Audio Plug-Ins Guide
Audio Plug-Ins Guide
Version 10.3
Legal Notices
This guide is copyrighted ©2012 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,
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Guide Part Number 9329-65246-00 REV A 09/12
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Contents
Part I
Introduction to Pro Tools Plug-Ins
Chapter 1. Audio Plug-Ins Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Avid Audio Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Plug-In Formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Using Plug-Ins in Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Conventions Used in Pro Tools Documentation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
System Requirements and Compatibility for Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Contents of the Boxed Version of Your Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
About www.avid.com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Chapter 2. Installing Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
Plug-Ins Included with Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
Factory-Installed VENUE Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
Updating Older Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
Using Pro Tools Plug-Ins with Avid Media Composer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Installing Plug-Ins for Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Installing Plug-Ins for VENUE Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Authorizing Paid Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Removing Plug-Ins for Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
Removing Plug-Ins for VENUE Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
Contents
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Part II
EQ Plug-Ins
Chapter 3. AIR Kill EQ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Kill EQ Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Chapter 4. EQ III. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
EQ III Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Adjusting EQ III Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
EQ III I/O Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
EQ III EQ Band Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
EQ III Frequency Graph Display. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
7 Band EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
2–4 Band EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
1 Band EQ III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
Chapter 5. JOEMEEK VC5 Meequalizer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
JOEMEEK Meequalizer Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
Chapter 6. Pultec Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
Pultec EQP-1A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
Pultec EQH-2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
Pultec MEQ-5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
Part III
Dynamics Plug-Ins
Chapter 7. BF-2A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
BF-2A Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
BF-2A Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
Chapter 8. BF-3A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
BF-3A Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
BF-3A Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
Chapter 9. BF76 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
BF76 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
BF76 Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
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Audio Plug-Ins Guide
Chapter 10. Channel Strip . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Sections and Panes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
Output . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
FX Chain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
Dynamics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
EQ/Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Chapter 11. Dynamics III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Dynamics III Shared Features and Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Compressor/Limiter III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
Expander/Gate III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
De-Esser III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
Dynamics III Side-Chain Input. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
Chapter 12. Fairchild Plug-Ins. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
Fairchild 660 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
Fairchild 670 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Chapter 13. Impact . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Impact Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Using the Impact Compressor. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
Chapter 14. JOEMEEK SC2 Compressor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
JOEMEEK Compressor Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
JOEMEEK Compressor Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Chapter 15. Maxim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
About Peak Limiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
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
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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
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
Vari-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 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
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Audio Plug-Ins Guide
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
Chapter 29. D-Verb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
D-Verb Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
Selections for D-Verb AudioSuite Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Chapter 30. Reverb One. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
A Reverb Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
Reverb One Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
Reverb One Graphs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
Other Reverb One Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Chapter 31. ReVibe. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
Using ReVibe . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
Adjusting ReVibe Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
ReVibe Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
ReVibe Decay Color & EQ Section Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
ReVibe Contour Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
ReVibe Input/Output Meter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182
ReVibe Help Button . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182
ReVibe Room Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
Chapter 32. ReVibe II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Using ReVibe II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Adjusting ReVibe II Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 188
ReVibe II Input and Output Meters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
ReVibe II Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
ReVibe II Decay EQ Graph . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
ReVibe II Decay Color Graph . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
ReVibe II Contour Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
ReVibe II Room Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
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Chapter 33. TL Space TDM and TL Space Native . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
TL Space Feature Highlights . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
TL Space Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205
TL Space and System Performance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208
Impulse Response (IR) and TL Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
TL Space Presets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
TL Space Snapshots . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 215
TL Space Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 216
TL Space Display Area . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
TL Space IR Browser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
TL Space Primary Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 222
TL Space Group Selectors and Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223
TL Space Info Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 225
Using TL Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
TL Space IR Library . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 228
Part VI
Delay Plug-Ins
Chapter 34. AIR Dynamic Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 231
Dynamic Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 231
Chapter 35. AIR Multi-Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Multi-Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Chapter 36. Mod Delay II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
Mod Delay II Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 238
Multichannel Mod Delay II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
Selecting Audio for ModDelay II AudioSuite Processing . . . . . . . . . . . . . . . . . . . . . . . . . . 240
Chapter 37. Mod Delay III . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 241
Mod Delay III Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 241
Selections for Mod Delay III AudioSuite Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Chapter 38. Moogerfooger Analog Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 245
Moogerfooger Analog Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Chapter 39. Multi-Tap Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
Multi-Tap Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
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Chapter 40. Ping-Pong Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 249
Ping-Pong Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 249
Chapter 41. Reel Tape Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
Reel Tape Common Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
Reel Tape Delay Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
Chapter 42. Tel-Ray Variable Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255
Tel-Ray Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 256
Tel-Ray Tips and Tricks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 256
Chapter 43. TimeAdjuster . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
TimeAdjuster Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
Using TimeAdjuster for Manual Delay Compensation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
When to Compensate for Delays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260
Part VII
Modulation Plug-Ins
Chapter 44. AIR Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
AIR Chorus Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
Chapter 45. AIR Ensemble. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Ensemble Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Chapter 46. AIR Filter Gate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Filter Gate Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Chapter 47. AIR Flanger. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
AIR Flanger Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Chapter 48. AIR Fuzz-Wah. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Fuzz-Wah Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Chapter 49. AIR Multi-Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Multi-Chorus Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Chapter 50. AIR Phaser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 275
Phaser Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 275
Chapter 51. AIR Talkbox . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 277
Talkbox Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 277
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Chapter 52. AIR Vintage Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
Vintage Filter Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
Chapter 53. Cosmonaut Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Cosmonaut Voice Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Chapter 54. Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Chorus Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Chapter 55. Flanger . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
Flanger Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
Chapter 56. Moogerfooger Lowpass Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
Chapter 57. Moogerfooger 12-Stage Phaser. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
Chapter 58. Moogerfooger Ring Modulator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 295
Chapter 59. Reel Tape Flanger . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 297
Reel Tape Common Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
Reel Tape Flanger Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
Chapter 60. Sci-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
Sci-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
Chapter 61. TL EveryPhase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
TL EveryPhase Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
TL EveryPhase Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 308
Using TL EveryPhase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Chapter 62. Voce Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 317
Voce Chorus/Vibrato . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 317
Voce Spin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
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Part VIII
Harmonic Plug-Ins
Chapter 63. AIR Distortion. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
Distortion Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
Chapter 64. AIR Enhancer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
Enhancer Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
Chapter 65. AIR Lo Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
AIR Lo Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
Chapter 66. Lo-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333
Lo-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 334
Chapter 67. Recti-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 337
Recti-Fi Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 338
Chapter 68. Reel Tape Saturation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 341
Reel Tape Common Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 341
Reel Tape Saturation Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 342
Chapter 69. SansAmp PSA-1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
Part IX
Noise Reduction Plug-Ins
Chapter 70. DINR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
How Broadband Noise Reduction Works . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
BNR Spectral Graph . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
Broadband Noise Reduction Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Using Broadband Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 355
Using BNR AudioSuite . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 359
Part X
Dither Plug-Ins
Chapter 71. Dither . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 363
Dither Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 363
Chapter 72. POW-r Dither . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
POW-r Dither Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
Contents
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Part XI
Sound Field Plug-Ins
Chapter 73. AIR Stereo Width. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 369
Stereo Width Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 369
Chapter 74. Down Mixer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 371
Chapter 75. SignalTools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 373
SignalTools SurroundScope . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 373
SignalTools PhaseScope . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 374
SignalTools Display Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 375
SignalTools Level Meters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376
Chapter 76. TL AutoPan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
TL AutoPan Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
Using TL AutoPan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 384
Part XII
Instrument Plug-Ins
Chapter 77. Boom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 389
Boom Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
Inserting Boom on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 394
Creating a Drum Pattern Using Boom. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Saving a Boom Pattern as a Preset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Playing with Patterns in Boom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Controlling Boom with MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 396
Playing Boom Patterns Using MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 397
Creating Boom Pattern Chains. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 398
Assigning MIDI Controllers to Boom Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 398
Chapter 78. Bruno and Reso . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 399
Bruno/Reso Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 399
Bruno/Reso DSP Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 400
Inserting Bruno/Reso onto an Audio Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 400
Playing Bruno/Reso . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 400
Using an External Key Input with Bruno/Reso . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 401
Bruno Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 402
Reso Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 407
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Chapter 79. Click . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
Click Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
Creating a Click Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
Chapter 80. DB-33 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 417
DB-33 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 417
Inserting DB-33 on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 422
Assigning MIDI Controllers to DB-33 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 422
Chapter 81. Mini Grand . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 423
Mini Grand Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 423
Inserting Mini Grand on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 426
Assigning MIDI Controllers to Mini Grand Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 426
Chapter 82. Structure Free . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 427
Structure Free Keyboard Section Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 427
Structure Free Patch List Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 429
Structure Free Main Page Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 431
Structure Free Browser Page Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 433
Using Structure Free . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 434
Assigning MIDI Controllers to Structure Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 436
Chapter 83. TL Drum Rehab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 437
TL Drum Rehab Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 438
TL Drum Rehab Controls and Displays Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 442
TL Drum Rehab Main Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 442
TL Drum Trigger Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 443
TL Drum Rehab Expert Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 448
Samples Panel Display and Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 452
TL Drum Rehab Preferences Panel Display and Controls. . . . . . . . . . . . . . . . . . . . . . . . . . 454
TL Drum Rehab Library Browser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 455
Chapter 84. TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 459
Configuring Pro Tools for Use with TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 459
TL Metro Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 460
Synchronizing TL Metro to Pro Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 461
Customizing TL Metro . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 462
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Chapter 85. Vacuum . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 465
Vacuum Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 466
Inserting Vacuum on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 472
Assigning MIDI Controllers to Vacuum Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 472
Chapter 86. Xpand!2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 473
Xpand!2 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 474
Xpand!2 Patch Edit Controls Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 477
Xpand!2 Play Patch Edit Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 477
Xpand!2 Arp Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 478
Xpand!2 Mod Patch Edit Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
Inserting Xpand!2 on a Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 481
Assigning MIDI Controllers to Xpand!2 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 481
Chapter 87. ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
ReWire Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 485
Using ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 485
MIDI Automation with ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 487
Quitting ReWire Client Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
Session Tempo and Meter Changes and ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
Looping Playback with ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
Automating Input Switching with ReWire . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
Chapter 88. Using the MIDI Learn Function on Avid Virtual Instruments . . . . . . . . . . . . . . 491
Part XIII
Other Plug-Ins
Chapter 89. BF Essentials Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 495
BF Essential Clip Remover . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 495
BF Essential Correlation Meter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 496
BF Essential Meter Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 496
BF Essential Noise Meter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 496
Chapter 90. Signal Generator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 497
Signal Generator Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 497
AudioSuite Processing with Signal Generator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 498
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Chapter 91. SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 499
Audio Replacement Techniques . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 499
SoundReplacer Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 500
Using SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 504
Getting Optimum Results with SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 505
Using the Audio Files Folder for Frequently Used SoundReplacer Files . . . . . . . . . . . . . . . 507
Chapter 92. Time Compression/Expansion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 509
Time Compression/ Expansion Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 509
Chapter 93. TL InTune . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 511
TL InTune Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 512
TL InTune Presets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 514
Using TL InTune . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 516
Chapter 94. TL MasterMeter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 517
TL Master Meter Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 518
Using TL MasterMeter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 522
TL MasterMeter Controls and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 522
Chapter 95. Trim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Trim Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Chapter 96. Other AudioSuite Plug-In Utilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 527
DC Offset Removal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 527
Duplicate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
Gain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
Invert. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
Normalize . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
Reverse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 530
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Part XIV
Eleven
Chapter 97. Eleven and Eleven Free . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 533
Chapter 98. Eleven Input Calibration and QuickStart . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 535
Connect your Guitar and Configure Source Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
Set Hardware and Levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
Set Up a Pro Tools Track . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 537
Set Up Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 538
Getting Started Playing Music with Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 539
Chapter 99. Using Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 541
Inserting Eleven on Tracks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 541
Adjusting Eleven’s Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 541
Using a Pro Tools Worksurface with Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 542
Using MIDI and MIDI Learn with Eleven . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 542
Eleven Settings (Presets) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 543
Master Section . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Amp Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 545
Eleven Amp Controls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 546
Eleven Cabinet Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 548
Eleven Cabinet Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 549
Tracks and Signal Routing for Guitar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 550
Blending Eleven Cabinets and Amps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 554
Eleven Tips and Suggestions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 558
Eleven Signal Flow Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 561
Part XV
Synchronic
Chapter 100. Synchronic Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 565
Chapter 101. Synchronic Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 567
Synchronic Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 567
Playing Synchronic RTAS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 568
Performance and Edit Modes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 571
Synchronic Performance Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 571
Synchronic Presets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 572
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Chapter 102. Using Synchronic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 575
Adjusting Synchronic Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 575
Synchronic Sound Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 576
Synchronic Sound Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 576
Synchronic Sound Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 577
Synchronic Playback Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 583
Synchronic Playback Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 583
Synchronic Playback Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 584
Synchronic Effect Module . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 592
Synchronic Effect Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 592
Synchronic Effect Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 593
Synchronic XFade Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 598
Synchronic MIDI Module Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 600
Synchronic MIDI Performance Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 600
Synchronic MIDI Edit Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 602
Synchronic Keyboard Focus Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 603
Chapter 103. Using Synchronic as an AudioSuite Plug-In. . . . . . . . . . . . . . . . . . . . . . . . . . . 605
Synchronic AudioSuite Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 605
Synchronic AudioSuite Workflow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 607
Chapter 104. Automating Synchronic RTAS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 611
Using Automation Playlists . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 611
Using MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Chapter 105. Synchronic Plug-In Settings. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 615
Imported Audio Stored with Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 615
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 617
Contents
xvii
xviii
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 both Pro Tools® and VENUE systems. These include plug-ins that come with your
Pro Tools or VENUE 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.
Avid Audio Plug-Ins
®
Avid provides a comprehensive set of digital signal processing tools for professional audio production with Pro Tools and VENUE systems. A set of
sound processing, effects, and utility plug-ins are
included with Pro Tools and VENUE systems.
Pro Tools systems also include a suite of virtual instrument plug-ins. Other Avid plug-ins are available for purchase or rental from the Avid store
(visit shop.avid.com, or, in Pro Tools, choose Marketplace > Plug-Ins).
Avid Audio Plug-Ins Included
with Pro Tools and Pro Tools HD
An extensive suite of Avid plug-ins are included
with Pro Tools, providing a comprehensive suite
of digital signal processing effects that include EQ,
dynamics, delay, and other essential audio processing tools.
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.
The following sound-processing, effects, and utility plug-ins are included with Pro Tools and
Pro Tools HD:
• AIR Chorus
• AIR Distortion
• AIR Dynamic Delay
• AIR Enhancer
• AIR Ensemble
• AIR Filter Gate
• AIR Flanger
• AIR Frequency Shifter
• AIR Fuzz-Wah
• AIR Kill EQ
• AIR Lo-Fi
• AIR MultiChorus
• AIR Multi-Delay
• AIR Nonlinear Reverb
Chapter 1: Audio Plug-Ins Overview
3
• AIR Phaser
• AIR Reverb
• 7 Band
• AIR Spring Reverb
• 2–4 Band
• AIR Stereo Width
• 1 Band
• AIR Talkbox
• Maxim™
• AIR Vintage Filter
• Mod Delay II
• Avid Channel Strip
• Mod Delay III
• Avid Down Mixer
• Pitch
• BF76 Compressor
• Pitch Shift
• BF Essentials utility plug-ins
• POW-r Dither
• Essential Clip Remover
• ReWire
• Essential Correlation Meter
• SansAmp PSA-1
• Essential Meter Bridge
• Signal Generator
• Essential Noise Meter
• SignalTools
• Click
• SurroundScope
• D-Fi plug-ins
™
• Lo-Fi
• PhaseScope
• TimeAdjuster
™
• Recti-Fi
™
• Time Compression/Expansion
• Sci-Fi
• Time Shift
• Vari-Fi™
• TL AutoPan™
• D-fx plug-ins
• TL InTune™
• Chorus
• TL MasterMeter™
• Flanger
• TL Metro™
• Multi-Tap Delay
• Ping-Pong Delay
• Dither
• D-Verb
• Dynamics III
• Trim
• Other AudioSuite Plug-In Utilities
• DC Offset Removal
• Duplicate
• Gain
• Compressor/Limiter
• Invert
• Expander/Gate
• Normalize
• De-Esser
• Reverse
• Eleven Free™ guitar amp modeling plug-in
4
• EQ III
Audio Plug-Ins Guide
Avid Virtual Instrument Plug-Ins Included with
Pro Tools and Pro Tools HD
• AIR Multi-Delay
The following virtual instrument plug-ins are also
included with Pro Tools, but require separate installation using the Avid Virtual Instruments installer (available on the Pro Tools DVD as well as
online):
• AIR Vintage Filter
• Boom— drum machine and sequencer
• Dither
• DB-33 — tonewheel organ emulator with
rotating speaker simulation
• Dynamics III
• Mini Grand — acoustic grand piano
• Structure Free — sample player
• AIR Phaser
• BF76 Compressor
• Click
• D-Verb
• Compressor/Limiter
• Expander/Gate
• De-Esser
• Vacuum — vacuum tube–modeled monophonic
synthesizer
• Eleven Free
• Xpand!2 — multitimbral synthesizer and sampler workstation
• 7 Band
• EQ III
• 2–4 Band
Avid Plug-Ins Included with
Pro Tools Express
A basic suite of Avid audio plug-ins are automatically installed with Pro Tools Express. These plugins provide an essential suite of digital signal processing effects that include EQ, dynamics, delay,
and other vital audio processing tools.
All of these plug-ins are automatically
installed when installing Pro Tools Express.
For more information, see the Pro Tools
Express Installation Guide.
• 1 Band
• POW-r Dither
• ReWire
• Time Compression Expansion
• Time Shift
• TL InTune digital tuner
• Other AudioSuite Plug-In Utilities
• DC Offset Removal
• Duplicate
• Gain
The following sound-processing, effects, and utility plug-ins are included with Pro Tools Express:
• Invert
• AIR Chorus
• Reverse
• Normalize
• AIR Distortion
• AIR Dynamic Delay
• AIR Flanger
• AIR Frequency Shifter
• AIR Lo-Fi
Chapter 1: Audio Plug-Ins Overview
5
Avid Virtual Instrument Plug-Ins Included with
Pro Tools Express
The following virtual instrument plug-ins are also
included with Pro Tools Express, but require separate installation using the Avid Virtual Instruments
Express installer (included with the Pro Tools Express installer download):
Additional Avid Audio Plug-Ins
The following plug-ins are available separately for
purchase and rental:
• BF-3A
• BF-2A
• Bruno™ & Reso™ cross-synthesis plug-ins
• Boom drum machine and sequencer
• Cosmonaut Voice
• Structure Free sample player
• DINR™ intelligent noise reduction
• Xpand!2 multitimbral workstation
• Eleven™ guitar amplifier modeling plug-in
Factory Installed Plug-Ins for
VENUE Systems
• Fairchild 660 and 670
VENUE systems, provide the following factory installed Avid Audio plug-ins:
• BF76 Compressor
• Dynamics III
• Impact®
• JOEMEEK SC2 Compressor
• JOEMEEK VC5 Meequalizer
• Moogerfooger plug-ins
• Moogerfooger Analog Delay
• Compressor/Limiter
• Moogerfooger Ring Modulator
• Expander/Gate
• Moogerfooger 12-Stage Phaser
• De-Esser
• Moogerfooger Lowpass Filter
• D-Verb
• Purple Audio MC77
• EQ III
• Reel Tape™ plug-ins:
• 7 Band
• Reel Tape Saturation
• 2–4 Band
• Reel Tape Delay
• 1 Band
• Reel Tape Flanger
• Mod Delay II
• Reverb One™
• Pitch
• ReVibe®
• Signal Generator
• Slightly Rude Compressor
• SignalTools
• Smack!™
• PhaseScope
• SoundReplacer™ drum and sound replacement
plug-in
• TimeAdjuster
• Trim
• Synchronic™ beat slicing and processing plug-in
• Tel-Ray Variable Delay
• TL Aggro™
• TL Drum Rehab™
6
Audio Plug-Ins Guide
• TL EveryPhase™
AAX Plug-Ins
• TL Space™ TDM and TL Space Native
AAX (Avid Audio Extension) plug-ins provide
real-time plug-in processing using host-based
(“Native”) or DSP-based (Pro Tools HD with Avid
HDX hardware accelerated systems only) processing. The AAX plug-in format also supports AudioSuite non-real-time, file-based rendered processing.
• Voce Spin
• Voce Chorus/Vibrato
• X-Form™ high-quality time compression and
expansion plug-in
Plug-In Formats
AAX plug-in files use the “.aaxplugin” file suffix.
There are three plug-in formats used in Pro Tools:
RTAS Plug-Ins
• AudioSuite™ plug-ins (non-real-time, file-based
processing)
• Native: real-time, host-based plug-ins:
• AAX Native plug-ins
• RTAS® plug-ins
• DSP: real-time, DSP-based plug-ins:
• AAX DSP plug-ins (Avid HDX only)
• 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.
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.
AudioSuite plug-in files may use either the
“.aaxplugin” or “.dpm” file suffix.
Chapter 1: Audio Plug-Ins Overview
7
TDM Plug-Ins
(Pro Tools|HD and VENUE Systems Only)
Conventions Used in
Pro Tools Documentation
TDM (Time Division Multiplexing) plug-ins provide real-time, DSP-based (“DSP”) processing
with Pro Tools HD software on Pro Tools|HD
hardware.
Pro Tools documentation uses the following conventions to indicate menu choices, keyboard commands, and mouse commands:
TDM plug-in files use the “.dpm” file suffix.
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.
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:
Convention
Action
File > Save
Choose Save from the
File menu
Control+N
Hold down the Control
key and press the N key
Control-click
Hold down the Control
key and click the mouse
button
Right-click
Click with the right
mouse button
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.
• Inserting plug-ins on tracks
• Plug-In Window controls
• Adjusting plug-in controls
Important Notices include information that
could affect your Pro Tools session data or
the performance of your Pro Tools system.
• Automating plug-ins
• Using side-chain inputs
• Using plug-in presets
Shortcuts show you useful keyboard or mouse
shortcuts.
• Clip indicators
Cross References point to related sections in
this guide and other Avid documentation.
8
Audio Plug-Ins Guide
System Requirements and
Compatibility for Plug-Ins
Contents of the Boxed
Version of Your Plug-In
To use Pro Tools plug-ins, you need the following:
If you bought your plug-in as a boxed version, it
includes the following:
• Any of the following systems:
• An Avid-qualified system running Pro Tools
or Pro Tools HD
• A qualified Avid VENUE system
(TDM only)
• A qualified Avid Media Composer® system
(AudioSuite and RTAS only)
• An Avid-qualified system and a third-party
software application that supports the AAX,
RTAS, TDM, or AudioSuite plug-in standards
• 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
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.
• 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 plug-ins, applications, and hardware.
News and Events Get the latest news from Avid or
sign up for a Pro Tools demo.
Chapter 1: Audio Plug-Ins Overview
9
10
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.
Plug-Ins Included with
Pro Tools
A suite of Avid audio effects and virtual instruments plug-ins are included with Pro Tools.
Avid Effects A set of free audio effects plug-ins
that can be installed with Pro Tools.
Factory-Installed 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.
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.
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:
Audio effects plug-ins included with
Pro Tools Express are installed automatically when you install Pro Tools Express.
The Pro Tools Express installer download
also includes a separate installer for Avid
Virtual Instruments Express plug-ins and
content.
Updating Older Plug-Ins
Because plug-in installers contain the latest versions of the plug-ins, use them to update any plugins you already own. When installing Pro Tools,
the Pro Tools installer automatically updates the
core Pro Tools plug-ins.
Be sure to use the most recent versions of
plug-ins. For more information, see the Avid
website (www.avid.com).
Chapter 2: Installing Plug-Ins
11
Using Pro Tools Plug-Ins with
Avid Media Composer
Installing Plug-Ins for VENUE
Systems
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 12, “Authorizing Paid Plug-Ins” on page 12, and “Removing Plug-Ins for Pro Tools” on page 14.
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 CD-ROM inserted into
the CD-ROM drive available on FOH Rack or Mix
Rack.
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).
For complete instructions on installing plugins for VENUE systems, see the documentation that came with your VENUE system.
Authorizing Paid Plug-Ins
Pro Tools plug-ins are authorized using the iLok
USB Smart Key (iLok), manufactured by PACE
Anti-Piracy.
• 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.
3
Follow the on-screen instructions to complete
the installation.
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 12).
iLok USB Smart Key
Not all Pro Tools plug-ins require authorization. For example, no authorization is
required for the free plug-ins included with
Pro Tools.
This key 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,
or purchase one separately.
12
Book Title
Authorizing Downloaded
Plug-Ins for Pro Tools
Authorizing Plug-Ins on VENUE
Systems
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.
After installing a plug-in on a VENUE system, the
system re-creates the list of available plug-ins.
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 plug-in.
For more information, visit the iLok website
(www.iLok.com).
Authorizing Boxed Versions of
Plug-Ins for Pro Tools
If you purchased a boxed version of software, it
comes with an Activation Code (on the included
Activation Card).
To authorize a plug-in using an Activation Code:
1
2
If you do not have an iLok.com account, visit
www.iLok.com and sign up for an account.
Transfer the license for your plug-in to your
iLok.com account by doing the following:
• Visit www.avid.com/activation.
• 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.
3
Transfer the licenses from your iLok.com account to your iLok USB Smart Key by doing the
following:
• Insert the iLok into an available USB port on
your computer.
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 au-
thorizations 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).
• Go to www.iLok.com and log in.
• Follow the on-screen instructions for transferring your licences to your iLok.
4
Launch Pro Tools.
5
If you have any installed unauthorized plug-ins
or software options, you are prompted to authorize them. Follow the on-screen instructions to
complete the authorization process.
Chapter 2: Installing Plug-Ins
13
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.
Mac OS X
To remove a plug-in on Mac:
1
Locate and open the Plug-Ins folder on your
Startup drive, which will be in one of the following locations:
• Library/Application Support/Digidesign/
Plug-Ins
• Library/Application Support/Avid/Audio/
Plug-Ins
2
Do one of the following:
• Drag the plug-in to the Trash and empty the
Trash.
• Drag the plug-in to the Plug-Ins (Unused) folder.
Windows
To remove a plug-in in Windows:
14
1
Choose Start > Control Panel.
2
Click Programs and Features.
3
Select the plug-in from the list of installed applications.
4
Click Uninstall.
5
Follow the on-screen instructions to remove the
plug-in.
Book Title
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
17
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: AAX, TDM, RTAS, and AudioSuite
• 2–4 Band: TDM and RTAS only
• 1 Band: AAX,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, multimono, or stereo plug-in.
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 can be operated from the following control
surfaces:
EQ III Configurations
• D-Command
The EQ III plug-in appears as three separate
choices in the plug-in insert pop-up menu and in
the AudioSuite menu:
• D-Control
• ProControl
• C|24
• Control|24
• Digi 003
• 1 Band (“1-Band EQ 3”)
• 2–4 Band (“4-Band EQ 3”)
• 7 Band (“7-Band EQ 3”)
• Digi 002
• Command|8
• EUCON™
• Mackie HUI-compatible controllers
Chapter 4: EQ III
19
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
20
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 Option-clicking (Mac) or Altclicking (Windows) 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 Control-dragging (Mac) or Start-dragging (Win-
dows) a control point up decreases the Q setting.
Control-dragging (Mac) or Start-dragging (Windows) a control point down increases 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 midrange frequencies. It is especially useful for adjusting limited bandwidth in order to solo and finetune 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 Control+Shift (Mac) or Start+Shift (Windows), 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
21
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 mid-range
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
frequencies centered around the Frequency setting.
Clip
Indicators
To switch an EQ III control out of Band-Pass mode:

Release Control+Shift (Mac) or Start+Shift
(Windows).
Controlling EQ III from a Control
Surface
EQ III can be controlled from any supported control surface, including our D-Control, D-Command, 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 plug-in is
being used.
22
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
Yellow Indicates pre-clipping levels, starting at
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.
–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
23
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
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.
Peak and Notch On Peak and Notch bands, the Q
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
24
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 colorcoded. 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
25
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
26
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 below
the Frequency setting at the selected slope while
letting all frequencies above pass through.
Low Pass Filter Attenuates all frequencies above
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
120 Hz to 20 kHz
Frequency Default
20 Hz
Frequency Default
20 kHz
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
High Notch Q Range
0.1 to 10.0
Low Notch Q Default
1.0
HIgh Notch Q Default
1.0
Chapter 4: EQ III
27
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.
Control
Value
Low Shelf EQ Boosts or cuts frequencies at and
Frequency Range
20 Hz to 500 Hz
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.
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
Low 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.
Low Shelf EQ
button
Q
control
Low Peak EQ
button
Q
control
Band
Enable
button
Band
Enable
button
Gain
control
Frequency
control
Gain
control
Frequency
control
Low Shelf EQ (left) and Low Peak EQ (right)
28
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:
Audio Plug-Ins Guide
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
29
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 frequencies
centered around the Frequency setting. The width
of the affected band is determined by the Q setting.
Gain
control
High Shelf EQ
button
Q
control
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
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)
30
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
31
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
selector
Q controls
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.
32
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
33
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
1 Band EQ set to Low Pass Filter
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
34
Audio Plug-Ins Guide
Chapter 5: JOEMEEK VC5 Meequalizer
The JOEMEEK VC5 Meequalizer is an EQ
plug-in that is available in AAX, 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.
JOEMEEK Meequalizer VC5 EQ
How the Meequalizer Works
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 in this device.
Mid and Mid Freq The Mid and Mid Freq controls
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
35
36
Audio Plug-Ins Guide
Chapter 6: Pultec Plug-Ins
Pultec plug-ins are a set of EQ plug-ins that are
available AAX, TDM, RTAS, and AudioSuite formats. The following plug-ins are included:
• Pultec EQP-1A (see “Pultec EQP-1A” on
page 37)
• Pultec EQH-2 (see “Pultec EQH-2” on page 38)
• Pultec MEQ-5 (see “Pultec MEQ-5” on
page 39)
Pultec EQP-1A
(AAX, TDM, RTAS, and AudioSuite)
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 real-world studio
equipment. And a clean and well-designed tube
amplifier provides a fixed amount of make-up
gain.
Pultec EQP-1A Controls
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 frequencies 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
37
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.
Consider anything that encourages very careful
and infrequent use of peaky boosts to be a Very
Good Thing.
Pultec EQH-2
(AAX, 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.
38
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 EQP1A, 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.
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
High Frequency Boost Section Boost mid and
high frequencies using the KCS (kilocycles per
second) and Boost knobs on the second row.
Low Frequency Peak Boost low frequencies (200,
300, 500, 700, 1000 Hz) using the upper left controls.
High Frequency Attenuate Section Cut high frequencies using the 10k Atten knob located at the
right side of the plug-in.
Mid Frequency Peak Boost mid-frequencies
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
(AAX, 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.
(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
39
40
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 AAX, TDM, RTAS, and AudioSuite formats.
Meticulously crafted to capture every nuance of
the legendary LA-2A tube studio compressor, BF2A 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 photo-conductive 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 (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
43
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 metering
To access the side-chain filter on-screen:
are provided. The Meter knob operates as follows:
1
Click the Plug-In Automation button in the
Plug-In window to open the Automation Enable
window.
2
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).
3
Click OK to close the plug-in automation window.
• 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.
44
Using the BF-2A Side-Chain
Filter
Audio Plug-Ins Guide
4
In the Edit window, do one of the following:
• Click the Track View selector and select SideChain Filter from the BF-2A sub-menu.
• Reveal an Automation lane for the track, click
the Automation Type selector and select SideChain Filter from the BF-2A sub-menu.
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:
1
Focus the BF-2A plug-in on your control surface.
2
Adjust the encoder or fader current targeting the
Side-Chain Filter parameter.
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
45
46
Audio Plug-Ins Guide
Chapter 8: BF-3A
The BF-3A is a vintage-style compressor plug-in
that is available in AAX, TDM, RTAS, and AudioSuite formats. BF-3A is based on the classic LA3A that adds a smoothness and sonic texture that
makes sounds jump right out of the mix.
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
How BF-3A Works
Designed and manufactured in the late 1960s, the
original LA-3A shares many components in common with the LA-2A compressor. Just like the LA2A, 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.
While the LA-2A’s gain comes from a tube amplifier, the LA-3A's gain comes from a solid-state
(transistor) amplifier. This gives the LA-3A 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.
Peak Reduction Peak Reduction controls the
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.
Output Gain Output Gain provides makeup gain to
make the signal louder after passing through the
peak reduction.
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.
Chapter 8: BF-3A
47
Meter Both Gain Reduction and Output metering
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.
48
Audio Plug-Ins Guide
Chapter 9: BF76
The BF76 is a vintage-style compressor plug-in
that is available in AAX, TDM, RTAS, and AudioSuite formats.
Modeled after the solid-state (transistor) 1176 studio compressor, BF76 preserves every sonic subtlety of this classic piece of studio gear.
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
BF76
How 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 enjoy this
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
49
Ratio The Ratio Push switches select the compres-
sion 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 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.” It’s also a
handy way to make quick headphone submixes
when tracking overdubs.
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.
50
Audio Plug-Ins Guide
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. With 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 with 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 (DSP, 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.
Channel Strip plug-in, Compressor/Limiter tab shown
Chapter 10: Channel Strip
51
Sections and Panes
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.
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
You can independently show or hide the Dynamics
and EX/Filters sections of the Channel Strip plugin to use less screen space. These sections are
shown by default.
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Pro Tools Reference Guide
Channel Strip plug-in, Dynamics section hidden
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.
 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.
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.
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).
Channel Strip plug-in, Side Chain Listen mode
enabled
Control-Shift-click (Mac) or Start-Shift-click
(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.
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.
 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.
Chapter 10: Channel Strip
53
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 postprocessing gain reduction meters.
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.
White Indicates full scale levels from 0 dB to
+6 dB.
Gain Reduction Meters
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.
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.
To Trim the input signal, do one of the following:

Click in the Input Trim field to type the desired
Trim value (–36.0 dB to +36.0 dB).

Click Trim and drag up or down to adjust the Input Trim setting.
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.
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:

54
Click the Phase Invert button so that it is highlighted. Click it again so that it is not highlighted to disable it.
Pro Tools Reference Guide
Toggling between Input and Gain Reduction meters
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:
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).
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 56.
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).

Click Vol and drag up or down to adjust the Output Volume setting.
Chapter 10: Channel Strip
55
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.
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 set the FX Chain:
1
Click the FX Chain show/hide button to reveal
the Process Order options.
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.
Dynamics
Showing the FX Chain Process Order
2
Click the desired effects chain ordering option
to select it. The available options include:
• EQ > FILT > DYN
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.
• EQ > DYN > FILT
• DYN > EQ > FILT
• FILT > DYN > EQ
3
Select PRE or POST to place the Output Volume control at the beginning or at the end of the
effects signal chain.
Dynamics section, All tab shown
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Pro Tools Reference Guide
Dynamics Graph
Dynamics Graph Gain Reduction
Resolution
Input signal
level (x-axis)
Graph
Resolution
toggle
Output signal
level (y-axis)
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.
To change the Dynamics Graph Gain Reduction
resolution:

Click the Graph Resolution toggle.
Using the Dynamics Graph to Adjust
Controls
Compressor/Limiter Threshold
Expander/Gate Threshold
Dynamics graph display
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:
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.
• Expander/Gate Ratio
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.
• Compressor/Limiter Threshold
Use this graph as a visual guideline to see how
much dynamics processing you are applying to the
incoming audio signal.
• Expander/Gate Knee
• Expander/Gate Threshold
• Gate Depth
• Hysteresis
• Compressor/Limiter Ratio
• Compressor/Limiter Knee
• 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).
Chapter 10: Channel Strip
57
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.
Dynamics section, Expander/Gate tab
Hold
Threshold
The Dynamics Graph display shows the threshold
as a vertical line.
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.
Attack
Release
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 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.
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.
Ratio
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.
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Pro Tools Reference Guide
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
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
Compressor/Limiter Ratio set to LMTR
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.
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.
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.
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.
Chapter 10: Channel Strip
59
Release
Side Chain Processing Controls
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.
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 sidechain. Compressor/Limiter and Expander/Gate
processing features external key capabilities and
filters for the side-chain.
Knee
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.
Gain
The Gain control lets you boost overall output gain
to compensate for heavily compressed or limited
signals.
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Pro Tools Reference Guide
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.
Source
The Source selector lets you set the source for side
chain processing: Internal, Key, or All-Linked.
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.
Filter Frequency
Key If Key is selected, the plug-in uses the ampli-
Filter Type
tude 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.
Four Filter Type options are available for sidechain 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.
The Filter Frequency control lets you set the frequency for the selected Filter Type.
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.
Notch Select the Notch option to apply a notch filter to the side-chain processing at the selected frequency.
Band Pass Select the Band Pass option to apply a
band pass filter to the side-chain processing at the
selected frequency.
Side Chain Processing Graph
Selecting the Source setting for Side Chain
processing
The Side Chain Processing Graph display shows
the frequency curve for the selected Filter Type at
the selected Filter Frequency.
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 side-
chain processing according to the detected average
amplitude.
Chapter 10: Channel Strip
61
EQ/Filters
The EQ/Filters section of Channel Strip provides a high-quality 4-band parametric equalizer for adjusting
the frequency spectrum of audio material.
EQ/Filters Graph
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).
Frequency
(x-axis)
Graph Resolution
toggle
Gain
(y-axis)
Filter control point
EQ control point
EX/Filters section, High Mid Frequency tab shown
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.
To change the Frequency Graph Gain resolution:

62
Click the Graph Resolution toggle.
Pro Tools Reference Guide
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.
Option-Shift-click (Mac) or Alt-Shift-click
(Windows) an EQ control point to invert its
Gain setting.
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 Control-click (Mac) or
Start-click (Windows) and drag a control
point up or down to increase or decrease
the Q 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.
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.
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.
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64
Low Mid Frequency EQ Controls
High Mid Frequency EQ Controls
EQ/Filters section, Low Mid Frequency tab
EQ/Filters section, High Mid Frequency tab
The LMF tab provides controls for the low mid frequency band of the EQ. This band is a peak EQ.
The HMF tab provides controls for the high mid
frequency band of the EQ. This band is a peak EQ.
Frequency
Frequency
The Frequency control lets you set the center frequency for the peak low mid frequency band.
The Frequency control lets you set the center frequency for the peak high mid frequency band.
Gain
Gain
The Gain control lets you boost or attenuate the
corresponding frequencies for the low mid frequency band.
The Gain control lets you boost or attenuate the
corresponding frequencies for the high mid frequency band.
Q
Q
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.
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.
Pro Tools Reference Guide
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:
High Pass, Low Pass, Band Pass, and Notch.
Frequency
The Frequency control lets you set the center frequency for the high frequency band (Peak or Shelf
EQ).
Frequency
The Frequency control lets you set the center frequency for the selected Filter Type (from 20 Hz to
21.0 kHz).
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.
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.
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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 AAX, TDM, RTAS, and AudioSuite formats:
Dynamics III Shared Features
and Controls
• Compressor/Limiter (see “Compressor/Limiter
III” on page 70)
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.
• Expander/Gate (see “Expander/Gate III” on
page 73)
• De-Esser (see “De-Esser III” on page 76)
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 greaterthan-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 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 76
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)
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67
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 plug-in
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 79.
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.
Input (left) and Output (right) meter buttons
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Audio Plug-Ins Guide
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)
Use this graph as a visual guideline to see how
much dynamics processing you are applying.
Threshold
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.
Output signal
level (y-axis)
Input signal
level (x-axis)
LFE Enable button (Compressor/Limiter III shown)
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.
Dynamics graph display
The Compressor/Limiter and Expander/Gate plugins also feature an animated, multi-color cursor in
their gain transfer curve displays.
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.
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.
Chapter 11: Dynamics III
69
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 77 for information on using
the
De-Esser’s graph display.
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 Side-Chain
Processing” on page 81.
Compressor/Limiter III
The Compressor/Limiter plug-in applies either
compression or limiting to audio material, depending on the ratio of compression used.
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.
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.
Of course, compression has many creative uses
that break these rules.
About Limiting
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 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.
Compressor/Limiter III
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Audio Plug-Ins Guide
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 67 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.
Threshold arrow on input meter
The Dynamics Graph display also shows the
threshold as an orange vertical line.
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 69.
Threshold indicator on Dynamics Graph display
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.
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).
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71
Compressor/Limiter III Attack
Control
Compressor/Limiter III Knee
Control
The Attack control sets the attack time, or the rate
at which gain is reduced after the input signal
crosses the threshold.
The Knee control sets the rate at which the compressor reaches full compression once the threshold has been exceeded.
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.
As you increase this control, it goes from applying
“hard-knee” compression to “soft-knee” compression:
This control ranges from 10 s (fastest attack time)
to 300 ms (slowest attack time).
Compressor/Limiter III Release
Control
• 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.
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).
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).
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Audio Plug-Ins Guide
Compressor/Limiter III Gain
Control
Expander/Gate III
The Gain control lets you boost overall output gain
to compensate for heavily compressed or limited
signals.
The Expander/Gate plug-in applies expansion or
gating to audio material, depending on the ratio
setting.
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 sidechain 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 frequency-sensitive. See “Dynamics III Side-Chain Input” on page 78.
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.
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.
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73
Expander/Gate III
Expander/Gate III Threshold
Control
The Input and Output meters show peak signal levels before and after dynamics processing. See “Dynamics III Levels Section” on page 67 for more information.
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.
Input/Output Level Meters
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 69.
An orange arrow on the Input meter indicates the
current threshold, and can also be dragged up or
down to adjust the threshold setting.
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.
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Audio Plug-Ins Guide
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).
Expander/Gate 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. Use this along with the Ratio
setting to control how soft the Expander’s gain reduction curve is.
This control ranges from 10 s (fastest attack time)
to 300 ms (slowest attack time).
Expander/Gate III Hold Control
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.
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).
Expander/Gate III Range
Control
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 –80 dB (lowest depth) to
0 dB (highest depth).
Expander/Gate III Side-Chain
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 sidechain 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 frequency-sensitive. See “Dynamics III Side-Chain Input” on page 78.
This control ranges from 5 ms (shortest hold) to
4 seconds (longest hold).
Chapter 11: Dynamics III
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.
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.
Input
meter
Output
meter
Gain
Reduction
meter
De-Esser III
Using De-Essing Effectively
De-Esser III I/O Meter display
To use de-essing most effectively, insert the DeEsser after compressor or limiter plug-ins.
Input and Output Meters
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 de-essing 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.
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Audio Plug-Ins Guide
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 Listen Control
When enabled, the Listen button lets you monitor
the sibilant peaks used by the De-Esser as a sidechain 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.
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
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.
Gain
(y-axis)
Range
This control ranges from –40 dB (maximum de-essing) to 0 dB (no de-essing).
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.
Frequency
(x-axis)
De-Esser graph display
Chapter 11: Dynamics III
77
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 sidechain. The Compressor/Limiter and Expander/Gate plug-ins feature external key capabilities and filters for the 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.
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.
Compressor/Limiter and Expander/Gate Side-Chain
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Audio Plug-Ins Guide
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.
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.
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 sidechain. 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.
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.
Low Pass filter
HF and LF Filter Enable buttons
Chapter 11: Dynamics III
79
Dynamics III Side-Chain HF Frequency
Control
The HF frequency control sets the frequency position for the Band Pass or Low Pass filter, and
ranges from 80 Hz to 20 kHz.
High Pass Filter Makes triggering of dynamics
processing more sensitive to frequencies above the
Frequency setting rolling off at a slope of 12 dB
per octave.
High Pass filter
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.
Dynamics III Side-Chain LF Frequency
Control
The Frequency control sets the frequency position
for the Band-Pass or High Pass filter, and ranges
from 25 Hz to 4 kHz.
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.
LF frequency controls
Band-Pass filter
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Audio Plug-Ins Guide
Using Dynamics III Key Input for
Side-Chain Processing
4
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.
To use a filtered or unfiltered external key input to
trigger dynamics processing:
5
Click the Key Input selector and select the input
or bus carrying the audio from the reference
track or external audio source.
Begin playback. The plug-in uses the input or
bus that you chose as an external key input to
trigger its effect.
6
Adjust the plug-in’s Threshold (Thresh) control
to fine-tune external key input triggering.
7
Adjust other controls to achieve the desired effect.
1
Using a Filtered Input Signal for
Side-Chain Processing with
Dynamics III
Selecting a Key Input
2
Click External Key to activate external sidechain processing.
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).
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
Side-Chain Listen
Chapter 11: Dynamics III
81
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
82
4
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.
5
Begin playback. The plug-in uses the filtered input signal to trigger dynamics processing.
6
To fine-tune side-chain triggering, adjust the
plug-in controls.
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 AAX, TDM,
RTAS, and AudioSuite formats. The following
plug-ins are included:
• Fairchild 660 (see “Fairchild 660” on page 83)
• Fairchild 670 (see “Fairchild 670” on page 85)
Fairchild 660
(AAX, TDM, RTAS, and AudioSuite)
Re-introducing the undisputed champion in price,
weight, and performance: the $35,000, one-hundred pound, Fairchild 660.
Avid’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
Fairchild 660
Chapter 12: Fairchild Plug-Ins
83
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
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.
unit and the compression threshold, just like the Input control on an 1176. Full clockwise is loudest.
Bonus Settings
Threshold Threshold adjusts the gain to the side-
chain, 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 custom presets.
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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Audio Plug-Ins Guide
Fairchild 670
(AAX, TDM, and RTAS)
Avid’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. Both a Fairchild 660 and a
Fairchild 670 were modeled 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 custom presets. See “Fairchild 670” on
page 85 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
To exactly match the settings between channels,
hold down the Shift key while adjusting a control.
This is useful when trying to preserve the existing
Left/Right balance on stereo material.
Fairchild 670
Chapter 12: Fairchild Plug-Ins
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86
Audio Plug-Ins Guide
Chapter 13: Impact
Impact is a high-quality AAX (DSP, Native, and
AudioSuite) and TDM dynamics processing plugin.
The Impact plug-in provides critical control over
the dynamic range of audio signals, with the look
and sound of a mixing console’s stereo-bus compressor.
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.
The TDM version of Impact requires one or
more Pro Tools|HD Accel cards.
Impact Controls
Impact Ratio Control
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 Attack 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.
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.
Impact plug-in
Chapter 13: Impact
87
Impact Release Control
Impact Ext Control (Side-Chain)
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 low-volume 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.
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.
Impact Make-up Control
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 MakeUp control determines master output levels for all
channels. The range of this control is from 0 dB of
attenuation to +40 dB of gain.
Applying large amounts of Make-Up gain
will boost the level of any noise or hiss
present in audio material, making it more
audible.
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Audio Plug-Ins Guide
See “Using the Impact Compressor” on
page 89 for instructions on setting up and using a key input.
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 side-chain processing). This is useful for fine-tuning Impact’s
compression settings to the Key Input.
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.
Impact Meters
Side-Chain Processing
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.
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 side-chain processing.
A red clip indicator appears at the top of each meter. Clicking a clip indicator clears it. Alt-clicking
(Windows) or Option-clicking (Mac) clears the
clip indicators on all channels.
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 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.
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.
Chapter 13: Impact
89
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.
2
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.
3
To activate external side-chain processing, click
Ext.
4
Begin playback. Impact uses the input or bus
that you selected as a Key Input to trigger its effect.
5
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).
Remember to disable Listen to resume
normal plug-in monitoring.
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6
Adjust Impact’s Threshold parameter to finetune Key Input triggering.
7
Adjust other parameters to achieve the desired
effect.
Audio Plug-Ins Guide
Chapter 14: JOEMEEK SC2 Compressor
The JOEMEEK SC2 Compressor is a dynamics
processing plug-in is available in AAX, 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.
Slope Slope is similar to the compression ratio
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.
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, the 5 setting found on the later-model JOEMEEK SC2.2 were added. 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.
Release Release sets the time during which signal
returns to normal after compression. With longer
release times, the compression is less noticeable.
Chapter 14: JOEMEEK SC2 Compressor
91
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.
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.
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Audio Plug-Ins Guide
Attack/Release Times
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 AAX
(DSP, Native, and AudioSuite), TDM, RTAS, and
AudioSuite formats. Maxim is ideal for critical
mastering applications, as well as standard peaklimiting tasks.
Dither for noise shaping during the final mixdown.

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 randomaccess 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.
On Pro Tools|HD systems, the multichannel
TDM version of Maxim is not supported at
192 kHz. Use the multi-mono TDM or RTAS
version instead.
Maxim (AAX version)
Maxim features include:
“Perfect attack-limiting” and look-ahead analysis accurately preserve 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 leveloptimized for recording.
Maxim (TDM and RTAS version)
Chapter 15: Maxim
93
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.
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.
• Adjusting the dynamic range of an entire final
mixdown for premastering purposes
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.
• Adjusting the dynamic range of individual instruments for creative purposes
How Maxim Differs From Conventional Limiters
Limiting has two main uses in the audio production cycle:
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.
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Limiting Individual Instruments
Audio Plug-Ins Guide
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 AAX DSP version of Maxim introduces
1033 samples of delay at 48 kHz, and the
AAX Native version introduces 2049 samples
of delay at 48 kHz. 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
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.
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.
Chapter 15: Maxim
95
By dragging the Threshold slider downwards, you
can visually adjust the level at which limiting will
occur. Maxim displays the affected range in orange.
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.
Maxim Attenuation Meter
dB level of
waveform
peaks
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.
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Audio Plug-Ins Guide
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.
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.
Maxim Mix Slider
Maxim Noise Shaping Control
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.
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.
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.
Maxim Dither 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.
Maxim Bit Resolution Button
These buttons select dither bit resolution. In general, set this control to the maximum bit resolution
of your destination media.
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.

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.
Chapter 15: Maxim
97
Using Maxim
Following are suggestions for using Maxim most
effectively.
To use Maxim:
98
1
Insert Maxim on the desired track.
2
Select the portion of the track containing the
most prominent audio peaks.
3
Loop playback and look at the data displayed by
the Histogram and Attenuator meter.
4
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.
5
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.
6
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.
7
Use the Bypass button to compare the processed
and unprocessed sound and to check if the results are acceptable.
8
Avoid pumping effects with heavier limiting by
setting the Release slider to longer values.
9
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.
Audio Plug-Ins Guide
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.
Maxim and Mastering
If you intend to deliver audio material as a 32-bit
floating point or 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 AAX, 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 Purple Audio MC77 preserves every audio nuance and sonic
subtlety of the classic originals.
How the Purple Audio MC77 Works
Purple Audio MC77 has controls identical in name
to those of the BF76, and which function similarly.
For more information, see Chapter 9, “BF76.”
The Purple Audio MC77
Chapter 16: Purple Audio MC77
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Audio Plug-Ins Guide
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 a completely
custom designed compressor. 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 simple to use
controls with the highest (and rudest) quality 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 clock-
wise is fastest and provides the most “pump.”
Rudeness Affects the sound of the compression
action.
Slightly Rude/Super Rude Switch Affects the
sound of the compression action.
Slightly Rude Compressor
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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 Fairchild 660 (or 670) or 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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Chapter 18: Smack!
Smack! is a dynamics processing plug-in that is
available in AAX, TDM, RTAS, and AudioSuite
formats.
Smack! provides support for 192 kHz, 176.4 kHz,
96 kHz, 88.2 kHz, 48 kHz, and 44.1 kHz sessions.
Smack! provides support for mono, stereo, and all
Pro Tools-supported multichannel audio formats.
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.
Smack! Plug-In (TDM version shown)
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.
• 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.
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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-electrical-based 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.
,
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.
Warm Mode Button
Enable the Warm button for compression that is
based on Norm mode, but which has program-dependent 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.
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.
Setting the Input and Output controls to 5 is
equal to unity gain at a compression ratio of
1:1.
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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.
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.
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.)
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.
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! 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.
As you increase the Ratio control, Smack! goes
from applying “soft-knee” compression to “hardknee” compression, as follows:
When you apply Smack! to stereo or multichannel
tracks, the Output control determines master output levels for all channels.
• With soft-knee compression, gentle compression begins and increases gradually as the input
signal approaches the threshold. This creates
smoother compression.
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.
• 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.
Setting the Input and Output controls to 5 is
equal to unity gain at a compression ratio of
1:1.
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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 frequency-sensitive.
See “Using the Smack! Side-Chain Input” on
page 108 for more information on using the
Side-Chain EQ on a Key Input.
Band-Emphasis Side-Chain EQ
Combined Enables the High Pass and peak set-
tings simultaneously to make the compressor's detector more sensitive to high frequencies and less
sensitive to low frequencies.
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.
Combined Side-Chain EQ
Off Disables the Side-Chain EQ control.
High Pass Side-Chain EQ
Band-Emphasis Makes the compressor's detector
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.
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Audio Plug-Ins Guide
Smack! Distortion Control
Smack! VU Meter
When enabled, Distortion adds subtle second-order and third-order harmonic distortion to the output signal.
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.
• 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.
Input
Clipping
indicator
Internal
Clipping
indicator
Meter Mode
button
Output
Clipping
indicator
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-
Input meter
Gain meter
Output
meter
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.
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.
This is especially useful for removing “thumps” or
“pops” from vocals, bass, or kick-drums.
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 applied
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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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.
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 greaterthan-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.
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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.
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.
To use an external Key Input to trigger
compression:
1
Insert Smack! on a track you want to compress
using external side-chain processing.
2
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.
The Key Input must be monophonic.
3
In the track that you are compressing, click the
instance of Smack! in the Inserts pop-up menu.
4
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.
5
Begin playback. Smack! uses the input or bus
that you selected as a Key Input to trigger its effect.
6
To fine-tune the amount of compression, adjust
the send level from the Key Input track.
When you are using a Key Input to trigger
compression, the Input control has no effect
on the amount of compression.
7
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).
8
Adjust other parameters to achieve the desired
effect.
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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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111
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 side-chain 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.
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.
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113
Post Gain
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.
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.
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 breathing. For example, Bass Compensation sounds great on bass guitar or when you
have TL Aggro on your master fader as stereo bus
compressor.
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Audio Plug-Ins Guide
Tube Drive control
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
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.
TL Aggro provides LED and Needle meters
Using the TL Aggro
Side-Chain Input
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.
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.
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.
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 limited 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.
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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
Frequency Shifter Controls
down, and the two shifted signals are heard simultaneously.
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 (pitch-shifted) 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.
Chapter 20: AIR Frequency Shifter
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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 vari-speeding
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.
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.
Chapter 21: Pitch
121
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. Alt-clicking (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 4-octave range of adjustment.
–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.
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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 transposi-
tion 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 plug-in 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 maximum
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.
Chapter 21: Pitch
123
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Audio Plug-Ins Guide
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.
Pitch Shift plug-in
Crossfade Use this to manually adjust crossfade
length in milliseconds to optimize performance of
the Pitch Shift plug-in according to the type of audio material you are processing. This plug-in
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 require smaller
Chapter 22: Pitch Shift
125
crossfades. Long crossfade times may oversmooth 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 plug-
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).
in’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.
Detune Provides finer adjustment of the frequency
of the reference tone in cents (100ths of a semitone).
Use lower values when processing lower frequency audio material. Use higher values when
processing higher frequency audio material.
Level Adjusts the volume of the reference tone
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 effect 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.
126
Pitch Shift Reference Pitch
Controls
Audio Plug-Ins Guide
in dB.
Using Reference Pitch
To use Reference Pitch:
1
Select the audio material you want to use as a
pitch reference. Click the preview button to begin playback of the selected audio.
2
Click the Reference Pitch button to activate the
reference sine wave tone.
3
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.
4
Select the audio material to be pitch shifted.
5
Adjust the Coarse and Fine Pitch Shift controls
to match the pitch of the audio playback to the
reference pitch.
6
Click Render to apply pitch shift to the
selection.
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 se-
Time Shift controls in the interface for are organized in the following four sections:
lect 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 For-
mant 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
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 fre-
quency 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.
complex sounds (such as a multipart musical selection).
Rhythmic Select Rhythmic for processing percus-
The range pop-up menu is unavailable in
Rhythmic mode and Varispeed mode.
Gain
sive 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
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 ex-
pansion 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 fol-
lower 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.
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.
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.
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 percent-
age. 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.
4
Select the desired preset setting from the Default Settings pop-up menu.
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
To change the time of a selected audio clip:
1
Select AudioSuite > Pitch Shift > Time Shift.
2
Select the Audio Mode appropriate to the type
of material you are processing (Monophonic,
Polyphonic, or Rhythmic).
3
In Monophonic or Polyphonic mode, select the
appropriate Range for the selected material
(Low, Mid, High, or Wide).
4
If compressing the duration of the selection, attenuate the Gain control as necessary.
5
If using Monophonic mode, adjust the Formant
Shift control as desired.
6
If using Polyphonic or Rhythmic mode, adjust
the Transient controls as desired.
7
Make sure Pitch Shift is set to 100% (unless you
also want to change the pitch of the selection).
8
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.
9
4
If transposing the pitch of the selection up, attenuate the Gain control as necessary.
5
If using Monophonic mode, adjust the Formant
Shift control as desired.
6
If using Polyphonic or Rhythmic mode, adjust
the Transient controls as desired.
7
Make sure Time Shift is set to 0% (unless you
also want to change the duration of the section).
8
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.
9
Click Render.
Changing the Time and Pitch
Using Time Shift
To change the time and pitch of a selected audio
clip:
1
Select AudioSuite > Pitch Shift > Time Shift.
2
Select Varispeed from the Audio Mode pop-up
menu.
3
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.
4
Click Render.
Click Render.
Changing the Pitch Using Time
Shift
To change the pitch of a selected audio clip:
1
Select AudioSuite > Pitch Shift > Time Shift.
2
Select the Audio Mode appropriate to the type
of material you are processing (Monophonic,
Polyphonic, or Rhythmic).
3
In Monophonic or Polyphonic mode, select the
appropriate Range for the selected material
(Low, Mid, High, or Wide).
Using the Monophonic, Polyphonic, or
Rhythmic modes, you can adjust both the
Time Shift and Pitch Shift controls independently before processing.
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 a non-real-time AudioSuite (AAX)
plug-in that 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:
Vari-Fi Controls
Vari-Fi Change Controls
Slow Down
• Speed up from a complete stop to normal speed
When selected, 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. This effect does not change the duration
of the audio selection.
• Slow down to a complete stop from normal
speed
Speed Up
When selected, 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. This effect does not change the duration
of the audio selection.
Vari-Fi Range Controls
Vari-Fi
The Range setting determines the duration of the
rendered clip in relation to the processing.
Partial
When the Partial option is selected, the length of
the audio selection is retained when rendering the
AudioSuite effect. This is useful for rendering the
effect in place (especially if the selection is constrained by the grid or by adjacent clips).
Chapter 24: Vari-Fi
135
When this option is enabled, processing is applied
to only two-thirds of the selection so that the resultant rendering maintains the original duration of
the selection.
Full
When the Full option is selected, all audio in the
current Edit selection is processed and rendered.
The resulting rendering is 150% the duration of the
Edit selection. The selection start point does not
change, but the rendered clip extends beyond the
end of the Edit selection.
This can be useful if the last third (for speeding up)
or the first third (for slowing down) of the Edit selection needs to be heard in the rendered effect.
Vari-Fi Volume Ramp Controls
On
When the On option is selected, a fade-out is applied if the Slow Down option is selected or a fade
in is applied if the Speed Up option is selected.
Off
When the Off option is selected, no fade-in or fadeout is applied in the rendered Edit selection.
This can result in a more pronounced “tape-stop”
or “tape-start” effect and can also be useful for preserving the dynamic level at the end of the Edit selection when the Slow Down option is selected, or
the beginning of the selection when the Speed Up
option is selected.
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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 highquality Polyphonic setting is used.
Monophonic Use for processing monophonic
sounds (such as a vocal melody).
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 clipping 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
X-Form 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.
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).
4x Lets you apply Time Shift, Pitch Shift, and 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).
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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). XForm also lets you transpose the formant shape independently of the fundamental frequency.
X-Form plug-in, Pitch section
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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 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.
To enable or disable formant shifting:

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.
X-Form AudioSuite Input
Modes
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.
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 supports the Pro Tools AudioSuite Input
Mode selector for use on mono or multi-input processing.
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%.
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.
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.
Refer to the Pro Tools Reference Guide for
more information about the TCE Trim tool.
To select X-Form 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
Digidesign X-Form.
4
Select the desired preset setting from the Default Settings pop-up menu.
5
Click OK.
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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 time of a selected audio clip:
142
1
Select AudioSuite > Pitch Shift > X-Form.
2
Select the Audio Type appropriate to the type of
material you are processing (Monophonic or
Polyphonic).
3
If compressing the duration of the selection, attenuate the Gain control as necessary.
4
Adjust the Transient controls as desired.
5
Enable the desired Range button (2x, 4x, or 8x)
to set the possible range for time change.
6
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.
7
Click Render.
Audio Plug-Ins Guide
To change the pitch of a selected audio clip:
1
Select AudioSuite > Pitch Shift > X-Form.
2
Select the Audio Type appropriate to the type of
material you are processing (Monophonic or
Polyphonic).
3
If transposing the pitch of the selection up, attenuate the Gain control as necessary.
4
Adjust the Transient controls as desired.
5
Enable the desired Range button (2x, 4x, or 8x)
to set the possible range for pitch change.
6
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.
7
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
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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.
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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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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 PreDelay 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.
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.
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149
Mix
Large Studio Simulates a large, live, empty room.
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.
Scoring Stage Simulates a scoring stage in a
Reverb Early Reflections
Section Controls
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.
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.
Type
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.
Medium Chamber Simulates a bright, medium-
sized room.
Large Chamber Simulates a bright, large-sized
room.
Small Studio Simulates a small, live, empty room.
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Audio Plug-Ins Guide
medium-sized hall.
Philharmonic Simulates the space and ambience
of a large, symphonic, concert hall.
Concert Hall Simulates the space and ambience of
a large 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
effect.
Vintage 2 Simulates a vintage digital reverb
effect.
Spread
Controls the length of the early reflections.
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 high-range 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.
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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 high-range frequency bands.
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Audio Plug-Ins Guide
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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Audio Plug-Ins Guide
Chapter 29: D-Verb
D-Verb is a studio-quality reverb plug-in that is
available in AAX, 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. Note that the AAX version of
D-Verb has a different user interface from the
TDM and RTAS versions, but most all of the controls and parameter values are the same (with a few
exceptions noted below).
Input Level Meters
(AAX Version Only)
Input meters indicate the input levels of the dry audio source 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.
D-Verb plug-in (AAX)
Output Level Meter
Output meters indicate the output levels of the processed signal. With stereo inserts, the TDM and
RTAS versions of D-Verb provide only a single
meter that 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. You may want
to monitor your send or insert signal levels closely
to help prevent this from happening.
D-Verb plug-in (TDM and RTAS)
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155
Clip Indicator (TDM and RTAS Only) The Clip indicator shows if clipping has occurred. It is a cliphold 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.
Gain and Input Level Controls
The AAX version of D-Verb provides a Gain control above the Input Level meter to let you adjust
the input gain. The TDM and RTAS versions of DVerb provide an Input Level slider to adjust the input volume of the reverb to prevent the possibility
of clipping and/or increase the level of the processed signal.
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%.
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.
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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.
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 natural
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.
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.
Diffusion Control
Hi Frequency Cut
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.
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).
Decay Control
Low Pass Filter
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.
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).
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. PreDelay 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.
Selections for D-Verb
AudioSuite Processing
Because AudioSuite D-Verb 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.
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Chapter 30: Reverb One
Reverb One is a world-class reverb processing
plug-in available in AAX (DSP, Native, and AudioSuite) and TDM plug-in formats. It provides
the highest level of professional sonic quality and
reverb-shaping control.
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.
Reverb One (AAX DSP version)
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159
Reverb One (TDM version)
Reverb One features include:
• Editable Reverb EQ graph
• Editable Reverb Color graph
• Reverb Contour graph
• Dynamic control of reverb decay
• Chorusing
• Early reflection presets
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 plugin such as Reverb One, you are artificially creating
a sound space with a specific acoustic character.
• Extensive library of reverb presets
• Supports 44.1 khz, 48 kHz, 88.2 kHz, 96 kHz,
176.4 kHz, and 192 kHz processing.
The TDM version of Reverb One does not
fully support sample rates above 96 kHz. For
sessions with a sample rate greater than
96 kHz, the TDM version of Reverb One
downsamples and upsamples accordingly.
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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
Reflected Sound
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.
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 pre-delay.
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 acoustic environment, this could produce the warm sonic
characteristics we associate with reverberation, or
it could produce echoes or other unusual sonic
characteristics.
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 natural-sounding
reverb effects can be created and applied in the Pro
Tools mix environment.
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.
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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 166.
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.
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.
100% Wet
Toggles the Wet/Dry control between 100% wet
and the current setting.
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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.
Decay Ratio
Chorus Controls
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.
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.
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.
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.
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.
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 side-band harmonics
and change the reverb’s tone color, producing
some very interesting special effects.
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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.
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.
Level
Controls the output level of the reverb tail. When
set to 0%, the reverb effect consists entirely of the
early reflections (if enabled).
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.
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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. Predelay 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 PreDelay time to a beat interval such as 1/8,
1/16, or 1/32 notes.
Reverb One Early Reflection
Controls
ER Settings
The Early Reflections section has controls for the
various early reflection elements, including ER
setting, level, spread, and delay.
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.
Calculating Early Reflections
Early reflection presets include:
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 listener. 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.
• 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.
• 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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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.
Reverb One Graphs
Spread
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.
Reverb One EQ graph controls (AAX version)
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.
Reverb One graph controls (TDM version)
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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.
Band Cut/Boost
HF Cut/HF Damp
To select the EQ or Color graph for editing
(AAX version only):

Select the EQ icon or the Color icon.
Frequency/Crossover
Frequency/Crossover
Adjusting graph controls (TDM version)
Reverb EQ Graph
Reverb One (AAX version), EQ icon selected
To cut or boost a particular band:

Drag a Band Cut/Boost breakpoint up or down.
To adjust frequency or crossover:

Drag a Frequency/Crossover slider right or left.
To adjust high-frequency cut or damp:

Drag the HF Cut/HF Damp control point right
or left.
Band Cut/Boost
HF Cut/HF Damp
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.
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.
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/Crossover
Frequency/Crossover
Adjusting graph controls (AAX version)
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.
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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.
Set the crossover to 500 Hz to boost low frequencies most effectively. Set it to 1.5 kHz to cut low
frequencies most effectively.
Crossover Sliders Sets the frequency boundaries
between the low, mid, and high frequency ranges
of the reverberation filter.
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 settings,
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.
Reverb Contour Graph
The Reverb Contour graph displays the envelope
of the reverb, as determined by the early reflections and reverb tail.
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 high-frequency 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.
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Reverb Contour graph (AAX version)
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.
Tool Tips (AAX Version Only)
To use tool tips, move the cursor over the name of
any control or parameter and an explanation appears as a tool tip.
Online Help (TDM Version Only)
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 (TDM version only)
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.
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Chapter 31: ReVibe
ReVibe is 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 and ReVibe II provide essentially
the same controls with the same parameter
values.
Revibe requires one or more HD Accel cards.
ReVibe makes it possible to model extremely realistic acoustic spaces and place audio elements
within them in a Pro Tools mix.
ReVibe plug-in
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171
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
Plug–in Insert Format
Mono
Stereo
LCR
LCRS
Quad
5.0
5.1
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Mono
Stereo
LCR
•
LCRS
•
Quad
•
5.0
•
5.1
Adjusting ReVibe Parameters
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.
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To cut or boost a particular EQ band:

Drag a control point up or down.
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.
Editing Graph Display Parameters with a Mouse
Editing Parameters with a Computer Keyboard
You can adjust parameters on the Decay Color &
EQ graph displays with a mouse by dragging the
appropriate dot on the graph.
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:
Cutting or boosting an EQ frequency band
To adjust EQ frequency crossover:

Drag the control point right or left.
1
Click on the parameter text that you want to
edit.
2
Change the value by doing one of the following.
• To increase a value, press the Up Arrow on your
keyboard. To decrease a value, press the Down
Arrow on your keyboard.
• 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
Setting the EQ crossover frequency
To adjust high frequency rear cut:

