
2 Configure the source audio track by doing the
following:
• Assign the audio track Output a bus (such
as Bus 1 if mono, or Bus 1-2 if stereo).
Import and place your audio in a Pro Tools audio track.
1
– and –
• Click the Insert selector and select Eleven.
Assign the audio track Output to Bus 1 (or
Bus 1-2 if working with stereo material).
3
3 Create an Aux Input track, and configure it as
follows:
4
2
Choose Track > New and create one mono audio track.
Configure the new audio track as follows:
• Click its track Input selector and choose
Bus 1 (or Bus 1-2).
• Click its track Input selector and choose the
Bus 1 (or Bus 1-2).
– and –
– and –
• Click the Insert selector and select Eleven.
• Click the Insert selector and select Eleven.
Begin playback and watch Eleven’s Input LED
to check your level. Make sure you’re not clipping Eleven’s input.
4
Record enable the new audio track (or enable
TrackInput monitoring if using Pro Tools HD).
5
6
While listening, adjust Eleven’s Input knob to
increase or decrease input level.
7
6 After setting your gain structure, do any of the
following:
8
5
• Try different Settings files (presets) to get
your basic amp/cab/mic combination.
Begin playback and start listening.
While listening, adjust Eleven’s Input knob to
increase or decrease input level.
When everything sounds and looks good, locate to where you want to begin recording (or
make a time selection), arm the Pro Tools Transport and press Play to start recording.
• Adjust amp controls.
• Try different cabinets and varying amounts
of Speaker Breakup.
• Try different mics and positions to hear
how they affect the track.
Apply other plug-ins, or bus the Aux Input to
another track for additional processing.
7
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551
Blending Eleven Cabinets
and Amps
You can use Eleven for multi-cabinet and multiamp setups so you can blend their signals together. This classic technique lets you get tones
that no single combo, cabinet, or amp could produce. Unlike working with real amps, this is
simple to achieve with Pro Tools track, signal
routing, and plug-in features.
Blending Eleven Cabinets
4 Select all three Aux Input tracks by Shift-clicking their Track Name displays (make sure your
audio track isn’t still selected). This lets you
work with the three Aux tracks “as one” in the
next few steps.
Three tracks, selected
Hold Option+Shift (Mac) or Alt+Shift (Windows) while doing each of the following:
5
• Choose Bus 1 from the Input selector of any
of the three selected Aux Inputs.
In this example you’ll see how to take the output
of one Eleven amp and send it to multiple cabinets so you can blend different cabinets, multimic one cabinet, or both.
• Click the Insert selector of any of the three
and select Eleven.
• Click the next available Insert selector on
any of three selected Aux Inputs and select
the TimeAdjuster (short) plug-in.
To blend multiple cabinets:
1
Choose Tracks > New.
2
Configure the New Tracks dialog as follows:
• Create one mono Audio Track.
• Click the Add Row button.
• Create three mono Aux Inputs.
• Click Create.
In the Mix or Edit window, configure the audio
track by doing the following:
3
• Click the audio track Input selector and
choose your guitar input (the audio interface input your guitar is plugged in to).
• Click the Output selector and choose Bus 1.
• Click the Insert selector and select Eleven.
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Audio Plug-Ins Guide
Open the Eleven plug-in on the audio track
and click the Cabinet Bypass to bypass Cabinet
and microphone processing.
6
7 Open one of the Eleven plug-ins on any of the
three selected Aux Input tracks and
Opt+Shift+click (Mac) or Alt+Shift+click (Windows) the Amp Bypass switch.
8
Solo the first Aux Input track.
When you have set your cabinet tones, make
sure to un-solo all the Aux Inputs and begin
playing so you can hear the combined tone of all
three cabinet channels.
13
Amps bypassed/Cabs on
Amp on,
Cab bypassed
14
Do the following to continue:
• Balance the tracks using the volume faders
on the Aux Input tracks.
• Try different pan positions for each Aux Input track.
• Evaluate the phase relationships of the
combined signals and adjust accordingly
(see “Phase Considerations with Blending
in Eleven” on page 555).
If You Plan on Blending Cabinets
The Eleven plug-in emulates the variation in
cabinet response that is unique to each amp/cab
combination. In the physical world, these variations are the result of the distinct loads put out
by each amp, and the way the cabinet handles
(responds to) that particular type of signal.
Though subtle, the effect of this is a unique cabinet resonance.
Setup for blending cabinets
Click to open the Eleven plug-in window on
the first Aux Input, and do any of the following:
9
• Choose a cabinet.
• Choose a mic and its position.
• Adjust Speaker Breakup as desired.
When you’re done, close the plug-in window
and then unsolo the track.
10
11 Solo the next Aux Input track, and repeat to
configure its cabinet and mic settings.
Repeat for other Aux Input tracks to configure their cabinet and mic settings.
12
In each Eleven plug-in you insert on a track, the
currently selected Amp Type has a similar effect
on the sound of its current cabinet, even when
the amp section itself is bypassed.
This does not mean that the (bypassed) amp settings affect the cabinet tone, only the chosen
amp type. This could bring just the right amount
of extra low, low-mid, or mid-range response to
the cabinet.
Different amps can also have a different
number of stages, which can affect polarity.
See “Phase Considerations with Blending in
Eleven” on page 555 for more information.
Chapter 98: Using Eleven
553
How Do I Use This?
No insert
Here are a few suggested ways you can pair
Eleven’s amps and cabinets:
Amps and Cabs on
 To accurately capture the sound of one amp
split to and driving multiple cabinets, make sure
the same Amp Type is selected in all the Eleven
plug-ins (all the cabinets as well as the active
amp).
 For maximum variety, mix and match bypassed amps with active cabinets.
 For realism with the combo amps (such as the
Tweed Lux and AC Hi Boost), make sure to use
their default cabinets.
Blending Eleven Amps
You can easily set up tracks and Eleven for
multi-amp setups.
To blend multiple amps:
Set up tracks and signal routing as explained
in the previous workflow (see “To blend multiple
cabinets:” on page 552).
1
Remove (or simply bypass) the Eleven plug-in
on the source input/track.
2
Setup for blending amps
3
Solo the first Aux Input track.
Click to open the Eleven plug-in window on
the soloed Aux Input, and do any of the following:
4
To maximize processing resources, remove
the Eleven plug-in on the source track, or
make the plug-in Inactive. See the Pro Tools
Reference Guide for more information.
• Make sure the amp and cabinet are active
(not bypassed).
• Choose a preset (Settings file).
• Pair any amp with any cabinet.
• Choose a mic and its position.
• Adjust Speaker Breakup as desired.
Solo the next Aux Input track, and repeat to
configure its settings for a different tone.
5
6 Repeat for other Aux Input tracks to configure
their settings.
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When you have set your tones, make sure to
un-solo all the Aux Inputs.
7
Continue playing so you can hear the combined tone of all the amps.
8
9
Do the following to continue:
• Balance the tracks using the volume faders
on the Aux Input tracks.
– and –
• Try different pan positions for each Aux Input track.
Evaluate the phase relationships of the combined signals and adjust accordingly (see “Phase
Considerations with Blending in Eleven” on
page 555).
10
Phase Considerations with
Blending in Eleven
When multi-tracking guitar, experienced engineers know how to identify and take advantage
of the phase relationships that exist between different signals. Adjusting phase is not just a corrective technique either, it’s also a powerful creative technique for tone, as well as for special
effects.
You can use the TimeAdjuster plug-in to flip
phase and to adjust timing in single-sample increments, as described in the next sections.
Flipping Phase (Polarity)
Electric guitar is often recorded to more than
one track, such as one dry or DI track, plus one
or more tracks of a mic’d amp. The different signal paths of direct tracks versus mic tracks affect the timing relationships of the audio. Depending on the signal chain of each track, the
signals can get so out of alignment that they
nearly cancel each other out.
Sending a single source track through multiple,
unique amps can pose an additional challenge in
that each tube stage in an amp usually inverts
the signal. So, depending on whether the number of tube stages in an amp is odd or even, that
amp will either be inverting or non-inverting,
respectively. If you send an identical signal to
two amps and one is inverting while the other is
non-inverting, signal cancellation will result.
All amps in Eleven accurately model the number
of amp stages found in all the original hardware.
If you want to keep it simple and be able to experiment with phase flip, do the following.
To use the TimeAdjuster plug-in to flip phase when
blending amps or cabinets:
1 Configure your audio track and Aux Inputs as
instructed in “Blending Eleven Cabinets and
Amps” on page 552. Make sure each Aux Input
has an Eleven plug-in followed by a TimeAdjuster plug-in.
Open the plug-in window for each of the TimeAdjuster plug-ins (click the first one to open it,
then Shift-click each of the other TimeAdjuster
plug-ins).
2
Click the Phase switch in the first TimeAdjuster plug-in to invert the polarity. Listen to the
effect it has on the combined signal. Click it
again to disengage (flip back).
3
Click the Phase switch on the next channel’s
TimeAdjuster plug-in, listen, then disengage.
4
Repeat for additional Eleven/TimeAdjuster
channels.
5
Try combinations of flipped and non-flipped
Phase settings to find the ideal relationship for
the currently blended amps and cabinets.
6
Chapter 98: Using Eleven
555
Tweaking Phase
If each of the mics used on a single cabinet are
not positioned carefully, comb filtering and
other frequency anomalies can occur. With real
amps, the engineer moves one or more mics to
find their optimal positions relative to the
source, and to each other.
To hear the effect of small adjustments to the
phase relationships of signals, do the following.
To use the Time Adjuster plug-in to control phase:
1 Configure your audio track and Aux Inputs as
instructed in “Blending Eleven Cabinets and
Amps” on page 552. Make sure each Aux Input
has an Eleven plug-in followed by a TimeAdjuster (short) plug-in.
Open the plug-in window for each of the TimeAdjuster plug-ins (click the first one to open it,
then Shift-click each of the other TimeAdjuster
plug-ins).
2
Adjust the Delay slider in one sample increments. Listen to the effect it has on the combined signal. Repeat, increasing the Delay by one
sample each time.
3
Try combinations of TimeAdjuster settings
with flipped and non-flipped Phase settings for
endlessly variable tonal possibilities.
4
Eleven Tips and Suggestions
This section leaves you with some tips and suggestions for other ways you can integrate Eleven
into your sessions.
Changing Settings versus
Switching Amps
Many guitarists use different tones to maximize
the contrast between sections of a song (intro,
verse, chorus, or bridge). Some examples include:
• Soft (or clean) tone for the verse, kick in the
distortion for the chorus.
• Using tremolo during the intro and the
bridge.
• Doubling the rhythm track halfway through
the verse to build momentum.
Pro Tools automaton is the key to these and
other techniques:
 For simple, single amp contrasts such as
soft/loud, choose an amp and automate its gain,
drive, volume or other parameter to achieve the
desired tone change. This uses the least amount
of processing resources of the examples provided here.
 To switch amps, automate the Amp Type selector and any other controls (you cannot automate
the selection of Pro Tools plug-in Settings files).
Depending on the amount of overlap or crossfading you want between tones, you might be
better off using the next, multi-Eleven workflow.
See the Pro Tools Reference Guide to learn
about Snapshot automation, Glide, and
other automation features,
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Audio Plug-Ins Guide
 For maximum flexibility, control and variety,
use a dry track bussed to multiple Aux Inputs,
each with a different Eleven tone (see “Blending
Eleven Amps” on page 554 for instructions).
Configure one for tone A, configure the next
Eleven (on the next Aux Input) for tone B (which
could be a completely different amp and sound)
and so on. Then use Pro Tools track Volume
(fader) automation to fade the different Eleven
tracks in and out at the right times. This gives
the greatest amount of control over the transition between amps and tones, while also letting
you stack and layer amps.
Managing Eleven Plug-In
Resources
If system resources need to be conserved or minimized, you can “bus record” with effects to
commit Eleven tones to disk. See “Recording
Wet: Record Eleven-Processed Track to Disk” on
page 549.
Or, use the AudioSuite version to print Eleven
tracks to disk. AudioSuite is especially useful
when you’re processing loops or other shorterform guitar material.
Working Faster with Templates
and Import Session Data in
Eleven
Using templates and importing tracks are great
ways to make sure your creative moments aren’t
interrupted by session chores.
Pro Tools 8.0 provides the Quick Setup dialog
for working with templates. You can use this
feature to store and recall different setups of
tracks, bussing, and effects for Eleven.
In all versions of Pro Tools, the Import Session
Data feature lets you import tracks and their attributes from one session into another, including their I/O and signal routing assignments,
plug-ins, and settings.
See the Pro Tools Reference Guide for more
information about templates, Quick Setup,
and Import Session Data.
Beyond Eleven: Some
Suggested Effects
If you’re new to guitar or new to Pro Tools, you
might want to know about a few simple effects
you can add to your Eleven guitar tracks using
nothing more than a few of the plug-ins included with Pro Tools.
Bussing and Submixing
Not so much a plug-in or effect as a standard operating procedure, multiple guitar tracks are often submixed to stereo Aux Input for centralized
level control of those tracks. This is especially
useful for applying compression or limiting,
creating stem mixes, and many other practical
uses. See your Pro Tools Reference Guide for
mixing and submixing setups and suggestions,
and try them out while exploring some of the
following effects suggestions.
Chapter 98: Using Eleven
557
Dynamics
Compression, limiting, expansion and gating
are all useful effects for guitar. Different results
can be achieved using each of the different types
of dynamics processing, in combination with
signal routing for individual (discrete) versus
submix (shared resource) processing. Here are a
few examples:
 If all you seek is the taming of occasional dynamic aberrations within a track (meaning, you
just need to clamp a couple “overs”), try putting
a limiter on the individual track (after Eleven).
 To “glue” multiple rhythm tracks or tones together, bus them to a stereo Aux Input and apply
heavy compression or limiting to that Aux Input.
Experiment with different dynamics plug-ins
such as Dyn 3 or any of the Bomb Factory processors to find one that works best for the material. Don’t be afraid to use extreme compression
ratios to achieve this effect.
EQ
Simple EQ processing can be used to soften “hot
spots” in the playing range of some guitars. Using any of the included EQ plug-ins, you can also
try applying drastic shelving or band-limiting as
a special effect, or automate a filter sweep to
simulate a wah-style effect.
Echo and Delay
To add echo to the guitar track, bus an Eleven
track to an Aux Input and put a Delay plug-in on
the Aux. Try other delay plug-ins to unlock the
secrets of multi-tap, ping-pong, and other specialized applications.
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Audio Plug-Ins Guide
Eleven Signal Flow Notes
The following figure shows the signal flow through Eleven from its input source to its output destination.
Input
from Pro Tools
track (disk) or live input
Input LED
Input knob
Output
to Pro Tools
output or bus
Amp
Cabinet/Mic
Output knob
Gate
Signal flow through Eleven
Plug-Ins are Pre-Fader
Input Knob and Amp Gain
Keep in mind that inserts (plug-ins) in
Pro Tools are post-disk/live input but pre-fader.
The track fader does not affect the signal into
any plug-ins inserted on that same track. This is
the same for all Pro Tools inserts, not just
Eleven.
Eleven actually gives you two separate input
gain stages to the plug in:
 The Input knob in the Master section, which
affects the signal level before entering the amplifier model.
– and –
Input LED before the Input Knob
The Input LED is before the Input section of the
Master section, which is prior to the first input
stage of each amp. This lets you determine
whether you’re clipping a signal before it enters
the Eleven signal chain. The Input LEDs will
light red when the signal has clipped the input.
(If this occurs, insert the Trim plug-in before
Eleven and use its (Trim) gain control to attenuate the signal.)
 The gain knob(s) on each amplifier, which
control the main input stage of that particular
amplifier model.
This makes the Input knob useful for increasing
or decreasing gain on amps that don’t have a
separate preamp.
Noise Gate After the Input Knob
The Noise Gate is keyed (triggered) from the input signal. The gate is applied to the output of
the amp; when open it lets sound pass from the
amp to the cabinet module, and when closed silences amp output to the speaker cabinet.
Chapter 98: Using Eleven
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Part XV: Synchronic
Chapter 99: Synchronic Introduction
Synchronic is a loop processing and playback
plug-in that is available in RTAS and AudioSuite
formats. Synchronic is designed to manipulate
audio loops to create new and interesting rhythmic variations. Synchronic is the ideal recombinatorial rhythm machine for anyone who works
with audio loops.
Synchronic is essentially an instrument plug-in
that is most effective at manipulating (slicing,
dicing, and recombining) rhythmic audio loops.
After you load your loops into Synchronic, you
can control Synchronic using its on-screen interface, Pro Tools MIDI tracks, an external
MIDI controller, or Pro Tools plug-in automation.
Synchronic Features
• A DJ rig–inspired user interface for live
performance, including Sound, Playback,
Effect, and XFade presets and performance
parameters
• Control directly through its own plug-in interface, a MIDI controller, the computer
keyboard (Keyboard Focus mode), MIDI or
plug-in automation, or a Avid-qualified
Pro Tools control surface
• Support for 44.1 kHz, 48 kHz, 88.2 kHz,
and 96 kHz sessions
Synchronic plays back synchronized to the session tempo (including tempo changes) while
creating modifications in the playback order,
speed, and volume of individual beats and subdivisions of the beat (or “slices”) within a loop.
Synchronic also includes a multi-effects processor that synchronizes to the session tempo to
create in-tempo effects (such as filter sweeps
and delays).
Since Synchronic synchronizes to MIDI
Beat Clock, it only sounds during
Pro Tools playback.
Chapter 99: Synchronic Introduction
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Chapter 100: Synchronic Overview
This section provides an overview of Synchronic
features.
Synchronic Modules
Synchronic has a modular configuration for the
import, slicing, playback, and manipulation of
audio files (loops). Synchronic’s five modules
are: Sound, Playback, Effect, XFade, and MIDI.
Playback Module
Manipulates the output of the Sound module.
Various aspects of sound playback, such as
speed, order, and direction are controlled by the
Playback module.
Sound Module
Can load up to twelve different audio files, either mono or stereo, of any bit depth and sample
rate. After importing a file, it can be sliced up to
play in synchronization with the Pro Tools MIDI
Beat Clock. Any two sounds (A and B) can be
played back simultaneously.
Effect Module
Processes the output of the Playback module.
Four concurrent effects are available: Gain,
Noise, Filter, and Delay.
XFade Module (RTAS Only)
Mixes the A and B sounds after they have been
processed by the Sound, Playback, and Effect
modules. The crossfade between the A and B
sounds can be controlled either in Preset or
Manual mode.
MIDI Module (RTAS Only)
Synchronic, all modules in Performance mode (RTAS)
Lets you assign and trigger combinations of
sounds and presets using MIDI. You can also
map MIDI controllers to Synchronic controls.
Chapter 100: Synchronic Overview
565
Playing Synchronic RTAS
See “Using Synchronic as an AudioSuite
Plug-In” on page 603 for detailed information on playing the AudioSuite version of
Synchronic.
Synchronic RTAS does not play back the sound
(input) on a track as is the case with many plugins. Instead, Synchronic works as follows:
Loading a Loop in Synchronic
For detailed information on loading audio
files (loops) into Synchronic see “Importing
a Sound into Synchronic” on page 576.
To load a loop in Synchronic:
1
Insert Synchronic on an Instrument track.
Load audio files (loops) into Synchronic’s
Sound module, much like you would add sound
to a sampler.
1
Use the Detection Slider to slice up the loops
into rhythmically logical units (beats and subdivisions of the beat).
2
3
Play back the sliced-up loop in tempo.
Synchronic’s Playback module lets you manipulate playback of each slice. The RTAS version of
Synchronic also lets you add in-tempo effects
and mix between two different sounds (Sounds
A and B).
Synchronic starts and stops playback with the
Pro Tools Transport.
Synchronic Plug-In window (no audio loaded)
Configuring MIDI for Synchronic
(RTAS Only)
You can control Synchronic using MIDI (Instrument track data, an external MIDI controller, or
a MIDI control surface). You must first configure Pro Tools for MIDI.
See the Pro Tools Reference Guide
or Pro Tools Help for configuration
information.
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When Synchronic is inserted on a track, it
is the sole sound source for that track. If
Synchronic is inserted on an audio track
containing audio clips, those clips will
effectively be muted.
2 If necessary, switch the Sound module to Edit
mode. Click the Edit/Performance Mode toggle
to switch between Performance and Edit modes
(see “Performance and Edit Modes” on
page 569).
Edit/Performance Import
Mode toggle
button
Current
preset
Choose an audio file (loop) for import
4
Sound module, Edit mode
To import an audio file (mono, dual mono, or
stereo) into the Waveform display, do one of the
following:
3
• Drag an audio file from the Workspace to
the Waveform display.
Click Open.
The selected file is loaded into Synchronic and is
immediately stored with the current preset. The
waveform for the loaded file appears in the
Waveform display. When selecting more than
one loop, they are loaded into consecutive available presets.
– or –
• Use the Import button file to import an audio file into the current preset, and click
the Import button. In the Open dialog, navigate to and select one or more audio files
(loops) for import. Mono files are imported
as mono presets, and dual mono (.L and .R
files) and interleaved stereo files are imported as stereo presets.
Waveform display, mono file, before slicing
Shift-click to choose multiple contiguous
files. Control-click (Windows) or Command-click (Mac) to choose multiple noncontiguous files.
Chapter 100: Synchronic Overview
567
Slicing a Loop in Synchronic
Performing Synchronic
Once you have loaded a loop, it is ready to be
sliced up and can be played back in tempo with
the session.
After loading up to twelve loops into Synchronic
and slicing them up, you can “perform” those
loops using the Playback, Effects, and XFade
modules.
To slice a loop in Synchronic:
Adjust the Detection slider until you see the
desired number of slices in the loop. As you increase the detection percentage, slices will appear at detected attack transients in the
waveform.

slices
Detection slider
Loop loaded into Sound preset 1, sliced up
For detailed information on slicing up a
loop, see “Slicing Up a Sound in Synchronic”
on page 579.
Playing a Loop
To play a loop in the RTAS version of Synchronic:
 Click Play on the Pro Tools Transport, or press
the Spacebar.
To play a loop in the AudioSuite version of
Synchronic:

Click Preview in the Plug-In window.
Synchronic plays back the loaded loop according
to the session tempo.
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Audio Plug-Ins Guide
Slices can be played back in different orders, intempo effects can be added (such as Filter and
Delay), and different combinations of sounds
and effects can be crossfaded.
For detailed information on the Playback,
Effect, and XFade modules, see
Chapter 101, “Using Synchronic.”
The flexibility of Synchronic’s playback and effects possibilities, along with its DJ-rig inspired
interface, invite real-time performance. You can
play it live using the on-screen interface, a MIDI
controller, your computer keyboard, or a control surface; or you can control it using MIDI
data on a Pro Tools Instrument or MIDI track,
or using Pro Tools plug-in automation.
For information on automating Synchronic
see Chapter 103, “Automating Synchronic
RTAS.”
Performance and Edit Modes
Each of the Synchronic modules can be independently switched between Edit and Performance
modes with the Mode toggle. This lets you have
one module in Edit mode (for example, to fine
tune a sound) while playing another in Performance mode.
The AudioSuite version of Synchronic functions only in Edit mode. Performance mode
is not available, and the Mode toggle does
not exist. See “Synchronic AudioSuite Modules” on page 603 for more information.
To toggle a module between Performance and Edit
mode:
 Click the Mode toggle to the right of the module’s name.
Mode toggle (Sound module)
When the triangle in the Mode toggle points to
the right, it indicates that the module is in Performance mode. When the triangle points down,
it indicates that the module is in Edit mode.
Edit Mode
Edit mode provides detailed access to the module’s controls. Changing parameters in Edit
mode immediately alters the currently selected
preset. Edit mode lets you load audio files, edit
presets, and make MIDI assignments that can be
instantly recalled in Performance mode.
Detailed information about each module’s
Edit parameters is described in
Chapter 101, “Using Synchronic”
Synchronic Performance
Controls
(RTAS Only)
Synchronic provide several types of controls for
real-time performance, including presets, userassignable performance controls (User Knobs),
the Sound A and B selectors, and an on-screen
keyboard.
Presets
Store and recall the Edit mode settings for each
module. Synchronic provides twelve presets for
each module. For more information, see “Synchronic Presets” on page 570.
Performance Mode
In the Synchronic RTAS, each module can be
viewed in Performance mode to provide presets
and performance controls. Performance mode
lets you select sounds loaded into the Sound
module, select presets and manipulate performance controls for the Playback, Effect, XFade,
and MIDI modules.
Presets, Playback module
Detailed information about each module’s
Performance mode controls is presented in
Chapter 101, “Using Synchronic.”
Chapter 100: Synchronic Overview
569
User Knobs
On-Screen Keyboard
Control predefined parameters in the Playback,
Effect, and XFade modules. In Edit mode, you
can assign a module’s User Knobs to control certain performance parameters. The MIDI module
lets you assign MIDI controllers to each of the
User Knobs in each module.
Can be used to trigger presets in the other four
modules. You can also assign the on-screen keyboard, or an external MIDI keyboard, to trigger
different presets. For more information, see
“Synchronic MIDI Module Overview” on
page 598.
Keyboard, MIDI module
User Knobs, Playback module
Synchronic Presets
Sound A and B Selectors
Each module has 12 presets available, as follows:
Are available both in Edit and Performance
mode for the Playback and Effect modules.
These buttons toggle which sound—Sound A or
Sound B—the current Playback or Effect preset
is processing. Both Sound A and B selectors can
be enabled at the same time, allowing for a single Playback or Effect preset to modify both
Sound A and Sound B at the same time. When
neither Sound A nor Sound B selectors are enabled, the currently selected Playback or Effect
preset is disabled until one of the Sound selectors is enabled.
Sound A/B selectors, Playback module
• Sound module presets are used to load and
store audio files (loops).
• Playback, Effects, and XFade module presets are used to store and recall various Edit
mode settings.
• MIDI module presets store MIDI control
assignments (which can include combinations of presets from each of the other modules).
Plug-In settings and presets can be shared
between the AudioSuite and RTAS versions
of Synchronic. However, the AudioSuite version of the plug-in can import and export
only information stored for the Sound, Playback, and Effect modules in the first preset.
Synchronic presets are unique to each instance
(each insert) of Synchronic in a session. To save
the global state of Synchronic presets in a given
instance, use the plug-in librarian (see
Chapter 104, “Synchronic Plug-In Settings”).
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Synchronic presets can be triggered by the MIDI
module in Performance mode, or a Pro Tools Instrument or MIDI track, or an external MIDI
keyboard controller for performance situations
(live or in the studio).
For information about assigning keys and
MIDI controllers in the MIDI module, see
“Synchronic MIDI Module Overview” on
page 598.
Presets can also be selected using plug-in automation or even your computer keyboard.
In Keyboard Focus mode, you can select presets using your computer keyboard (see
“Synchronic Keyboard Focus Mode” on
page 601).
To edit and store a preset:
Select the preset you want to edit. For the
Sound module, you must be in Edit mode to select the preset you want to edit (see “Synchronic
Sound Module Overview” on page 574).
1
If the module is in Performance mode, toggle
to Edit mode and edit as desired (for detailed information on the Edit mode parameters of each
module, see Chapter 101, “Using Synchronic”).
2
Any edits to any module’s parameters are immediately applied and stored in the selected preset.
To duplicate a Playback, Effect, or XFade preset:
1
Select the preset you want to duplicate.
Control-click (Windows) or Command-click
(Mac) the destination preset.
2
To select a preset, do one of the following:
In Performance mode, click a Preset button in
any module. MIDI module presets are the “keys”
of the on-screen keyboard.

The settings from the selected preset will be copied to the destination preset.
– or –
 In Edit mode, click a Preset button in the
Sound module, or select a preset from the Preset
pop-up menu for the Playback, Effects, and
XFade modules. In Edit mode, the MIDI module’s presets are not available on-screen.
Preset pop-up menu, Playback module
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Chapter 101: Using Synchronic
This section covers Synchronic controls and
functions.
To type a parameter value:
Adjusting Synchronic
Parameters
2
Click on the parameter text that you want to
edit.
1
• Type the desired value.
– or –
You can adjust Synchronic parameters with a
mouse or a computer keyboard.
• To increase a value, press the Up Arrow on
your keyboard. To decrease a value, press
the Down Arrow on your keyboard.
Editing Parameters Using a Mouse
You can adjust rotary controls with a mouse by
dragging horizontally or vertically. Parameter
values increase as you drag upward or to the
right, and decrease as you drag downward or to
the left.
Editing Parameters Using a Computer
Keyboard
Each rotary control has a corresponding parameter text field directly below it. This displays the
current value of the parameter. You can edit the
numeric value of a parameter with your computer keyboard.
Change the value.
3
Do one of the following:
• Press Enter on the numeric keyboard to input the value and remain in keyboard editing mode.
– or –
• Press Enter on the alpha keyboard (Windows) or Return (Mac) to enter the value
and leave keyboard editing mode.
To move forward through the different parameters, press the Tab key. To move backward, press Shift+Tab.
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Synchronic Sound Module
Overview
Synchronic Sound
Performance Mode
The Sound module can store and recall up to
twelve audio files. On import, audio files are
loaded into RAM and are saved with the plug-in
settings file. You can toggle between Sound Performance mode and Sound Edit mode.
(RTAS Only)
In Sound Performance mode (see “Synchronic
Sound Performance Mode” on page 574) the
Sound module can play back any of its twelve
presets independently on both the A and B channels (Sound A and Sound B).
In Sound Edit mode (see “Synchronic Sound
Edit Mode” on page 575), you can import, delete, and “slice” audio files (loops). When Synchronic “slices up” an audio file, it automatically edits the file (loop) into clips—according
to its own sophisticated internal transient detection algorithms—based on musical criteria,
such as meter, number of bars, and subdivisions
of the beat. In this way, Synchronic can quickly
and easily isolate individual hits in rhythmic
loops. Each slice is played back in tempo by synchronizing to the Pro Tools MIDI Beat Clock.
Performance mode lets you recall and interact
with different audio files and loops loaded into
Synchronic.
Performance mode applies only to the RTAS
version of Synchronic.
Sound Presets
Sound presets recall audio files loaded in Synchronic. In Performance mode, Sound presets
are numbered from 1–12 and are arranged in
two banks located above and below the Waveform display. Each Sound preset may contain an
individual mono or stereo audio file of any bit
depth or sample rate. Selecting a Sound preset
will instantly recall the audio file (loop) for
playback. You can select a Sound preset by clicking the desired preset button, or by pressing the
corresponding key on your MIDI keyboard.
Performance/Edit
Mode toggle
Sound A Waveform
display (mono file
loaded)
Sound B Waveform
display (stereo file
loaded)
Sound B
presets
Interactive Waveform display
Sound A
presets (Sounds A and B loaded and sliced
Sound module, Performance mode, with audio files loaded into Sound A preset 4 and Sound B preset 1 (RTAS
shown)
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Audio Plug-Ins Guide
Sound A and Sound B
Starting and Stopping Playback
In Performance mode, two sounds can be
queued for playback, much like a typical DJ
setup with two turntables: one on the “A Side”
and one on the “B Side.” Both the Sound A and B
use the same twelve sound presets. The waveform for Sound A appears above the waveform
for Sound B; and each waveform is labeled on
the left “A” and “B.”
You can only start or stop Synchronic playback
using the Pro Tools Transport. Any Synchronic
insert will play back when the Pro Tools Transport is engaged.
When Sound A and B are playing simultaneously, you can isolate one of the sounds by clicking the A or B labels to the left of the Waveform
display. The Crossfade fader in the XFade module will move to one side or the other, effectively
muting the non-selected sound.
Interactive Waveform Display
In Performance mode, the Sound module displays two waveforms for the A and B sounds.
The Waveform display shows the currently selected audio file (loop). The Waveform display
can show both mono and stereo audio loops for
the A and B sides.
During playback, the Waveform animates to indicate the pulse of the Pro Tools MIDI Beat
Clock. Clicking on the Sound module’s Waveform display during playback will reposition
playback to that location in the waveform. You
can create syncopated rhythms by repeatedly
clicking the waveform in either the A Sound or
the B Sound during playback. Alt-clicking (Windows) or Control-clicking (Mac) the Waveform
display re-synchronizes playback with the current Pro Tools MIDI Beat Clock location (according to bars and beats).
Use Track Mute to manage multiple Synchronic inserts. You can also use the Synchronic Plug-In Bypass to mute the output
of a Synchronic insert.
Synchronic Sound Edit Mode
In Edit mode, sounds can be imported, deleted,
sliced-up, and fine-tuned for Synchronic playback. Audio files loaded into Synchronic’s presets are unique to each insert and are saved with
the Pro Tools session file or plug-in settings file.
The global state of all presets, including loaded
audio files, can be stored and recalled using the
Synchronic plug-in librarian (see Chapter 104,
“Synchronic Plug-In Settings”).
When the Sound module is in Edit mode in
Synchronic RTAS, only Sound A is visible
and Sound B is muted. Sound A is effectively
soloed (the XFade module only passes
Sound A).
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575
Performance/Edit
Mode toggle
Selected
preset
Sound
presets
Interactive Waveform
display (stereo file
loaded and sliced)
Sound Attributes
Import Sound Delete Sound
button
button
Slice Detection slider
Sound module, Edit mode (audio loaded into preset 1) (RTAS shown)
Preparing Audio Files for Import
into Synchronic
To take full advantage of Synchronic’s rhythmic
editing and playback capabilities, you should
prepare your “loops” before importing them
into Synchronic. You can do this in Pro Tools by
editing a clip (loop) on a track in the Edit window and then consolidating the clip. Trim the
clip (loop) to an exact bar length. There should
be no gap between the start of the clip and the
downbeat, and no additional audio at the end of
the clip. Once you have defined your loop as a
clip, consolidate the clip (Edit > Consolidate Selection) and import the resulting audio file into
Synchronic.
In preparing your loops in the Pro Tools
Edit window, use Tab To Transients to locate downbeats and use the Separate Clip
command (Edit > Separate) to create “loopable” clips from longer clips.
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Audio Plug-Ins Guide
For more information about editing clips in
Pro Tools, see the Pro Tools Reference
Guide.
Importing a Sound into
Synchronic
You can import one or more audio files into Synchronic by using the Import button or dragging
and dropping from the Workspace.
Supported Audio Formats
Synchronic supports AIFF, BWF (WAV), and
SD II (Mac only) audio file formats, and 8-, 16-,
and 24-bit mono or stereo audio files. Any combination of supported bit rates and audio file
formats can be imported and played back at the
same time. All audio files are converted to 32-bit
floating point (RTAS native format) on import.
However, Synchronic does not convert the sample rate of files on import. For example, if you
import a 44.1 kHz file into a 96 kHz session, it
will playback at the wrong pitch (it will play
back at tempo since each slice is quantized to
MIDI Beat Clock).
To import a sound into a preset:
1 If necessary, switch the Sound module to Edit
mode (see “Performance and Edit Modes” on
page 569).
Select the preset where you want to store the
audio file (loop).
2
Click the Import button. If there is already a
file loaded into the current preset, you will be
prompted to delete it.
3
Control-click (Windows) or Command-click
(Mac) Import button to bypass this prompt.
4 In the Open dialog, navigate to and select one
or more audio files (loops) for import. Mono
files are imported as mono presets, and dual
mono (.L and .R files) and interleaved stereo
files are imported as stereo presets.
Shift-click to choose multiple contiguous
files. Control-click (Windows) or Command-click (Mac) to choose multiple noncontiguous files.
5
Drag the selected audio to the Waveform display in Synchronic.
4
Dragging and dropping an audio file works
only from the Workspace, and not from Windows Explorer, Mac Finder, or the Edit window.
Selecting Multiple Files for
Import into Synchronic
When multiple files are selected, they are loading into the next available unoccupied presets.
For example, if preset 2 is the selected preset,
and preset 3 and 4 already have sounds loaded,
importing three sound files will load them into
presets 2, 5, and 6.
Additionally, all matching .L and .R files are imported as a stereo Sound preset. For example, selecting Happy.L, Happy.R, Kyne.L and Kyne.R
will load two stereo sounds into the selected preset and the next available preset.
After Loading Sound into Synchronic
The selected file will be loaded into Synchronic
and stored with the preset. The waveform for the
loaded file will appear in the Waveform display.
In Performance mode, the same preset can be selected for both Sound A and Sound B.
Click Open.
To drag and drop audio files into Synchronic:
Make sure the Sound module is set to Edit
mode. (see “Performance and Edit Modes” on
page 569).
1
Select the preset where you want to store the
audio file (loop).
2
Waveform display, stereo file, before slicing
Open the Workspace, and navigate to and select one or more audio files (loops) for import.
Mono files are imported as mono presets, and
dual mono (.L and .R files) and interleaved stereo files are imported as stereo presets.
3
Chapter 101: Using Synchronic
577
Mono files loaded in a stereo Synchronic insert
are panned center. Stereo files loaded in a mono
Synchronic insert are summed to mono. Dual
mono audio files can be imported if both .L and
.R files are selected (Shift-click) for import. Interleaved stereo audio files can also be imported.
If you try to import a large sound file, Synchronic prompts you to reconsider. Large
files are hard to view in the Waveform display, and will be slow to load, so you may
want use Pro Tools to edit larger files into
multiple smaller files. You can then import
the smaller files into Synchronic.
How Synchronic Stores Sound Presets
Synchronic loads audio files into RAM on import and then stores them with the plug-in settings. The plug-in settings are stored with the
Pro Tools session file or using the Settings Librarian to save a plug-in settings file (.tfx). The
size of the Pro Tools session file or plug-in settings file will increase corresponding to the
number and size of audio files loaded into Synchronic.
Importing Acid Files
Synchronic can import audio in the Acid wave
file format. Synchronic will reveal the pre-existing slice data as you adjust the Detection slider
(see “Slicing Up a Sound in Synchronic” on
page 579).
Entering Sound Attributes in
Synchronic
After importing an audio file (loop), you need to
enter additional information regarding the following attributes: Length, Time Signature, and
Subdivision of the beat. When accurately defined, these attributes help Synchronic properly
play back the loop in synchronization with the
Pro Tools MIDI Beat Clock (assuming you have
not applied too many of Synchronic’s beat
scrambling processing capabilities). This way
you can play back a loop at nearly any tempo
with reasonable accuracy.
Sound attributes
When you close and later open a session with
Synchronic inserted and files loaded into it,
Pro Tools will re-load any stored audio files into
RAM when opening the session.
Depending on the number and size of the
audio files loaded into Synchronic, plug-in
settings file sizes will vary in size. Typically,
you will only load audio files that are a few
bars long. If you import large files into Synchronic, saving and restoring settings files
will take more time.
Name Displays the name of the audio file for the
currently loaded audio loop. The name displayed in File Name will be the same name as the
audio file as it appears on the hard drive or storage medium from which the file was loaded.
Length Lets you enter the number of bars for the
currently selected loop.
Time Signature Lets you enter the time signature
for the currently selected loop.
Contains Lets you select whether the current au-
dio loop contains eighth, sixteenth, or thirtysecond note subdivisions of the beat, and
whether the loop contains a triplet subdivision.
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Audio Plug-Ins Guide
Deleting a Sound from a
Synchronc Preset
To delete a sound from a preset:
Make sure the Sound module is set to Edit
mode. (see “Performance and Edit Modes” on
page 569).
1
2
Select the preset you want to clear.
3
Click the Delete button.
4
You will be prompted to confirm deletion.
Control-click (Windows) or Command-click
(Mac) the Delete button to bypass this
prompt.
The selected preset will be empty, and the sound
will be deleted from the current settings file.
Slicing Up a Sound in
Synchronic
Once you have imported and assigned attributes
to an audio file, the imported sound needs to be
“sliced up.” A slice is like an audio clip (internal
to Synchronic only) that typically contains only
a single hit (articulated by a clear attach transient). You can adjust the Detection slider to automatically slice up the sound file based on attack transients. Each slice is indicated with a
distinct line in the waveform at the start of the
slice, and the slice number. In most cases, you
can quickly identify the musically significant
rhythmic events in a sound file by adjusting the
Detection slider.
Once a loop is sliced up, Synchronic can replay
the loop at any tempo with any number of modifications. Furthermore, each slice of the loop
starts playback according to the Pro Tools MIDI
Beat Clock, so it maintains its original rhythmic
pattern at any tempo.
To add a missing slice:
 Control-click (Windows) or Command-click
(Mac) the Waveform display at the correct point
in the waveform to create a new slice.
To delete an erroneous slice:
 Alt-click (Windows) or Option-click (Mac) the
slice.
All sound files must contain at least one
slice. consequently, the first slice of a loop
can not be deleted.
Synchronic Slice Settings
Slice settings let you adjust the sensitivity for
slice detection, generate missing slices, and trim
or delete the current slice. This section also provides controls for auditioning the loop or individual slices.
Slice settings
Detection (0–100%) Automatically slices up the
loaded audio file based on transient detection.
The higher the detection value, the more slices
are identified.
Waveform display, stereo file, sliced
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579
Generate Missing Adds additional slices where
Current Displays the currently selected Slice for
transients appear to be missing. This can be useful when an imported audio loop contains few
transients.
auditioning, trimming, or deleting. Click on a
Slice in the Waveform display to update the Current slice. A Slice number can also be manually
entered in the Current field.
Use Generate Missing to create rhythmically
logical slices on ambient loops or drones,
and apply playback manipulations and effects for interesting results.
Audition Modes Are initiated using the
Pro Tools transport. The default audition mode
is Off. To audition a single slice without starting
the Pro Tools transport, click the slice in the
Waveform display. Synchronic provides the following Audition modes:
• Off: Plays back each slice in a sound synchronized to the Pro Tools MIDI Beat
Clock.
• Original: Loops the audio file at its original
tempo.
• Half Speed: Plays the audio file at half the
original tempo and pitch. This is useful for
listening to slices closely to hear whether or
not a slice has been correctly detected,
• Single Slice: Loops a single slice. This is
useful for adjusting the slice end point.
• Double Slice: Loops two consecutive slices.
This is useful for detecting when a false
slice is present and needs to be deleted.
Use Single Slice mode to edit slices in hard to
view waveforms, such as in heavily compressed or extra long loops.
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Audio Plug-Ins Guide
In Keyboard Focus mode, use the Left and
Right arrow keys to increment/decrement
the current slice number.
Delete Deletes the currently selected Slice. Option-click can also be used to delete erroneous
slices.
Shift-clicking on a slice will play two consecutive slices. If the second slice played
turns out to erroneous, that is only one true
percussive event is heard when shift-clicking a pair of slices, then you may want to
delete the second slice of the pair. (Be sure
to select the correct slice to delete!)
Trim (–.30 to .30 sec.) Trims the currently selected Slice’s end point. Slices generated during
detection (or by Command-clicking) may need
to be adjusted slightly. For example, if you audition a slice (by clicking on it), and you hear a
click at the end of the slice you will need to trim
the end of the slice. The Trim slider can adjust
the end of any slice (except the last one) by
+/–30 milliseconds. When in Single Slice Audition mode, the Waveform Display zooms to the
currently selected slice, letting you visually adjusted the slice end using the Trim slider.
Certain operations (such as Generate Missing and Delete Slice) can change the number
of slices in a Sound preset. Changing the
number of slices during playback may cause
the playback position to jump, and cause
Synchronic to play back out of synchronization. Stopping and starting playback will
re-synchronize Synchronic playback.
Synchronic Slice Options
Additional Sound Options let you adjust the
overall gain (+/–24 dB) of the selected preset
and extend the slices to compensate for slower
tempos.
Options settings
Gain (–24 through 24 dB) Adjust the gain of a
sound in order to boost or attenuate the current
loop. To achieve a good balance between all
loaded sounds, a “gain” control is available for
each sound with +/– 24 dB of gain.
Slice Extension (Off, Type 1, Type 2) Since a
sliced up sound can be played back at a slower
tempo than its original performance, each slice
may need to play for a longer duration than the
original audio file length. Otherwise, a slice will
stop when it reaches its end point, and sound is
unnaturally truncated. Slice Extension lets you
designate if and how slices can be artificially extended. Synchronic provides the following Slice
Extension types:
• Off: Adds no tail extension, however it does
ramp down any DC offset that may occur at
the last sample of a slice.
• Type 1: Sounds best on loops that predominantly contain high frequencies, such as a
tambourine or a hi-hat loop.
Synchronic Playback Module
Overview
The Playback module determines how the
sounds loaded into the Sound module presets
are played back. Each sound slice in the selected
Sound module preset is played back synchronized to the Pro Tools MIDI Beat Clock, and the
the Playback module determines the order, duration, direction, starting slice, and mode by
which each slice is played back.The Playback
module additionally provides controls for the
playback pitch and offset of the selected sound
presets.
You can toggle between Playback Performance
mode (see “Synchronic Playback Performance
Mode” on page 581) and Playback Edit mode
(see “Synchronic Playback Edit Mode” on
page 582)
Synchronic Playback
Performance Mode
(RTAS Only)
In Performance mode, the Playback module provides Preset, A/B Sound selectors, and User
Knob controls.
Sound A/B
selectors
Performance/Edit
Mode toggle
Presets
• Type 2: Sounds most natural for most loops
and is the default setting.
User Knobs
(PB1 and PB2)
Playback module, Performance mode
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581
Sound A and B Selectors
The Sound A and B selectors determine whether
or not the A and B sounds are patched into the
Playback module. Click the Sound A or B selectors to toggle the A or B sound on or off.
Synchronic Playback Edit
Mode
In Edit mode, you can edit the Playback parameters for the selected preset.
Playback Presets
Sound A/B
selectors
Twelve Playback presets let you recall stored
playback effects.
Pitch and Offset
Select and Enable
buttons
Selecting a preset recalls the last edited set of
parameters for the preset. Playback presets let
you invoke different playback manipulations in
rapid succession, to create a compelling musical
performance. Presets can also be used to store
sound design variations for use in a Pro Tools
session.
Pitch or Offset
parameters
Playback module, Edit mode
Playback User Knobs (PB1 and PB2)
Synchronic Playback Modes
The Playback module provides two assignable
User Knob controls (PB1 and PB2) that provide
direct control over any of the Playback module’s
Edit mode settings. The current control assignment is displayed below each User Knob.
The Playback module provides five playback
modes: Standard, Stretch, Stab, Spin, and
Smear. These modes determine the character of
each slice as it is played—how each slice is
played back.
Preset selector
Performance/Edit
Mode toggle
User Knob
assignments
Selecting Playback Mode
Standard Plays each slice without any additional
manipulation and plays each slice from start to
end.
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Audio Plug-Ins Guide
Stretch Uses a granular time compression/expansion technique to make a slice “fit” the
amount of time the slice has to play. The Playback Offset settings can be used to further increase the stretch factor for extreme granulation
effects (see “Synchronic Playback Offset” on
page 588).
Stab Based upon the “stab” scratching gesture
that turntablists use when manually starting
and stopping the turntable with their hand. The
playback of each slice always starts at a speed
and pitch of zero, and then ramps up to full
speed and correct pitch halfway through the
slice, and then returns to zero speed and pitch
by the end of the slice. The “full” speed and “correct” pitch are determined by the Playback Pitch
settings (see “Synchronic Playback Pitch” on
page 587).
Spin Lets each slice loop asynchronously during
the playback duration of the slice. If Pitch and
Offset manipulations are used, a slice may repeat itself a multiple times before advancing to
the next slice.
Synchronic Playback Start
Start determines where to start playback within
a sound, either Clocked to the MIDI Beat Clock
or from a specific slice.
Clocked Sets the start position relative to the
current position of the MIDI Beat Clock. For example, if Pro Tools starts playing back at beat 2
of bar 13, and Synchronic is playing back a two
bar drum loop, then Synchronic will start playback at the slice corresponding to the second
beat of the first bar of the loop.
Slice # Sets playback to start at a specific slice
regardless of the position of the MIDI Beat
Clock. When Slice # is selected for Playback
Start, enter the number of the slice you want to
start with in the numeric enter field to the right
of the Start pop-up menu.
Playback Start setting, Slice # 7 selected
Smear Uses crossfading and reverse playback to
“smear” one slice into the next. The Playback
Offset settings control the depth of smearing
(see “Synchronic Playback Offset” on page 588).
If the playback tempo is slower than the
original tempo such that it creates a gap between the end of one slice and the beginning
of the next, Synchronic will extend the first
slice according to the setting of the Slice Extension Option in the Sound module (see
“Synchronic Slice Options” on page 581).
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583
Synchronic Playback Order
Order determines the sequence in which the
slices of the selected sound are played back. Synchronic provides several options for Playback
Order: Off, One Slice, Reverse, Diverge, Random,
Step, Span, Straddle.
Diverge Plays the Start slice followed by last
slice, followed by the second slice, followed by
the second to last slice, and so on following a divergent order from the Start slice.
Random Plays slices in random order.
Random Swap Randomly selects “sibling” slices
based on a half-note spacing. For example, the
“2 and” of a bar, and the “4 and” of a bar are siblings. Unlike Random, which can sound very
“outside,” Random Swap preserves much of the
original feel of a loop, while simultaneously
adding random variations with each repeat of
the loop.
Step (Step 2–5) Plays slices by stepping over
some number of adjacent slices (2–5). For example, selecting Step 2 with a loop containing eight
slices will play back slices 1, 3, 5, 7, 2, 4, 6, 8, and
repeat.
Span (Span 2–5) Plays slices by stepping forward
some number of slices (2–5) then skipping back
to play the stepped over slices in order. For example, selecting Span 3 with a loop containing
eight slices would play slices 1, 4, 3, 2, 5, 4, 3, 6,
5, 4, 7, 6, 5, 8, 7, 6, and so on.
Selecting Playback Order
Off Plays the sound slices in their original se-
quence. Each slice increments from the previous
one until the end of the sound is reached, and
then continues to loop from the starting slice.
One Slice Plays back and loops only the selected
Start slice.
Reverse Plays the original sequence of slices in
reverse order. Each slice plays in the specified
direction (see “Synchronic Playback Direction”
on page 586), but order of slices is reversed.
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Audio Plug-Ins Guide
Straddle (Straddle 2–5) Plays slices by stepping
forward then making a small step backward. For
example, selecting Straddle 4 with a loop containing eight slices would play slices 1, 5, 2, 6, 3,
7, 4, 8, and so on.
Depending on where you start playback in
the Pro Tools timeline, the start position
will affect the exact sequence of slices when
using Step, Span, or Straddle. Typically,
you will want to be sure to start playback on
the downbeat of a bar, and that the aligns
with the number of bars in the loop.
Synchronic Playback Duration
Playback Duration determines how long a slice
will play before the next slice starts. In addition
to the predominant pulse (beat or subdivision of
the beat), rhythmic loops usually contain a variety of durations. Thus, to maintain the character
of a rhythm at different tempos, the default
value for Duration is for the duration of the
slice. This means that each slice plays for its
original note length (rhythmic value).
Synchronic also lets you override the original
duration of a slice and impose alternate rhythmic patterns on the sliced-up loop. In addition
to the standard note-value durations of eighth
notes, eighth note triplets, and sixteenth notes,
the following groups of rhythmic patterns are
included: Off Beats, Syncopate, Clave, Pick Up,
and Swing.
Off Beats (Off Beats 1–5) Five variations that
generally emphasize the “and” of the beat.
Syncopate (Syncopate 1–5) Five variations fea-
turing dotted rhythms.
Selecting Playback Duration
Chapter 101: Using Synchronic
585
Latin (Latin 1–5) Five variations based on AfroCaribbean, Cuban, and Brazilian rhythms.
Swing (Swing 1–5) Five variations that use incorporate eighth-note swing (Swing 1–2) or sixteenth-note swing (Swing 3–5.)
Pick Up (Pick Up 1–5) Five variations that start
steady and then speed up over the course of a
bar.
Synchronic Playback Direction
Synchronic can play sound slices forward (from
beginning to end) or backward (from end to beginning). Synchronic provides seven possibilities for the direction of the playback of slices:
Forward, Backward, For/Back, Back/For, F/B
Diddle, F/B Beats, and Random.
Selecting Playback Direction
Forward Plays all slices forward (from beginning to end).
Backward Plays all slices backward (from end to
beginning).
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For/Back Plays alternating slices forward then
backward.
Back/For Plays alternating slices backward then
forward.
F/B Diddle Plays consecutive slices forward,
backward, forward, forward, backward, forward, backward, backward, and so on.
F/B Beats Plays slices on downbeats (quarter
notes) forward, and all other slices are played
backward.
Random Plays back slices forward or backward
randomly.
Synchronic Playback Pitch
Pitch transposition can range from –60 to +60
semitones. Synchronic provides two pitch modulating LFOs for Playback Pitch. These LFOs can
be combined to create complex and interesting
modulation patterns.
Low Frequency Oscillator (LFO)
Playback Pitch and Playback Offset, and Gain,
Noise, Filter, and Delay effects can be continuously modulated by one or more Low Frequency
Oscillators (LFO1 and LFO2). Each LFO has a selectable Waveshape and Duration. The LFO will
sweep through the selected waveshape according to the selected duration.
If an LFO Waveshape is set to Off, that LFO does
not modulate the associated parameter.
If both LFO waveshapes are set to Off, then only
the BEG parameter will affect the playback
pitch. The END parameter will have no affect on
playback pitch.
LFO Waveshapes Select the modulation wave
shape for LFO1 and LFO2. The available wave
shapes include: Off, Sawtooth, Triangle, Vee,
Square, Short Pulse, Long Pulse, Sawtangle,
Staircase, and Random.
Pitch Enable
button
Pitch Select
button
Beginning Pitch
(in semitones)
End Pitch
(in semitones)
LFO Waveshap
selectors
LFO Duration
selectors
LFO Waveshape pop-up menu
Playback Pitch settings
Enable (On, Off) Enables or disables Pitch mod-
ulation.
BEG and END Range (–60 to +60) Sets the Begin-
ning transposition and End transposition values
(in semitones) referenced by both LFOs.
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The LFO will sweep from the BEG parameter
value to the END parameter value according to
the selected waveshape. It is important to note
that the LFO waveshapes are iconic representations of the type of transitions you can select,
and, depending on the BEG and END values,
they may not necessarily sound like they look.
LFO2 Duration (8 Bars, 4 Bars, 2 Bars, 1 Bar, Half
Note, Quarter Note) Applies the LFO according
to the selected duration.
LFO1 Duration (Slice, 8th, 16th, 32nd, Off Beats,
Syncopate, Latin, Pick Up, Swing) Applies the
LFO according to the selected duration. The LFO
will sweep through the selected waveshape according to the selected rhythm. For more information on the rhythmic patterns, see “Synchronic Playback Duration” on page 585.
Selecting the LFO2 Duration for Playback Pitch
Synchronic Playback Offset
Playback Offset lets you define where in the slice
playback starts. Playback Offset can be continuously modulated by the LFOs. The effect of the
Playback Offset will vary depending on the selected Playback Mode.
Enable (On, Off) Enables or disables Offset mod-
ulation.
Selecting the LFO1 Duration for Playback Pitch
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BEG and END Range (0–100%) The Offset Range
references a “Start” and “End” offset value that
is set using the Range controls.
Offset Enable
button
Offset Select
button
Smear Playback Offset
If Playback Mode is set to Smear, Playback Offset determines the amount of smearing that occurs between adjacent slices.
Low Frequency Oscillator (LFO)
LFO Waveshape Selects the modulation waveshape for Playback Offset.
Beginning Offset
(0–100
End Offset
(0–100
LFO Waveshape
selector
LFO Duration
selector
Playback Offset settings
Standard, Spin, and Stab Playback Offset
LFO2 Duration (8 Bars, 4 Bars, 2 Bars, 1 Bar, Half
Note, Quarter Note) Applies the LFO according
to the selected duration.
For more information on LFO Waveshape
and Duration, see “Low Frequency Oscillator (LFO)” on page 589.
Assigning Synchronic Playback
User Knobs (PB1 and PB2)
(RTAS Only)
If Playback mode is set to Standard, Spin, or
Stab, Playback Offset determines where within a
slice playback should start. A slice normally
starts playback at the very first sample of the
slice. However, Synchronic lets you start playback of the slice at a point that is offset by some
percentage into a slice, thus creating some very
interesting effects. The Playback Offset can
range from 0% (the start of the slice) to 100%
(the end of the slice).
In Edit mode, you can assign the Playback User
Knobs to control any Playback Edit parameter.
User Knob assignments are made on a per preset
basis. This gives you a great deal of flexibility on
how you can control the Synchronic Playback
module in Performance mode.
Stretch Playback Offset
If Playback Mode is set to Stretch, Playback Offset determines that amount of granular recycling that occurs. Higher Offset percentages result in increased resonance as the beginning of
the slice is recycled more and more to fill the duration of the slice.
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589
To assign a User Knob to control Edit parameters:
Select the desired Edit parameter assignment
from the User Knob Assignment pop-up menu.

Synchronic Effect
Performance Mode
(RTAS Only)
In Performance mode, the Effect module provides Preset, Sound A/B selectors, and User
Knob controls.
Performance mode applies only to the RTAS
version of Synchronic.
Sound A/B
selectors
Performance/Edit
Mode toggle
Presets
User Knobs
(PB1 and PB2)
Effect module, Performance mode
Selecting a User Knob assignment (PB2 assigned to
Offset Beginning)
To select the same User Knob assignment for
all presets, press and hold Alt (Windows) or
Option (Mac) while selecting the User Knob
assignment.
Synchronic Effect Module
Synchronic provides an Effect module that include four effects: Gain, Noise, Filter, and Delay.
Gain, Noise, Filter, and Delay effects can be used
independently or in any combination.
You can toggle between Effect Performance (see
“Synchronic Effect Performance Mode” on
page 590) mode and Effect Edit mode (see “Synchronic Effect Edit Mode” on page 591).
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Sound A and B Selectors
The Sound A and B Sound selectors determine
whether or not the A and B sounds are patched
into the Effect module. Click the Sound A or B
selectors to toggle the A or B sound on or off.
Sound A or B may still be routed through
the Effect module even if it isn’t audible due
to the crossfader position in the XFade module.
Playback Presets
Twelve Playback presets let you recall stored effects. Selecting a preset recalls the last edited set
of parameters for the preset. Effect presets let
you invoke different effects in rapid succession,
to create a compelling musical performance.
Effect User Knobs (FX1 and FX2)
The Effect module provides two assignable User
Knob controls (FX1 and FX2) that provide direct
control over any of the Effect modules Edit
mode settings. The current control assignment
is displayed below each User Knob.
Synchronic Effect Edit Mode
In Edit mode, the Effect module provides easy
access to the Gain, Noise, Filter, and Delay effects and their parameters.
A/B Sound
selectors
Effect Enable and
Select buttons
Preset selector
Performance/Edit
Mode toggle
Selected Effect
parameters
Selecting a Synchronic Effect
for Editing
To view the Gain, Noise, Filter, or Delay effect
parameters:
 Click the Effect Select button (Gain, Noise,
Filter, or Delay). The selected effect’s button will
illuminate, and the effect’s parameters will be
available for editing.
Synchronic Gain Effect
Gain is used to add Volume, Distortion, or Saturation to the sound (A/B Sounds) coming from
the Playback module. Similar to the Playback
section, the Range controls set Beginning and
End values. Range settings can also be modulated by an LFO to create dynamic gain effects.
Gain Enable
button
Gain Select
button
User Knob
assignments
Beginning Gain
amount (in dB)
End Gain
amount (in dB)
LFO Waveshap
selectors
LFO Duration
selectors
Gain Mode
Effect module, Edit mode (Gain shown)
Enabling a Synchronic Effect
Enabling an effect processes the sound (A/B
Sounds) coming from the Playback module. Disabling an effect essentially bypasses the effect.
To enable or disable the Gain, Noise, Filter, or
Delay effect:
 Click the Effect Enable button above the Effect
Select button. The button is illuminated when
the effect is enabled.
Gain effect
BED and END Range (–96 to 24 dB) Use the BEG
and END parameters to set the amount of Gain
effect. The specified Range settings control the
amount of modulation by LFO1 and LFO2.
LFO1 and LFO2 Set the Waveshape and Duration for dynamic Gain effects. For static gain effects, set both LFO1 and LFO2 to Off.
LFO Waveshapes and Durations options are
the same as those provided in the Playback
module. See “Low Frequency Oscillator
(LFO)” on page 589.
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591
Gain Mode
The Gain effect provides three different modes:
Volume, Distortion, and Saturation.
Synchronic Noise Effect
Noise modulates the post-Gain signal with a
noise source. There are three different Noise
modes: Dark, White, and Brite, each with two
variations (Osc or AM).
Noise Enable
button
Noise Select
button
Beginning Noise
amount (%)
End Noise
amount (%)
LFO Waveshap
selector
LFO Duration
selector
Noise Mode
Noise effect
Selecting Gain Mode
Volume Applies linear gain with a range of –96
to +24 dB.
Distortion Applies a non-linear distortion curve
to clip the signal. Distortion is most pronounced
from 0 to +24 dB.
Saturation Applies a warmer, fuzzier distortion
than regular Distortion. Unlike Volume and
Distortion, when Saturation is selected, lower
BEG and END values do not result in attenuating
the signal. A lower setting (–96 dB) results in a
cleaner, less saturated signal, and a higher setting (+24 dB) results in a more saturated signal.
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BEG and END Range (0–100%) Use the BEG and
END parameters to control the amount of modulation by noise. At 0%, there is no modulation.
At 100%, the input signal is 100% modulated by
the selected noise (mode), such that only the
amplitude envelope of the input signal remains.
Use the Range controls to set the amount of
Noise effect that will be modulated by the LFO1
and LFO2 waveshapes.
LFO Set the Waveshape and Duration for dynamic Noise effects. For static noise effects, set
to Off.
LFO Waveshapes and Durations options are
the same as those provided in the Playback
module. See “Low Frequency Oscillator
(LFO)” on page 589.
Noise Mode
There are three different Noise modes: Dark,
White, and Brite; each with two variations: a
noise generator (Osc) or amplitude modulation
(AM).
Synchronic Filter Effect
The Filter effect processes the post-Noise signal
using a low pass filter, a high pass filter, or ring
modulation.
Filter Enable
button
Filter Select
button
Beginning Filter
amount (%)
End Filter
amount (%)
LFO Waveshap
selectors
LFO Duration
selectors
Filter Mode
Filter effect
Selecting Noise Mode
Dark Modulates the signal with a low pass filtered form of white noise, using either an oscillator or amplitude modulation (AM).
White Modulates the signal with white noise, using either an oscillator or amplitude modulation
(AM).
Brite Modulates the signal with a high pass filtered form of white noise, using either an oscillator or amplitude modulation (AM).
BEG and END Range (0–100%) Use the BEG and
END parameters to determine the range of a filter frequency sweep when modulated by the LFO
Waveshapes. The actual cutoff frequency in
Hertz varies from between filter types.
LFO1 and LFO2 Set the Waveshape and Duration for dynamic sweeping filter effects. For
static filter effects, set both LFO1 and LFO2 to
Off.
LFO Waveshapes and Durations options are
the same as those provided in the Playback
module. See “Low Frequency Oscillator
(LFO)” on page 589.
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593
Filter Mode
Synchronic Delay Effect
The Filter effect provides seven different types
of filters: Lowpass Filter (1–5), High pass Filter,
and Ring Modulation.
The Delay effect processes the post-Filter signal
with modulating delay that synchronizes to the
Pro Tools MIDI Beat Clock.
Delay Enable
button
Delay Select
button
Beginning Delay
amount (%)
End Delay
amount (%)
LFO Waveshap
selector
LFO Duration
selector
Delay Mode
Delay effect
Selecting Filter Mode
LPF1 Is a 6 dB per octave low pass comb filter.
LPF2 Is a 12 dB per octave low pass filter with no
resonance.
LPF3 Is a 12 dB per octave low pass filter with a
small amount of resonance.
LPF4 Is a 12 dB per octave low pass filter with a
moderate amount of resonance.
LPF5 Is a 12 dB per octave low pass filter with a
high amount of resonance.
HPF Is a 12 dB per octave high pass filter.
Ring Mod Is a ring modulator. Ring Modulation,
also known as Amplitude Modulation (AM), results in the sum and difference tones between
the input signal and the modulating signal.
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Audio Plug-Ins Guide
BEG and END Range (0–100%) Use the BEG and
END parameters to control the amount of delay
(which is a combination of delay level and feedback level). At 0% there is no delay. At 100% the
delay forms and “infinite” feedback loop. Use
the Range controls to set the amount of delay effect that will be modulated by the LFO1 and
LFO2 waveshapes.
LFO Set the Waveshape and Duration for dynamic delay effects. For static delay effects, set
to Off.
LFO Waveshapes and Durations options are
the same as those provided in the Playback
module. See “Low Frequency Oscillator
(LFO)” on page 589.
Delay Mode
Delay Mode selects a delay time based on a
rhythmic subdivision of the Pro Tools MIDI
Beat Clock (quarter-not, dotted eighth note,
eighth note, eighth note triplet, sixteenth note,
sixteenth note triplet, thirty-second note, or
thirty-second note triplet).
To assign a User Knob to control Edit parameters:
 Select the desired Edit parameter assignment
from the User Knob Assignment pop-up menu.
Selecting a User Knob assignment (FX2 assigned to
Delay Beginning)
Selecting Delay Mode
Assigning Synchronic Effect
User Knobs (FX1 and FX2)
(RTAS Only)
To select the same User Knob assignment for
all presets, press and hold Alt (Windows) or
Option (Mac) while selecting the User Knob
assignment.
In Edit mode, you can assign the Effect User
Knobs to control any Effect Edit parameter.
User Knob assignments are made on a per preset
basis. This gives you a great deal of flexibility on
how you can control the Synchronic Effect module in Performance mode.
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595
Synchronic XFade Module
Overview
(RTAS Only)
To assign a User Knob to control Edit parameters:
 Select the desired Edit parameter assignment
from the User Knob Assignment pop-up menu.
The XFade module mixes the A and B audio signals using either preset crossfade effects, or a
crossfade fader. The XFade module has three
modes: Manual mode, Preset mode, and Edit
mode. The XFade Module also has an assignable
User Knob control (XF).
Synchronic XFade User Knob
(XF)
The XFade module provides a single assignable
User Knob control (XF) that provides direct
control over any of the XFade module’s Edit
mode settings. The current control assignment
is displayed below the User Knob.
Assigning the XFade User Knob (XF)
In Edit mode, you can assign the XFade User
Knob to control any XFade Edit parameter. User
Knob assignments are made on a per preset basis. This gives you a great deal of flexibility on
how you can control the Synchronic XFade module in Performance mode.
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Selecting the User Knob assignment (XF assigned to
Rate 2, LFO2 Duration)
To select the same User Knob assignment for
all presets, press and hold Alt (Windows) or
Option (Mac) while selecting the User Knob
assignment.
Synchronic XFade Manual Mode
Synchronic XFade Preset Mode
Manual mode is the default Performance mode
for the XFade module. When in Manual mode,
the currently selected XFade preset is overridden and can control the mix between the A and B
sounds using a crossfade fader. Click the A or B
labels on either side of the crossfader to solo either sound.
The XFade module provides twelve presets to
store and recall crossfade settings.
Performance/Edit
Mode toggle
Manual/Preset
Mode toggle
Presets
Performance/Edit
Mode toggle
Manual/Preset
Mode toggle
XFade indicator
Sound B
Crossfade fader
Sound A
XFade indicator
User Knob (XF)
XFade module, Preset mode
Synchronic XFade Edit Mode
XFade module, Manual mode
To toggle between Manual and Preset mode:
In Edit mode, the XFade module lets you edit
crossfade modulation effects and the XFade
User Knob assignment.
If necessary, click the Performance/Edit Mode
toggle to switch to Performance mode.
1
Preset selector
Performance/Edit
Mode toggle
2 Click the Manual/Preset Mode toggle to switch
between Manual and Preset mode.
Beginning and End
XFade amount (%)
Manual mode takes precedence over any
XFade preset automation. When switching
out of Manual mode, any XFade automation previously received will take effect.
LFO Waveshape and
Duration selectors
XFade indicator
User Knob
assignment
XFade module, Edit mode
XFade Effects
Crossfade modulations can be used to create
complex crossfades between A and B sounds using two LFOs.
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597
Range (–100 to +100%) Use the Beginning (BEG)
and End (END) parameters to control the mix
between the A and B sounds. You can set both
start and stop points to determine the range of
modulation by the LFOs. A value of –100% sets
the crossfade fader to the full left position (the A
sound), 0% is an equal mix of the and A and B
sounds, and 100% sets the crossfade fader to the
full right position (the B sound).
LFO Set the Waveshape and Duration for dy-
Synchronic MIDI
Performance Mode
In Performance mode, the MIDI module displays a one octave on-screen keyboard. Octave
Transpose buttons lets you shift the focus of the
on-screen keyboard to any octave (from MIDI
note number 1 to 127).
Performance/Edit
Mode toggle
On-screen Keyboard
namic crossfade effects. For static crossfade settings, set both LFO1 and LFO2 to Off.
LFO Waveshapes and Durations options are
the same as those provided in the Playback
module. See “Low Frequency Oscillator
(LFO)” on page 589.
Synchronic MIDI Module
Overview
(RTAS Only)
The MIDI module lets you assign MIDI note
numbers and continuous controllers to the Presets and User Knobs of the Sound, Playback, Effect, and XFade modules.
You can toggle between MIDI Performance
mode (see “Synchronic MIDI Performance
Mode” on page 598) and MIDI Edit mode (see
“Synchronic MIDI Edit Mode” on page 600).
Since there are twelve presets for each module, you can comfortably map your MIDI
keyboard controller to the presets for each
module by octaves.
Transpose
Octave Up
Wait
Octave
button
Transposition button
Transpose
display
Octave Down
button
Assign
button
MIDI module, Performance mode
Synchronic On-Screen Keyboard
In Performance mode, the MIDI module provides an on-screen keyboard that can be used to
assign MIDI note numbers to store and recall
the current state of Synchronic settings. You can
also recall stored combinations of Synchronic
presets and settings by clicking the corresponding key on the on-screen keyboard.
Octave (OCT) Up and Down Arrow Buttons Click
the left arrow to transpose the on-screen keyboard down an octave or click the right arrow to
transpose the on-screen keyboard up an octave.
This lets you readily assign Synchronic preset
snapshots to any and all octaves of your MIDI
keyboard.
Octave Display Displays the current octave
transposition of the on-screen MIDI keyboard.
C3 is middle C.
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Pre-Mapped Synchronic MIDI
Keys
Synchronic provides a default MIDI keyboard
mapping for triggering Sound, Playback, Effect,
and Crossfade presets.
Synchronic’s default MIDI key mappings are:
• MIDI note numbers 12–23 are reserved for
custom snapshots (see “Assigning MIDI
Keys in Synchronic” on page 599).
• MIDI note numbers 24–35 select Sound
presets A1–A12.
• MIDI note numbers 36–47 select Sound
presets B1–B12.
• MIDI note numbers 48–59 select Playback
presets 1–12.
• MIDI note numbers 60-71 select Effect presets 1–12.
• MIDI note numbers 72-83 select XFade presets 1–12.
These pre-defined mappings can be overwritten in Assign mode.
Assigning MIDI Keys in
Synchronic
Synchronic lets you capture all aspects of the
current playback state (including the A Sound
preset, the B Sound preset, the Playback preset,
the Effect preset, and the XFade preset or fader
position), or some sub-set of the playback state
(for example, only the A and B Sound presets).
Once captured and stored (assigned to a MIDI
note number), this combination can be recalled
using either the on-screen keyboard or with a
MIDI controller.
To capture and assign a combination to a MIDI
note number:
Edit Synchronic's Sound, Playback, Effect, and
XFade parameters and select presets as desired.
1
If necessary, switch the MIDI module to Performance mode (see “Performance and Edit
Modes” on page 569).
2
Click the Assign button. The Assign button illuminates and the on-screen keyboard’s keys
display their module assignment icons (see
“MIDI Key Assignment Icons” on page 600).
3
Enable or disable the Module Assign Enable
buttons as desired (see “Selective Module Assignment” on page 600).
4
Click the MIDI key on the on-screen keyboard
(or press the corresponding key on your MIDI
keyboard).
5
The current state of the assign-enabled modules
will be stored and assigned to the selected key
and corresponding MIDI note number.
If a MIDI key is used to recall a combination that
does not include all modules, the modules that
are not included in the combination will remain
as they were. For example, a combination that
only includes A and B Sounds presets will have
no effect on the state of the Playback, Effect, or
XFade modules.
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599
Selective Module Assignment
MIDI Key Assignment Icons
A Synchronic combination need not consist of
changes in all Synchronic modules. When the
Assign button is enabled, the Sound, Playback,
Effect, and XFade modules display an Assign
Enable button. A module only displays its Assign Enable button in Performance mode.
In Assign mode, the on-screen keyboard displays icons to show which modules to which a
key is assigned. There are five different colored
icons for Sound A, Sound B, Playback, Effect,
and XFade.
(
MIDI Key Assignment icons (Key C1 is assigned to
control Sound A, Sound B, Playback, Effect, and
XFade)
Wait Bar Forces recalled combinations to wait
and start at the beginning of the next bar instead
of the next beat or slice.
Synchronic MIDI Edit Mode
Assign
button
Module Assign Module Assign
Enable buttons Enable button
(disabled)
(enabled)
In Edit mode, MIDI Pitch Bend or MIDI continuous controller numbers can be assigned to control Synchronic’s five User Knobs and the XFade
crossfade fader.
Module Assign Enable buttons
MIDI module, Edit mode
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Audio Plug-Ins Guide
Assigning an External MIDI Controller
Below the five User Knobs and crossfader (XF)
labels are pop-up menus for assigning MIDI
controllers.
To save a template of your MIDI control assignments, configure a default Synchronic
insert (with no sounds loaded) for your
MIDI controller and save it as a Synchronic
plug-in settings file using the Settings Librarian (see Chapter 104, “Synchronic PlugIn Settings”).
To assign an external MIDI controller:
If necessary, switch the MIDI module to Edit
mode (see “Performance and Edit Modes” on
page 569).
1
Select Pitch Wheel or Controller # from the
Source pop-up menu.
2
Pitch Wheel Assigns MIDI pitch bend (the pitch
wheel) to control the corresponding User Knob
or the XFade crossfade fader.
Controller # Assigns a continuous MIDI control-
ler to control the corresponding User Knob or
the XFade crossfade fader.
If Controller # is selected in the Source popup menu, and the Controller # field is selected, you can jiggle the MIDI controller to
make the correct controller assignment.
Synchronic Keyboard Focus
Mode
(RTAS Only)
In Keyboard Focus mode, you can use your computer keyboard to trigger Synchronic presets.
Keyboard Focus shortcuts for Synchronic are:
Keyboard Numbers 1–9, 0 Triggers sounds
A1–10.
Keyboard Characters QWERTYUIOP Trigger
Playback presets 1–10.
Selecting MIDI control source
3 If you selected Controller #, you will also need
to enter the MIDI controller number in the MIDI
Control Number field, either by selecting the
field and typing the number or by jiggling the
MIDI controller.
Keyboard Characters ASDFGHJKL Trigger Ef-
fect presets 1–10.
Keyboard Characters ZXCVBNM,./ Trigger
Crossfade presets 1–10.
MIDI Controller # field assignment for PB2 User Knob
(set to MIDI controller #1, the modulation wheel)
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601
To enable or disable Keyboard Focus mode for
Synchronic:
 Click the “a...z” Keyboard Focus Enable button
(in the upper-right corner of the Synchronic
plug-in window).
“a...z” Keyboard Focus Enable button (enabled)
If Synchronic Keyboard Focus is enabled, designated characters (listed above in QWERT Keyboard Mapping) will control Synchronic and
will not be used otherwise by Pro Tools.
Keyboard Focus has no effect on normal text
entry for Synchronic parameter values.
Although multiple Synchronic Plug-In windows can have Keyboard Focus (“a...z”) enabled, only the last clicked-on Plug-In window will respond to the Keyboard Focus key
commands.
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Chapter 102: Using Synchronic as an
AudioSuite Plug-In
The AudioSuite version of Synchronic shares
many of the features found in its RTAS counterpart, with some differences as noted in this section.
Synchronic AudioSuite is available in the AudioSuite menu in Pro Tools under the Instrument
(or Digidesign) categories.
To create an instance of Synchronic AudioSuite:
• Choose AudioSuite > Synchronic.
Plug-In settings and presets can be shared
between the AudioSuite and RTAS versions
of Synchronic. However, the AudioSuite version of the plug-in can import and export
only information stored for the Sound, Playback, and Effect modules in the first preset.
Synchronic AudioSuite
Modules
The AudioSuite version of Synchronic includes
three of the five modules found in its real-time
counterpart: Sound, Playback, and Effect. Each
of these modules has been modified slightly for
the AudioSuite version.
The XFade and MIDI modules are real-time
based and are included only in the RTAS
version of Synchronic.
Edit Mode Only in AudioSuite Modules
In the RTAS version of Synchronic, each module
can be independently switched between Edit and
Performance modes with the Mode toggle.
Synchronic AudioSuite
In the AudioSuite version of Synchronic, the
modules function only in Edit mode. Performance mode is not available since its features
are not relevant to non-real time functionality,
and the Mode toggle does not exist.
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603
Synchronic Sound Module
Synchronic Playback Module
The Sound module lets you import, delete, sliceup, and fine-tune audio (loops) for Synchronic
playback.
The Playback module lets you edit the Playback
parameters.
The AudioSuite version of this module replicates the Edit mode functionality of its real-time
counterpart, with the following differences:
Load Selection button The Load Selection but-
ton lets you automatically load a selected portion of audio from the Pro Tools Edit window
into the Waveform display.
Synchronic Playback module (AudioSuite version)
The AudioSuite version of this module replicates the Edit mode functionality of its real-time
counterpart, with the following differences:
Load Selection button
The Load Selection button is available only
in the AudioSuite version of Synchronic.
See “Using the Load Selection Button” on
page 605 for detailed information.
One loop limitation The AudioSuite version of
Synchronic lets you work with one audio loop at
a time, instead of the 12 presets available for
RTAS.
See “Synchronic Sound Edit Mode” on
page 575 for detailed information on using
the Sound module in Edit mode
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Audio Plug-Ins Guide
Playback User Knobs Unavailable
The Playback user knobs are only necessary and
available in the RTAS version of Synchronic.
See “Synchronic Playback Edit Mode” on
page 582 for detailed information on Playback module controls in Edit mode.
Synchronic Effect Module
The Effect module provides easy access to the
Gain, Noise, Filter, and Delay effects and their
parameters.
Synchronic AudioSuite
Workflow
Use the AudioSuite version of Synchronic to
work with and play back audio loops as follows:
Load audio files or a selected portion of audio
clips (loops) into the Sound module, much like
you would add sound to a sampler.
1
Use the Detection Slider to slice up the loops
into rhythmically logical units (beats and subdivisions of the beat).
2
3
Preview the sliced-up loop.
When you are finished with a loop, you render
it to a track.
4
Synchronic AudioSuite Effect module
The AudioSuite version of this module replicates the Edit mode functionality of its real-time
counterpart, with the following differences:
Effect user knobs unavailable The Effect user
knobs are only necessary and available in the
RTAS version of Synchronic.
See “Synchronic Effect Edit Mode” on
page 591 for detailed information on Playback module controls in Edit mode.
Loading Audio into the
Synchronic Waveform Display
In the AudioSuite version of Synchronic, you
can load audio into the Waveform display using
any of the following methods:
• Load Selection button
• Import button
• Drag and drop from Workspace
See “Importing a Sound into Synchronic” on
page 576 for detailed information on using
the Import button or dragging and dropping
from the Workspace.
Using the Load Selection Button
The AudioSuite version of Synchronic lets you
load a selected portion of a Pro Tools audio
track directly into the Waveform display.
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605
To load a selection from Pro Tools into
Synchronic:
1
Choose AudioSuite > Synchronic.
In the Pro Tools Edit window, select the portion of audio you want to loop in Synchronic.
Slicing a Loop in Synchronic
Once you have loaded a loop into the Waveform
display, you can slice it up and preview it.
2
See “Slicing Up a Sound in Synchronic” on
page 579 for detailed information on slicing
a loop.
Previewing a Loop in Synchronic
Before printing a finished audio loop to a
Pro Tools track, you can preview it in the AudioSuite window.
To preview a loop in Synchronic AudioSuite:
Making a selection in Pro Tools
In the Sound module of Synchronic AudioSuite, click Load Selection.
3
• Click Preview in the Plug-In window.
Synchronic plays back the loaded loop according
to the session tempo.
Previewing Synchronic AudioSuite
Clicking the Load Selection button
The selected portion of audio appears in the
Synchronic Waveform display.
Pro Tools selection loaded into Waveform display
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Audio Plug-Ins Guide
Rendering a Synchronic Loop to
a Pro Tools Track
Once you have finished editing your loop in Synchronic AudioSuite, you can render it to a selection in Pro Tools.
Rendering an audio loop from Synchronic
AudioSuite to a selection in a Pro Tools audio track overwrites any audio material in
that selection.
To render an audio loop to Pro Tools:
In the Pro Tools Edit window, select the portion of an audio track where you want to place
the rendered audio.
1
If you imported a selection from the Pro
Tools Timeline using the Load Selection button, you can improve any timing problems
that may exist by modifying the Timeline selection before rendering the loop from Synchronic. For example, if a percussive audio
event were late at the beginning of a bar,
you might load the bar into Synchronic with
the selection exactly at the onset of the
event. Then, you can modify the Timeline
selection to start exactly on “the grid.” Consequently, Synchronic will render the modified loops to the new selection.
Click Render to print the audio loop to the selection.
2
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Chapter 103: Automating Synchronic RTAS
You can automate changes to Synchronic RTAS
parameters in two ways:
You can also use this keyboard shortcut to
open the Plug-In Automation dialog: Control-Start-Alt-click (Windows) or Command-Option-Control-click (Mac) any plugin parameter in the Plug-In window, then
choose Open Automation dialog from the
pop-up menu.
• Using Pro Tools automation playlists
• Using MIDI
Using Automation Playlists
Pro Tools creates a separate playlist for each
plug-in parameter that you automate. Pro Tools
automation lets you record your interaction
with Synchronic parameters using the mouse, or
a control surface (including MIDI control surfaces).
3
Choose the parameters to automate and click
Add. If there are multiple plug-ins on the same
track, you can select from among these by clicking their buttons in the Inserts section of this dialog.
Enabling Plug-In Parameters for
Automation
To enable plug-in parameters for automation:
Open the Plug-In window for the plug-in you
want to automate.
1
2
Do one of the following:
• Click the Automation Enable button in the
Plug-In window.
– or –
• Control-Start-Alt-click (Windows) or Command-Option-Control-click (Mac) the
Track View Selector in the Edit window.
Selecting parameters to automate
Chapter 103: Automating Synchronic RTAS
609
4 Click OK to close the Plug-In Automation dialog.
As an alternative to using the Plug-In Automation window, you can enable individual
plug-in parameters directly from the PlugIn window by Control-Alt-Start-clicking
(Windows) or Command-Control-Optionclicking (Mac) the parameter’s text field or
control. See the Pro Tools Reference Guide
for more information.
.
Shortcut for enabling a Synchronic parameter for
automation
Recording Automation
To record automation:
In the Automation Enable window, make sure
that plug-in automation is write-enabled.
1
2 If using a MIDI control surface, do the following: On the MIDI control surface, assign the
MIDI Controller number for the parameter you
want to automate.
For more information on assigning a MIDI
Controller number, see See “Assinging MIDI
Controller Numbers to Synchronic Knobs”
on page 611.
On the track with Synchronic inserted, choose
an automation mode. For an initial pass, choose
Auto Write.
3
Click Play to begin writing automation, and
move the controls you want to automate.
4
5
610
When you have finished, click Stop.
Audio Plug-Ins Guide
After the initial automation pass, you can write
additional automation to the track without completely erasing the previous pass by choosing
Auto Touch mode or Auto Latch mode. These
modes add new automation only when you actually move the control for that parameter.
If you use automation to control preset
changes, place automation breakpoints
slightly ahead of the point at which the
change is desired. Since Synchronic triggers
all Sound and Playback changes according
to MIDI Clock boundaries, a thirty-second
note ahead of time is recommended.
For more information on creating and editing automation, see the Pro Tools Reference Guide .
Using MIDI
You can automate Synchronic RTAS parameters
by assigning MIDI note and controller data to
Synchronic presets and performance parameters, and recording them to an Instrument or
MIDI track. You can also edit and manually enter the MIDI data on the track as desired, and
use it to control Synchronic during playback.
For information on controlling Synchronic
with MIDI note and controller data, see
“Synchronic MIDI Module Overview” on
page 598.
Assigning MIDI Notes and
Controller Data to Presets
To assign MIDI notes to combinations of
Synchronic presets:
5 Edit Synchronic’s Sound, Playback, Effect, and
XFade parameters and select presets as desired.
The current state of the assign-enabled modules
will be stored and assigned to the selected key
and corresponding MIDI note number.
Assinging MIDI Controller
Numbers to Synchronic Knobs
To assign MIDI controller numbers to the
Playback, Effects, and XFade User Knobs:
If necessary, switch the MIDI module to Edit
mode (see “Performance and Edit Modes” on
page 569).
1
Select Pitch Wheel or Controller # from the
Source pop-up menu for the desired User Knob
assignment.
2
If necessary, switch the MIDI module to Performance mode (see “Performance and Edit
Modes” on page 569).
1
Click the Assign button. The Assign button illuminates and the on-screen keyboard’s keys
display their module assignment icons (see
“MIDI Key Assignment Icons” on page 600).
2
Selecting MIDI control source
If you select Controller #, you will also need to
enter the MIDI controller number in the MIDI
Control Number field, either by selecting the
field and typing the number or by jiggling the
MIDI controller.
3
MIDI Assign button
Click the MIDI key on the on-screen keyboard
(or press the corresponding key on your MIDI
keyboard).
3
MIDI Controller # field assignment for PB2 User Knob
(set to MIDI controller #1, the modulation wheel)
Enable or disable the Module Assign Enable
buttons as desired (see “Selective Module Assignment” on page 600). The MIDI Key Assignment icons update accordingly.
4
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611
Automating Synchronic Using
MIDI
If the desired controller doesn’t appear in the
Track Display Format pop-up menu, select
Add/Remove Controller.
2
To automate Synchronic using MIDI:
Insert Synchronic on a mono or stereo Instrument track.
1
2
Do one of the following:
• Record enable the Instrument track, start
playback, and perform the automation on
your MIDI controller.
– or –
• Use the Pencil tool to draw MIDI note and
controller data.
Viewing MIDI Automation
Pro Tools creates a separate playlist for each
type of automation you write. To view MIDI automation for Synchronic parameters, you may
need to add it to the Track Display Format popup menu using the Add/Remove Controller command.
Track Display Format selector
3 In the Automated MIDI Controllers dialog, locate the parameter by its MIDI Controller number and click Add, then OK.
MIDI Pitch Bend automation controlling Synchronic
To view MIDI automation:
In the Edit window, choose the controller that
you want to view.
1
Automated MIDI Controllers dialog
4 Select the new automation type from the Track
Display Format selector of the Instrument track.
For more information on Pro Tools automation, see the Pro Tools Reference Guide .
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Chapter 104: Synchronic Plug-In Settings
The Settings Librarian makes it easy to create
your own library of Synchronic configurations,
including loaded audio files. Using the Librarian
and Settings pop-up menus, you can copy, paste,
save, and import these patches from plug-in to
plug-in, or from session to session.
For more information Plug-In Settings files
and using the Librarian, see the Pro Tools
Reference Guide.
Pro Tools always saves the current session’s
plug-in settings with the session itself. The
Librarian allows you to access settings saved
during other sessions.
Plug-In settings and presets can be shared
between the AudioSuite and RTAS versions
of Synchronic. However, the AudioSuite version of the plug-in can import and export
only information stored for the Sound, Playback, and Effect modules in the first preset.
Imported Audio Stored with
Settings
Synchronic is unique in that it stores loaded audio files as part of the plug-in settings file. When
you import audio files into Synchronic, it loads
them into RAM and then stores them with the
plug-in settings. The plug-in settings are stored
with the Pro Tools session file or using the Settings Librarian to save a plug-in settings file
(.tfx). The size of the Pro Tools session file or
plug-in settings file will increase corresponding
to the number and size of audio files loaded into
Synchronic.
For more information on importing audio
files in Synchronic, see “Importing a Sound
into Synchronic” on page 576.
Presets within the Sound, Playback, Effect,
and XFade modules cannot be independently saved using the Pro Tools plug-in
settings librarian. Saving a Synchronic
plug-in setting using the Pro Tools Settings
Librarian will save all of the presets of
each module.
Chapter 104: Synchronic Plug-In Settings
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Audio Plug-Ins Guide
Index
Numerics
660 (Fairchild Limiter) 81
A
Activation Code 11
Amels, Dave 82
audio 125
AudioSuite 487
Input Mode selector 132
plug-ins 5
Preview 132
processing preferences 132
AudioSuite plug-ins
DC Offset Removal 525
Duplicate 525
Time Compression/Expansion 501
authorizing plug-ins 11
automation
enabling plug-in parameters 609
Avid Virtual Instrument plug-ins
Boom 375
DB-33 403
B
BF Essential Clip Remover 487
BF Essential Correlation Meter 488
BF Essential Meter Bridge 488
BF-2A plug-in
side-chain filter 42
BF-3A plug-in
aggressive tone 45
midrange 45
BF76 plug-in 47
FET 47
hit 48
side-chain 48
squishy 47
Bomb Factory plug-ins
BF76 47
Purple Audio MC77 99
SansAmp PSA-1 331
Slightly Rude Compressor 101
Tel-Ray Variable Delay 241
Boom
clearing patterns 380
controlling with MIDI 381
Copy and Clear buttons 376
copying patterns 381
creating patterns 380
Edit mode switch 378
Event Bar 378
Event switches 378
Global controls 377
Info display 379
Instrument section 376
Kit selector 377
Matrix display 376
Pattern Chains 383
Pattern display 376
patterns 375
presets 380
Setup page 379
Speed switches 378
Start and Stop buttons 377
step velocity 376, 379
switching between patterns 381
switching patterns using MIDI 382
synchronization modes 379
Boom plug-in 375
broadcast noise 267
Bruno
Pitch controls 391
Threshold control 389
Timbre controls 389
using MIDI 386
Voice controls 392
Index
615
Bruno and Reso plug-ins 385
ADSR Envelope Generator 391, 396
All (harmonics) control 395
Amplitude
controls 390, 396
envelope 391
Attack control 391
Bend Range control 391, 397
Bruno
features 385
Crossfade
control 389
frequency 389
cross-synthesis 385
Damping
Damping Amount control 394
Damping Velocity control 394
Decay control 391
Detune
Detune Amount control 392, 397
Detune Velocity control 392, 397
Envelope Follower 398
Envelope Generator 391
External Key 389, 395
Follower control 398
Frequency control 398
Gain
Gain Amount control 390, 396
Gain Velocity control 390, 396
Glide control 391
harmonic overtones of resonator 394
Key Input 387, 389, 395
Key Listen control 389
Latch bar 387
Low Pass Filter control 398
Master Tune control 391, 397
MIDI Beat Clock 389
MIDI Clock control 389, 395
Mix control 390
Mono voice mode for Reso 392, 398
Odd (harmonics) control 395
Online Help 393
on-screen keyboard 386, 393
Poly voice mode 392, 399
Portamento control 391, 397
Q control 398
Release control 391
Resonance (Q) control 398
Resonance Amount control 394
Resonance Velocity control 394
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Audio Plug-Ins Guide
Resonant peak 398
Resonator 385, 394
Spread control 390
Stereo spread 390
Sustain Level control 391
Switch control 389
Timbrometer 390
time-slicing 385
Toggle (harmonics) control 395
Voice Mode control 392
Voice Stack control 393
voice stacking 397
wah-wah effect 398
wave sequencing 388
buzz 267
C
Channel Strip
bypassing effects modules 53
Compressor/Limiter 55, 56
Attack control 56
Gain control 57
Knee control 57
Ratio control 56
Release control 57
Threshold control 56
Dynamics 54
adjusting controls graphically 55
Dynamics Graph display 54
enabling (or disabling) effects 50
Expander/Gate
Attack control 55
Depth control 55
Hold control 55
Hysteresis (Hyst) control 56
Knee control 56
Ratio control 55
Release control 56
Threshold (Thresh) control 55
FX Chain 53
Gain Reduction meters 52
Input meters 52
Input Trim 51
Listen button 51
Phase Invert 52
Side Chain
Detection options 58
Filter Frequency control 58
Filter Type options 58
Side Chain Processing Graph display 59
Source selector 58
side-chain processing 57
Channel Strip plug-in 49
Chorus plug-in 269
Delay control 270
Feedback control 270
LFO Rate control 270
LFO Waveform control 270
LFO Width control 270
Low-Pass Filter control 270
Mix control 270
clarify 92
Class-B 81
Click plug-in 401
Accented control 401
Unaccented control 401
clip remover 487
compression 85
Smack 103
Contour display 183
Copy and Clear buttons (Boom) 376
copying 381
Cosmonaut Voice plug-in
radio effects 267
shortwave simulation 267
squawk and squelch 267
Creative Collection plug-ins
Dynamic Delay 217
Enhancer 313
Frequency Shifter 119
Kill EQ 15
Lo-Fi 315
Mini Grand 411
Multi-Delay 221
Non-Linear Reverb 147
Reverb 149
Spring Reverb 153
Stereo Width plug-in 355
Structure Free 417
Vacuum 457
Xpand2 467
D
DB-33
Cabinet page 406
drawbars 404
Info display 407
Key Click 405
Mics controls 406, 407
Percussion 405
Rotation Speed switch 405
Scanner Vibrato
shapes 404
Speed controls 407
tonewheel settings 404
Tube Pre-amp 406
DB-33 plug-in 403
DC Offset Removal plug-in 525
Delay plug-in (Creative Collection) 217
D-Fi plug-ins 135, 289, 319, 323
Lo-Fi 319
Recti-Fi 323
Sci-Fi 289
Vari-Fi 135
D-Fx plug-ins
Chorus 269
Flanger 271
Multi-Tap Delay 233
Ping-Pong Delay 235
Digidesign Intelligent Noise Reduction 335
DigiRack plug-ins
DC Offset Removal plug-in 525
DINR LE plug-in 335
DINR plug-in 335
Auto Fit button 340
Broadband Noise Reduction 335, 337
Contour Line 336, 337
editing 343
downward expanders 336
dynamic audio signal modeling 335
Fit button 337, 339
High-Shelf EQ control 338
Learn button 339
Learn First Audio Mode 339
Learn Last Audio Mode 339
Noise Contour line 335
Noise Reduction limitations 336
Noise Signature 335, 337
Preamp noise 335
Release control 338
Response control 338
Scroll Left/Right buttons 340
Smoothing control 338
Spectral Graph 336
Super Fit button 339
tape hiss 335
Undo button for DINR 341
Zoom Out/In buttons 340
Index
617
Dither plug-in 349, 351
bit resolution for Dither plug-in 349
Noise Shaping 350
Down Mixer
multichannel to stereo 357
stereo to mono 357
supported multichannel formats 357
Down Mixer plug-in 357
Drawbars 404
DSP
and EQ III 29
Duplicate plug-in 525
flattening a track 525
D-Verb plug-in 155
Algorithm control 156
Church algorithm 156
Clip indicator 155
Diffusion control 156, 157
Hall algorithm 156
Hi Frequency Cut control 157
Low-Pass Filter control 157
Output Meter 155
Size control 156
dynamics
Fairchild 660 81
Fairchild 670 83
Dynamics III plug-ins 65
De-Esser III
sibilants 73
E
Edit mode switch (Boom) 378
Eleven Free