Drag the control point right or left.
Do one of the following:
• Press Enter on the numeric keyboard to input the
value and remain in keyboard editing mode.
• 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.
Setting the rear cut frequency
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173
Enabling Switches
To enable a switch, click on the switch (the round
LED indicator next to each switch name). Switch
LEDs illuminate when enabled.
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%.
Early Reflection switch LED (on)
ReVibe Controls
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 and EQ graph displays.
ReVibe Master Mix Section
Controls
The Master Mix section has controls for adjusting
the relative levels of the source signal and the reverb effect.
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.
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.
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Audio Plug-Ins Guide
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.
100% Wet Mix Button
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.
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.
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%.
Rate Control
Level Control
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.
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.
Setting the Rate above 20 Hz can cause frequency
modulation to occur. This will add side-band 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
This button toggles the chorus effect on or off.
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 widely spaced
early reflections that may sound unnatural. At
–100%, 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.
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.
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Pre-Delay Link Button
Input Control
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 Pre-Delay
amount, so that the total delay for the early reflections is the sum of the Early Reflection Pre-Delay
and the Reverb Pre-Delay.
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
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.
Center Control
Pre-Delay Link button
This button toggles early reflections on or off.
When early reflections are off, the reverb effect
consists entirely of reverb tail.
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.
ReVibe Levels Section Controls
The range of this control is from –INF (minus infinity) to 0.0 dB.
ER On/Off Button
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.
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.
176
Center controls the output level of the center channel outputs of multichannel formats that have a
center channel (such as LCR or 5.1).
Audio Plug-Ins Guide
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 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.
Room Type Number
Room Type Category pop-up
The Rear ER control has no effect when the
early reflections are turned off with the ER
On/Off button.
Rear Level Link Button
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.
Room Type Name pop-up
Preset Next and
Previous buttons
Room Type display and controls
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.
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.
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 183 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.
Next and Previous Buttons
Click the Next or Previous buttons to choose the
next or previous Room Type.
For more information on saving and importing
plug-in presets, see the Pro Tools Reference
Guide.
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177
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.
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:
In addition to letting you adjust the overall sound
of the room, the high-frequency and low-frequency components are split to allow you to emphasize or de-emphasize the low and high frequency response of the room.
• Automatic selects the reverb tail type stored with
the room type.
Coloration Control
• Fast Attack can be useful for plate reverbs.
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.
• Natural is an average reverb tail type with no extreme characteristics.
• Smooth is optimized for large rooms.
• Dense is similar to smooth, and can also be good
for a plate reverb.
• 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.
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%.
LF Color Control
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
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.
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• Small is optimized for small rooms.
Level Control
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
Rear Shape Control
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.
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%.
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 pre-delay settings produce
effects that are unnatural but sonically interesting.
The range of this control is from 0.0 ms to
300.0 ms.
Diffusion Control
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.
Attack Time 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.
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%.
Size Control
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.
Spread Control
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.
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%.
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179
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.
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 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.
Low Frequency Ratio Control
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 173.)
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.
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.
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.
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Audio Plug-Ins Guide
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.
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.
Low Gain Control
Front reverb
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.
Rear reverb
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 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.
Early reflections
Contour display
ER Button
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.
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.
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.
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181
Front Button
internal clip indicator
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.
channel clip indicator
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.
ReVibe Input/Output Meter
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.
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).
ReVibe Help Button
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
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Audio Plug-Ins Guide
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
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 4
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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183
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
184
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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185
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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Audio Plug-Ins Guide
Chapter 32: ReVibe II
ReVibe II is studio-quality reverb and acoustic environment modeling plug-in available in the AAX
plug-in format (DSP, Native, and AudioSuite).
Using ReVibe II
ReVibe II makes it possible to model extremely realistic acoustic spaces and place audio elements
within them in a Pro Tools mix.
ReVibe and ReVibe II provide essentially
the same controls with the same parameter
values.
ReVibe II supports 44.1 kHz, 48 kHz, 88.2 kHz,
96 kHz, 176.4 kHz, and 192 kHz sessions.
ReVibe II works with mono and stereo formats,
and LCR, LCRS, quad, 5.0, and 5.1 greater-thanstereo multichannel formats.
In general, when working with stereo and greaterthan-stereo tracks, use the multichannel version of
ReVibe II.
ReVibe II plug-in (AAX)
Chapter 32: ReVibe II
187
Revibe II supports the following combinations of track types and plug-in insert formats:
Track
Type
Plug–in Insert Format
Mono
Stereo
LCR
LCRS
Quad
5.0
5.1
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Mono
Stereo
LCR
•
LCRS
•
Quad
•
5.0
•
5.1
Adjusting ReVibe II
Parameters
To cut or boost a particular EQ band:

Drag a control point up or down.
You can adjust ReVibe II 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.
Cutting or boosting an EQ frequency band
To adjust EQ frequency crossover:

Drag the control point 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.
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Audio Plug-Ins Guide
Setting the EQ crossover frequency
To adjust high frequency rear cut:

Drag the control point right or left.
3
Do one of the following:
• Press Enter on the numeric keyboard to input the
value and remain in keyboard editing mode.
• 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.
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:
1
Click on the parameter text that you want to
edit.
2
Change the value by doing one of the following.
• To increase a value, press the Up Arrow on your
keyboard. To decrease a value, press the Down
Arrow on your keyboard.
• 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.
Chapter 32: ReVibe II
189
ReVibe II Input and Output
Meters
The Input and Output meters indicates the input
and output signal levels. These meters range from
0 dB to –96 dB. The number of input and output
meters that operate simultaneously ranges from a
single meter for mono input and output, up to five
input and output meters for 5.0 and 5.1 multichannel processing. The number of meters displayed
depends on the channel format of the track on
which the plug-in is inserted.
ReVibe II Controls
ReVibe II 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 and EQ graph displays.
Room and Reverb Type
ReVibe II lets you select the type of Room and Reverb modeled. Each Room and Reverb type models early reflection characteristics for specific
types of rooms or effects devices. Each Room and
Reverb type also incorporates a complex room coloration EQ, which models the general frequency
response of various rooms and effects devices.
Choosing a new Reverb Type changes the early reflections and room coloration EQ only. All of the
other ReVibe II 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.
Input and Output meters (5.1)
Room Type Category menu
Clip Indicators
A red channel clip indicator appears at the top of
each meter. The clip indicator lights when the signal level exceeds 0 dB, and stays lit until cleared.
Clicking a meter’s clip indicator clears that meter.
Room Type Name menu
Preset Next and
Previous buttons
Reverb Type menu
Reverb Type display and controls
The Reverb Type display shows the Room Type
Category, Room Type Name, the Next and Previous buttons, and the Reverb Type.
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Audio Plug-Ins Guide
Room Type Category Menu
Clicking on the Room Type Category 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 menu to select from a
list of all available Room Type presets.
See “ReVibe II Room Types” on page 198 for
a list of room presets.
Next and Previous Buttons
Click the Next or Previous buttons to choose the
next or previous Room Type.
Reverb Type Menu
Click the Reverb Type menu to select the type of
reverb tail. There are nine basic reverb types, plus
Automatic. Select Automatic to use the reverb tail
type 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.
• Smooth is optimized for large rooms.
ReVibe II Reverb Section
Controls
The Reverb section has controls for the various reverb tail elements, including level, time, size,
spread, attack time, attack shape, rear shape, diffusion, and pre-delay. These determine the overall
character of the reverb tail.
Size Control
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.
Time Control
• Tight is good for small to medium rooms.
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.
• Sparse 1 produces sparse early reflections with a
high diffusion buildup.
Level Control
• Sparse 2 can be useful for a spring reverb.
Level controls the output level of the reverb tail.
• Fast Attack can be useful for plate reverbs.
• Dense is similar to smooth, and can also be good
for a plate reverb.
• Wide is a generic large reverb.
• Small is optimized for small rooms.
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.
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191
Diffusion Control
Attack Time Control
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.
Attack Time adjusts the length of time between the
Spread Control
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
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%.
Rear Shape adjusts the envelope of the reverb in
Pre-Delay Control
ReVibe II Early Reflection
Section
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.
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 pre-delay settings produce
effects that are unnatural but sonically interesting.
The range of this control is from 0.0 ms to
300.0 ms.
192
start of the reverb tail and its peak level. Settings
are Short, Medium, or Long.
Audio Plug-Ins Guide
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%.
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 II 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.
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.
Level Control
Level controls the output level of the early reflec-
tions. 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.
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 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.
Pre-Delay Link Button
The Early Reflections 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 Pre-Delay amount, so that the total delay for the early reflections is the sum of the Early
Reflection Pre-Delay and the Reverb Pre-Delay.
Early Reflections On Button
This button toggles early reflections on or off.
When early reflections are off, the reverb effect
consists entirely of reverb tail.
ReVibe II 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.
In addition to letting you adjust the overall sound
of the room, the high-frequency and low-frequency components are split to allow you to emphasize or de-emphasize the low and high frequency response of the room.
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.
Chapter 32: ReVibe II
193
High Frequency Color Control
Front Control
High Frequency Color (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%.
Front controls the output level of the front left and
Low Frequency Color Control
Center Control
Low Frequency 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%.
Center controls the output level of the center chan-
ReVibe II Levels Section
Controls
The Levels section has controls for adjusting
source input and ReVibe II output levels. ReVibe
II provides individual output level controls for
Front, Center, Rear reverb, and Rear early reflections.
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 pre-
vent 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.
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.
nel outputs of multichannel formats that have a
center channel (such as LCR or 5.1).
When ReVibe II 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 controls the output level of the rear outputs of
multichannel formats that have rear channels (such
as quad or 5.1).
When ReVibe II 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 Early Reflections Control
Rear ER controls the output level of early reflec-
tions 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.
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Audio Plug-Ins Guide
Rear Level Link Button
The Rear Level Link button toggles linking of the
Rear Reverb and Rear Early Reflections controls
on or off. The Rear Reverb and the Rear Early Reflections controls are linked by default. When
linked, the Rear Early Reflections and Rear Reverb controls move in tandem when either is adjusted. When unlinked, the Rear Early Reflections
and the Rear Reverb controls can be adjusted independently.
ReVibe II 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.
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%.
ReVibe II Mix Section Controls
The Mix section has controls for adjusting the relative levels of the source signal and the reverb effect.
Wet Control
The Wet control adjusts the mix between the dry,
unprocessed signal and the reverb effect. If you insert the ReVibe II 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.
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.
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.
Rate Control
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.
Setting the Rate above 20 Hz can cause frequency
modulation to occur. This will add side-band harmonics and change the reverb’s tone color, producing interesting effects. Typical settings are between 0.2 Hz and 1.0 Hz.
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.
Chorus On Button
This button toggles the chorus effect on or off.
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195
100% Wet and Dry Mix Buttons
Low Gain Control
These buttons set the Wet control to 100% Wet or
100% Dry 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.
The Lo Gain control 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.
ReVibe II Decay EQ Graph
The EQ display lets you graphically edit the Decay
EQ parameters for Revibe II. Click the EQ button
to toggle the display to show the Decay EQ parameters. To edit a parameter on the graph, drag the
corresponding control point.
High Frequency Control
The Hi Freq control 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 Gain Control
The Hi Gain control 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 Frequency Rear Cut Control
EQ display
Each control point (dot) on the graph has corresponding parameter text fields above and below
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 189.)
Low Frequency Control
The Lo Freq control 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.
The Rear control 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.
ReVibe II Decay Color Graph
The Color display lets you graphically edit the Decay Color parameters for Revibe II. Click the
Color button to toggle the display to show Decay
Color parameters. To edit a parameter on the
graph, drag the corresponding control point.
Color display
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Audio Plug-Ins Guide
You can use the controls in the Decay Color graph
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.
ReVibe II Contour Display
The Contour display shows the current reverb
shape and early reflections graphically. 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.
Low Frequency Crossover Control
The Lo Crossover control 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.
Low Frequency Ratio Control
The Lo Ratio control 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.
High Frequency Crossover Control
The Hi Crossover control 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.
High Frequency Ratio Control
The Hi Ratio control 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.
Contour display
Early Reflections Button
The Early Reflections button toggles display of
early reflections on or off within the Contour display. When the Early Reflections button is illuminated, early reflections data is displayed. When the
Early Reflections button is not illuminated, early
reflections data is not displayed. Both early reflections and reverb contour data can be displayed simultaneously.
Reverb Contour Button
The 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 Reverb Contour button is illuminated, the reverberation envelopes are displayed. When the Reverb
Contour button is not illuminated, the reverberation envelopes are not displayed. Both early reflections and reverb contour data can be displayed simultaneously.
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197
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.
Rear Button
ReVibe II Room Types
ReVibe II 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:
Studios
Large Natural Studio 1
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.
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
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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Audio Plug-Ins Guide
Rooms
Halls
Large Bright Room 1
Large Natural Hall 1
Large Bright Room 2
Large Natural Hall 2
Large Neutral Room 1
Large Natural Hall 3
Large Neutral Room 2
Large Natural Hall 4
Large Dark Room 1
Large Natural Hall 5
Large Dark Room 2
Large Natural Hall 6
Large Boomy Room
Large Dense Hall
Medium Bright Room 1
Large Sparse Hall
Medium Bright Room 2
Medium Natural Hall 1
Medium Bright Room 3
Medium Natural Hall 2
Medium Neutral Room 1
Medium Natural Hall 3
Medium Neutral Room 2
Medium Natural Hall 4
Medium Neutral Room 3
Medium Dense Hall
Medium Dark Room 1
Small Natural Hall 1
Medium Dark Room 2
Small Natural Hall 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
Theaters
Large Theater 1
Large Theater 2
Medium Theater 1
Medium Theater 2
Small Theater 1
Small Theater 2
Churches
Large Natural Church 1
Large Natural Church 2
Large Dense Church
Large Slap Church
Medium Natural Church 1
Medium Natural Church 2
Medium Dense Church
Small Natural Church 1
Small Natural Church 2
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199
Cathedrals
Large Chamber 1
Natural Cathedral 2
Large Chamber 2
Natural Cathedral 3
Large Chamber 3
Dense Cathedral 1
Large Chamber 4
Dense Cathedral 2
Large Chamber 5
Slap Cathedral
Large Chamber 6
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
Studio Spring 3
Studio Spring 4
Dense Spring 1
Dense Spring 2
Resonant Spring
Funky Spring 1
Funky Spring 2
Funky Spring 3
Funky Spring 4
200
Chambers
Natural Cathedral 1
Audio Plug-Ins Guide
Medium Chamber 1
Medium Chamber 2
Medium Chamber 3
Medium Chamber 4
Medium Chamber 5
Small Chamber 1
Small Chamber 2
Small Chamber 3
Small Chamber 4
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
Very Small Ambience
Film and Post
Large Spaces
Medium Kitchen
Parking Garage 1
Small Kitchen
Parking Garage 2
Bathroom 1
Parking Garage 3
Bathroom 2
Warehouse 1
Bathroom 3
Warehouse 2
Bathroom 4
Stairwell 1
Bathroom 5
Stairwell 2
Shower Stall
Stairwell 3
Hallway
Stairwell 4
Closet
Stairwell 5
Classroom 1
Gymnasium
Classroom 2
Auditorium
Large Concrete Room
Indoor Arena
Medium Concrete Room
Stadium 1
Locker Room
Stadium 2
Muffled Room
Tunnel
Very Small Room 1
Very Small Room 2
Very Small Room 3
Car 1
Car 2
Car 3
Car 4
Vintage Digital
Large Hall Digital
Medium Hall Digital
Large Room Digital
Medium Room Digital
Small Room Digital
Car 5
Phone Booth
Metal Garbage Can
Drain Pipe
Tin Can
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201
Effects
Mono Slapback 1
Mono Slapback 2
Mono Slapback 3
Wide Slapback 1
Wide Slapback 2
Wide Slapback 3
Multi Slapback 1
Multi Slapback 2
Multi Slapback 3
Multi Slapback 4
Spread Slapback 1
Spread Slapback 2
Mono Echo 1
Mono Echo 2
Mono Echo 3
Wide Echo 1
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 33: 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 33: TL Space TDM and TL Space Native
203
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.
Automation and Ease of Use Features
Reverb Features
• Snapshot mode supports rapid changes between
ten predefined reverb scenes
• Mono, Stereo, and Quad and 5.0–channel output
support
• Picture preview mode allows user to view image
files stored with impulse responses
• Multiband EQ
• Impulse responses stored directly in Pro Tools
presets and sessions for easy session sharing
• 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
• 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
• Precise control of low, mid, and high decay
crossover
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
Interface Features
• Full waveform view, zoom, and channel highlight functions
• Onscreen input and output metering with clip indicators
• Independent front and rear decay
• Snapshot mode ideal for post automation requirements
• Seamless snapshot switching (RTAS)
• Automatic phantom channel creation
• Impulse response information display
IR Library
Impulse Response (IR) Loading and
Organization Features
• A wide variety of both real and synthetic reverb
spaces and effects
• Scrollable IR browser makes finding impulse responses easy
• Browser supports user defined IR groups on any
local drives
• Browser keyboard shortcuts
• IR favorites function
204
• Quick browser buttons allow rapid IR loading
and preview
Audio Plug-Ins Guide
• 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 consists 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 signal to the console. An alternative
and less expensive analog reverb system is the
spring reverb, most commonly seen in guitar am-
plifiers 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.
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.
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205
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.
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.
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Audio Plug-Ins Guide
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 33: TL Space TDM and TL Space Native
207
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 plugin 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
—
Chapter 33: TL Space TDM and TL Space Native
209
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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Audio Plug-Ins Guide
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 221.
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.
2
Double-click the TL Space IR library installer
application to launch it. Read the license agreement. If you agree to the terms, click Accept.
3
Click Install to perform an easy install of the entire IR library on the system drive.
4
If you want to install only part of the library, select Custom Install and select the parts of the library you want to install.
5
When the installation is completed, click Quit to
finish the installation.
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.
Chapter 33: TL Space TDM and TL Space Native
211
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 multimono 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.
2
Locate and select the library on your hard drive.
3
Click Choose.
TL Space will add the new library to the IR
browser.
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Audio Plug-Ins Guide
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
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.
• 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.
• 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.
• 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.
The following examples show how various multimono 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
Cathedral.1.C.wav
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
Cathedral.2.Rs.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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213
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.
214
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.
Chapter 33: TL Space TDM and TL Space Native
215
TL Space Controls and Displays
The TL Space interface is divided into the following sections:
• Display area (See “TL Space Display Area” on page 217.)
• IR Browser (See “TL Space IR Browser” on page 220.)
• Primary controls (See “TL Space Primary Controls” on page 222.)
• Group Selectors and Controls (See “TL Space Group Selectors and Controls” on page 223.)
The TL Space interface
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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
the IR browser. If no such 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.
• Preferences mode
Display Mode selectors
The Display area changes based on the selected
mode.
Info Bar
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.
At all times, the Info bar at the bottom of the display area window shows the following controls
and information.
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
The controls in Waveform mode function as follows:
currently loaded IR.
Quick Browser Controls The Quick browser con-
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
Original Bypasses all waveform processing, al-
lowing the original IR to be auditioned. This control effectively bypasses the processing in the IR
computer as shown in the system diagram.
Chapter 33: TL Space TDM and TL Space Native
217
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.
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.
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.
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.
Name Displays the name of each snapshot. By de-
fault, 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).
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 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.
218
Audio Plug-Ins Guide
Sample Path Displays the name of the IR selected
for each snapshot.
Copy Copies the currently selected snapshot set-
tings 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
TL Space Meters
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.
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 re-
duces PCI contention for Pro Tools systems when
using PCI video capture hardware. For more information, see “PCI Bus Contention and TL Space”
on page 227.
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.
Installed IR Packages Displays a list of installed
TL Space IR packages and their versions.
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.
Chapter 33: TL Space TDM and TL Space Native
219
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.
IR Browser 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 214).
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.
To add an IR file or folder to the Favorites folder:
1
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.
220
Audio Plug-Ins Guide
2
In the IR browser, select the desired IR file or
folder.
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 221).
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.
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/
Add to Favorites Adds the currently selected IR to
These package files are supplied in a lossless compressed format.
the Favorites group at the top of the browser window.
To install a TL Space IR package:
New Folder in Favorites Creates a folder in the
1
In the TL Space IR browser, select Download IR
Package from the Edit menu. Your default Web
browser launches and loads the Avid TL Space
Online IR Library website (www.avid.com/tlspace/impulselibrary/).
2
Click Download
3
Login using your email address and password.
You may need to create a new account if you
have not yet registered TL Space.
Favorites group. Favorite IRs can be dragged and
dropped into the folder.
Rename Favorites Folder Lets you rename the
currently selected Favorites folder.
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 33: TL Space TDM and TL Space Native
221
7
In the resulting dialog, locate and select the file
you downloaded.
TL Space Primary Controls
8
Click Choose.
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.
TL Space will display a summary of the IR package with a short description, copyright statement,
and a list of the contents.
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.
TL Space primary controls
The details of all installed IR packages can be reviewed using the Show Packages option in Preferences mode.
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 reverb
signal, from –inf dB to +12 dB.
Decay Controls the overall decay of the IR wave-
form and is displayed as a percentage of the original. When Decay is adjusted, the waveform is recalculated in real time.
222
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.
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 reflec-
tions 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 output
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.
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
–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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223
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 reflections
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.
224
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
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.
Lo Freq Adjusts the frequency of a low frequency
filter from 20 to 500 Hz.
Size Changes the size of the Early reflections,
Lo Gain Cuts or boosts the frequency set in Lo
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.
Freq from –15 dB to +15 dB.
Audio Plug-Ins Guide
The Reverb group offers a low and high shelf EQ
in addition to width and balance controls. The EQ
operates prior to convolution processing.
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 spaciousness
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.
Balance Controls the balance of the reverb output.
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.
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 divides the IR into low and mid frequency portions.
Mid Decreases or increases the rate at which mid
frequencies decay.
High Xover Adjusts the frequency point that divides the IR into mid and high frequency portions.
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.
Chapter 33: TL Space TDM and TL Space Native
225
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.
3
Load an IR into each Snapshot and make any
desired changes to specific TL Space controls.
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
Select TL Space > Snapshot from the automation menu for the track.
8
Select the Pencil tool.
Automating TL Space Snapshots
9
Draw the desired automation. The names displayed in the automation track will match the
names entered for each Snapshot.
Snapshot automation is a powerful method of
changing the reverb parameters without having to
individually automate each parameter.
226
To automate TL Space Snapshots:
Audio Plug-Ins Guide
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.
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.
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.
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.
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.
Chapter 33: TL Space TDM and TL Space Native
227
TL Space IR Library
TL Space includes an extensive impulse response library, divided into the following categories.
228
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 34: 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 34: AIR Dynamic Delay
231
Select from the following rhythmic values:
Mix
• 16 (sixteenth note)
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.
• 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)
The delay section of the Dynamic Delay plug-in
provides L/R Ratio and Stereo Width controls.
• 4D (dotted quarter note)
L/R Ratio
• 2 (half note)
• 6/4 (dotted whole note)
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.
• 7/4 (seven tied quarter notes)
Stereo Width
• 1T (whole-note triplet)
• 3/4 (dotted half note)
• 4/4 (whole note)
• 5/4 (five tied quarter notes)
• 8/4 (double 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.
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Dynamic Delay Delay Section
Audio Plug-Ins Guide
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.
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.
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.
Fbk
Adjust the Feedback control to determine how
much the Envelope follower affects the Feedback
amount.
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 34: AIR Dynamic Delay
233
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Audio Plug-Ins Guide
Chapter 35: 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).
When Sync is disabled, the Time control lets you
the main delay time in milliseconds and seconds.
Multi-Delay plug-in window
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!
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.
Chapter 35: AIR Multi-Delay
235
From
The From control sets the tap from which signal
will be cross-routed.
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.
High Cut
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
Multi-Delay Delay Tap 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:
On
The On button turns the selected tap’s signal on or
off.
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.
Pan
Adjust the Pan control to pan the audio signal from
the Tap left or right in the stereo field.
Chapter 36: 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 36: Mod Delay II
237
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 Delay Control
This control sets the delay time between the original signal and the delayed signal.
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.
To enable Tempo Sync:

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 36: Mod Delay II
239
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 37: Mod Delay III
Mod Delay III provides mono, multi-mono, monoto-stereo, and stereo modulating delay effects.
Mod Delay III is available in the AAX (DSP, Native, and AudioSuite) plug-in format, and supports
sample rates up to 192 kHz.
Input
Input Meters
The Input meters show peak signal levels before
processing:
Dark Blue Indicates nominal levels from –INF to
Mod Delay III Controls
–12 dB.
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.
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 37: Mod Delay III
241
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.
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
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.
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.
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Pro Tools Reference Guide
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.
Duration buttons
Groove
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.
Modulation Section
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:
Dark Blue Indicates nominal levels from –INF to
Rate
–12 dB.
The Rate control sets the rate of modulation of the
delayed signal (from 0.00 Hz to 20.0 Hz).
Light Blue Indicates pre-clipping levels, from
Depth
Red Indicates full scale levels (clipping)
The Depth control sets the depth of the modulation
applied to the delayed signal (from 0% to 100%).
Output Gain
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.
–12 dB to 0 dB.
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 monoto-stereo instances of Mod Delay III, there are independent Gain controls for each channel (left and
right).
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 37: Mod Delay III
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Chapter 38: Moogerfooger Analog Delay
The Moogerfooger Analog Delay is a delay
plug-in that is available in AAX, TDM, RTAS, and
AudioSuite formats. It provides a warm sounding
delay in the digital domain.
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.
So Why Analog?
Moogerfooger Analog Delay
How the Moogerfooger Analog Delay Works
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.
Avid’s digital replica re-creates all the warm, natural sounds of its analog counterpart.
Chapter 38: Moogerfooger Analog Delay
245
Not Better—Different
The Moogerfooger Analog Delay plug-in was enhanced 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:
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 signal
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.
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Audio Plug-Ins Guide
HPF On/Off The HPF Off/HPF On enables or dis-
ables the highpass filter (HPF).
Drive The Drive control sets the input gain.
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 (by-
passed) or green (not bypassed) to show whether or
not the effect is in the signal path.
Moogerfooger Analog Delay Tips and Tricks
Infidelity
Because analog delay chips offer only a fixed number of cells, the extended delay times store a lowerfidelity version of the input signal. Try the Long
delay setting when going for cool “lo-fi” 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 39: 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.
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.
Pan Provides individual control over the apparent
location of each of the four taps in the stereo field.
Delay Sets the delay time between the original sig-
Multi-Tap Delay plug-in
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.
nal 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%.
Chapter 39: Multi-Tap Delay
247
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 plugin will then have space at the end of the clip in
which to write the final delay.
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Chapter 40: 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 ping-pong echo effect.
Delay Sets the delay time between the original sig-
nal 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 apPing-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 delayed 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 pingpongs 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 40: Ping-Pong Delay
249
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 plugin will then have space at the end of the clip in
which to write the final delay.
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Chapter 41: Reel Tape Delay
Reel Tape Delay is part of the Reel Tape suite of
tape-simulation effects plug-ins that are available
in AAX, 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.
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.
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
Reel Tape Delay
• 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.
Chapter 41: Reel Tape Delay
251
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 plugin 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 plug-in,
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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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 254.
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 plug-in
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
AAX, 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 plug-in
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
AAX, 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 41: Reel Tape Delay
253
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:
1
In the BPM Sync section, click the On button.
The Tempo/Rate display changes to match the
current session tempo.
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
3
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:
• 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 350series tape transport.
Rockabilly Plus Includes the Rockabilly setting
plus Feedback, Wow/Flutter, Bass and Treble adjustments on feedback.
Chapter 42: Tel-Ray Variable Delay
Tel-Ray Variable Delay is a delay/echo plug-in
that is available in AAX, 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 42: Tel-Ray Variable Delay
255
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 (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 (unprocessed)
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 con-
trol.
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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Audio Plug-Ins Guide
Since the original units are basically thirty year-old
tuna cans bolted to plywood with springs and motors flopping around inside, the Variation knob
was added so you can dial in a Tel-Ray in whatever
state of disrepair you desire.
Chapter 43: TimeAdjuster
TimeAdjuster is a time-processing plug-in that is
available in AAX (DSP and Native), TDM, and
RTAS formats.
With TDM and RTAS formats, there are three versions of the TimeAdjuster plug-in, each of which
supports different sample delay ranges:
The TimeAdjuster plug-in is an efficient way to
compensate for DSP or host-based processing delays in your Pro Tools system.
Short Supports a maximum delay of 256 samples
at all sample rates.
Medium Supports a maximum delay of 2048 samples at all sample rates.
Long Supports a maximum delay of 8192 samples
at all sample rates.
With the AAX format, there is a single version of
the TimeAdjuster plug-in that supports the entire
range of sample delays.
TimeAdjuster plug-in (AAX), stereo
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 (TDM and RTAS), mono
Use the TimeAdjuster plug-in for any of the following:
• Delay compensation
• Gain compensation (+/– 24 dB)
Phase Invert This controls inverts the phase (polarity) of the input signal. While most Avid plugins 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 258).
• Phase inversion for correcting out-of-phase signals
Chapter 43: TimeAdjuster
257
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.
Delay Provides up to 8192 samples of delay com-
pensation adjustment, or general adjustment of
phase relationships of audio recorded with multiple microphones (for TDM and RTAS the amount
of delay available depends on the version of TimeAdjuster: Short, Medium, or Long). 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.
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Audio Plug-Ins Guide
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 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.
To manually compensate for DSP-induced delays,
try one of the following methods:
• Phase inversion
• 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.
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:
1
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).
2
Apply the plug-in whose delay you want to calculate to the first track, and a Time Adjuster
plug-in to the second track.
3
With TimeAdjuster, invert the phase.
4
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.
5
Change the polarity back to normal.
6
Save the TimeAdjuster setting for later use.
Determining the DSP delay of track inserts (Mix
window shown)
2
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.
3
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.
4
Test the delay values by duplicating an audio
track and reversing its phase while compensating for delay.
Comb-Filter Effect Cancellation
Adjust the delay with the signal in phase until any
comb-filter effects cancel out.
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.
Chapter 43: TimeAdjuster
259
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, see the Pro Tools
Reference Guide.
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Audio Plug-Ins Guide
Part VII: Modulation Plug-Ins
Chapter 44: 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 millisec-
onds.
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.
Chorus plug-in window
AIR Chorus Controls
The Chorus plug-in provides a variety of controls
for adjusting plug-in parameters.
L/R Phase Sets the relative phase of the LFO’s
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 44: AIR Chorus
263
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Audio Plug-Ins Guide
Chapter 45: AIR Ensemble
AIR Ensemble is an RTAS plug-in that lets you apply fluid, shimmering modulation effects to the audio signal.
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 Delay
time.
Shimmer The Shimmer control lets you randomize
the Delay time, adding texture to the effect.
Chapter 45: AIR Ensemble
265
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Audio Plug-Ins Guide
Chapter 46: 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.
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 Gate Section
Filter Gate plug-in window
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
Filter Gate Controls
The Attack control lets you adjust the duration of
the attack as a percentage of the step duration.
The Filter Gate plug-in provides a variety of controls for adjusting plug-in parameters.
Hold
Pattern
The hold control lets you adjust the duration of the
hold (or sustain) as a percentage of the step duration.
The Pattern control let you select from a number of
preset rhythmic patterns that the gate will follow.
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.
Release
The Release control lets you adjust the duration of
the release as a percentage of the step duration.
Chapter 46: AIR Filter Gate
267
Filter Gate Filter Section
Filter Gate Modulation Section
The Filter controls provide controls for the selected filter type.
Env
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 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
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 47: 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
AIR Flanger Controls
• 4 (quarter note)
• 2T (half-note triplet)
• 4D (dotted quarter note)
The Flanger plug-in provides a variety of controls
for adjusting plug-in parameters.
• 2 (half note)
Sync
• 3/4 (dotted half 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.
• 4/4 (whole note)
• 1T (whole-note triplet)
• 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 47: AIR Flanger
269
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 48: 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.
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271
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 49: 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.
Rate
The Rate control sets the rate for the oscillation of
the LFO in Hertz.
Multi-Chorus Chorus Section
Controls
The Chorus section provides control over the lowfrequency content and stereo width of the MultiChorus effect.
Low Cut
The Low Cut control lets you adjust the Low Cut
frequency for the Chorus, to limit the Chorus
effects to higher frequencies.
Width
Depth
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 49: AIR Multi-Chorus
273
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 50: 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
Phaser Controls
• 2T (half-note triplet)
• 4D (dotted quarter note)
• 2 (half note)
The Phaser plug-in provides a variety of controls
for adjusting plug-in parameters.
• 1T (whole-note triplet)
Sync
• 4/4 (whole note)
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.
• 3/4 (dotted half 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
the rate of the Phaser in independently of the
Pro Tools session tempo.
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275
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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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 51: AIR Talkbox
AIR Talkbox is an RTAS plug-in that lets you add
a voice-like resonances to audio signals.
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).
Talkbox Plug-In window
Formant
Talkbox Controls
The Talkbox plug-in provides a variety of controls
for adjusting plug-in parameters.
The Formant control lets you shift the formant center of the processed audio up or down 12 semitones, changing the harmonic structure dramatically.
Vowel
Mix
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).
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 51: AIR Talkbox
277
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
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.
Random Provides random modulation.
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)
• 4 (quarter note)
• 2T (half-note triplet)
• 4D (dotted quarter note)
• 2 (half note)
• 1T (whole-note triplet)
• 3/4 (dotted half note)
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.
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)
Thresh
• 5/4 (five tied quarter notes)
Adjust the Thresh control to set the amplitude
threshold at which the Formant setting begins to be
modulated by the Envelope follower.
• 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 (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 52: 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 wavelike overtone at the Cutoff frequency.
Fat
Filter plug-in window
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 52: AIR Vintage Filter
279
Mode
Select one of the following options for the type of
filter:
LP24 Provides a low pass filter with a 24 dB
cutoff.
LP18 Provides a low pass filter with a 18 dB
cutoff.
LP12 Provides a low pass filter with a 12 dB
cutoff.
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 Filter 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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 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 Filter 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 53: 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 53: Cosmonaut Voice
281
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:
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:
1
Click the Plug-In Automation button in the
Plug-In window to open the Plug-In Automation window.
1
Focus the Cosmonaut Voice plug-in on your
control surface. All available parameters are
mapped to encoders, faders, and switches.
2
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
Adjust the control currently targeting the desired parameter.
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 B/S
Level from the Cosmonaut Voice sub-menu.
• Reveal an Automation lane for the track, click
the Automation Type selector and select B/S
Level from the Cosmonaut Voice sub-menu.
5
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Accessing Cosmonaut Voice
Controls from a Control Surface
Edit the breakpoint automation for the enabled
control.
Audio Plug-Ins Guide
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 54: 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 54: Chorus
283
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 sig-
nal 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 fre-
quency 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 ap-
plied 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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Chapter 55: 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 55: Flanger
285
LFO Rate Adjusts the rate of the LFO (low fre-
quency 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 ap-
plied 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 56: Moogerfooger Lowpass Filter
The Moogerfooger Lowpass Filter features a
2-pole/4-pole variable resonance filter with envelope follower and is available in AAX, 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.
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! The digital version preserves all the character, nuances, and personality of his original classic analog design.
Moogerfooger Low Pass Filter
Chapter 56: Moogerfooger Lowpass Filter
287
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.
“Envelope”
of the sound
Time
Audio waveform
of the sound
Audio waveform of a musical sound
Smooth/Fast The Smooth/Fast switch determines
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.
Envelope
Gain
Time
Envelope signal of the same sound
2-Pole
Moogerfooger Lowpass Filter
Controls
Envelope Section
4-Pole
Frequency
The 2/4 pole switch selects the filter slope
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
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 (2pole) 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 56: Moogerfooger Lowpass Filter
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Chapter 57: Moogerfooger 12-Stage Phaser
The Moogerfooger 12-Stage Phaser combines a 6or 12-stage phaser with a wide-ranging variable
LFO and is available in AAX, 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
5-Stage
Phaser
Moogerfooger 12-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 57: Moogerfooger 12-Stage Phaser
291
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 modu-
lation, 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 oscillates.
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.
Phaser Section
Control the Phaser with the Sweep and Resonance
knobs and the 6-Stage/12-Stage switch.
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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Frequency
Sweep adjusts the center frequency point
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.
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 phaseswept 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 58: Moogerfooger Ring Modulator
The Moogerfooger Ring Modulator that provides a
wide-range carrier oscillator and dual sine/square
waveform LFO and is available in AAX, 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
Control the LFO using the Amount and Rate knobs
and the Square/Sine waveform selector switch.
Amount Amount determines the amount of LFO
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.
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 oscillates,
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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295
Sine/Square The Square/Sine switch selects either
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 fre-
quency, 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 59: Reel Tape Flanger
Reel Tape Flanger is part of the Reel Tape suite of
tape-simulation effects plug-ins and is available in
AAX, 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.
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.
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
Reel Tape Flanger
• Bias: 0.0 dB
How Reel Tape Flanger Works
• Cal Adjust: +9 dB
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.
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.
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297
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 plugin 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 plug-in,
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)
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.
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
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 300.
Depth
zero point
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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Mix
Settings for this parameter are saved with
plug-in presets. If you use a preset for the
AAX, TDM, RTAS or AudioSuite version of
this plug-in, any settings for this parameter
will be active.
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).
Synchronizing Reel Tape
Flanger to Session Tempo
Invert
(Plug-In Automation Playlist or Control
Surface Access Only)
You can set the LFO Rate in Reel Tape Flanger to
synchronize to the session tempo (in beats per minute).
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.
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.
This parameter is accessible only from the plug-in
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
AAX, 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 plug-in
automation playlist or from a supported control
surface.
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Tempo/Rate
display
On
button
Note Value
display
Dot
button
Triplet
button
BPM Sync controls
2
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)
3
To adjust the rhythm further, do any of the following:
• 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.
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 works
well with acoustic guitar sounds
 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).
 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.

Flutter Flange Moderate-depth flange setting with
Wow/Flutter
Flutter Extreme Wow/Flutter setting with flanging
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 60: Sci-Fi
Sci-Fi part of the D-Fi family of plug-ins, providing analog synthesizer-type effects. It is available
in AAX (DSP, Native, and AudioSuite), 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-and-hold
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 (AAX)
Sci-Fi can be used as either a real-time plug-in
(AAX, TDM, or RTAS) or as a non-real-time AudioSuite plug-in.
With Pro Tools|HD systems, the multichannel TDM version of the Sci-Fi plug-in is not
supported at 192 kHz. Use the multi-mono
TDM or Native version instead.
Sci-Fi (TDM, RTAS)
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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 modulates
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.
Freak Mod Is a frequency modulation processor
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.
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Resonator+ and Resonator– Add a resonant frequency 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 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.
To use keyboard note entry:
1
Start-click (Windows) or Control-click (Mac)
the Effect Frequency slider to display the popup keyboard.
2
Select the note on the keyboard that you want
for the Effect Frequency.
Sci-Fi Mod Type Controls
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.
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.
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 wahwah-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.
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.
Trigger+Hold Trigger and hold modulation is 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.
Sci-Fi Mod Amount and Mod
Rate Controls
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 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.
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.
Chapter 60: Sci-Fi
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Chapter 61: 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.
The following figure shows the different modules
of TL EveryPhase and how they interact with the
audio signal.
Chapter 61: TL EveryPhase
307
TL EveryPhase Controls
The TL EveryPhase interface is divided into the
following sections of controls:
• Meter (see “TL EveryPhase Meter Section” on
page 308)
• Phaser (see “TL EveryPhase Phaser Section” on
page 309)
• Modulation (see “Modulation Section” on
page 310)
TL EveryPhase signal flow, processing, and controls
• LFO (see “LFO Section” on page 310)
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.
• Tempo (see “Tempo Controls” on page 312)
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
• Envelope (see “Envelope Section” on page 312)
TL EveryPhase Meter Section
There are two meters available, and Output meter
and a Modulation meter.
Output and Modulation meters
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
Resonance
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 plug-in, two scanning
bars are shown.
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 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.
TL EveryPhase Phaser Section
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
The Input slider lets you cut or boost of the input
signal level from –24 dB to +12 dB.
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.
Stages
Output
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.
The Output slider lets you cut or boost of the output signal level from –24 dB to +12 dB.
Phaser section
Input
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309
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
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.
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 se-
lected, 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.
Envelope When the Envelope trigger is selected,
the LFO is triggered directly by the Envelope detector, which listens to the audio signal. If the SideChain 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.
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 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
Envelope section
Selecting Duration
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.
When the Envelope Detector is not in use, the
controls in this section have no effect on the
sound.
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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.
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.
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.
Attack
• Some presets utilize the Side-Chain Input. If
necessary, ensure that you have a side-chain 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
Creating a Single Phased Sound
with TL EveryPhase
The Release slider sets the release rate of the Envelope Detector.
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.
2
Select an appropriate LFO Waveform, such as
Ramp.
3
Set the Rate to an appropriate value.
4
Enable the LFO Single trigger so the LFO will
only cycle once.
5
Select the Auto button at the top of the
TL EveryPhase plug-in window.
6
Add LFO Manual Trigger to the automation list.
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7
Set the Automation mode for the track to Write
or Touch.
8
Play the session
9
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.
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.
6
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.
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.
Bass line and drum loop tracks
The bass line can be phased by the drum loop as
follows:
1
Instantiate TL EveryPhase on the bass line
track.
2
Send the drum loop track to a bus.
3
Set the Side-Chain Input on TL EveryPhase to
listen to the selected bus.
4
Activate the Side-Chain Input in TL EveryPhase by selecting the key icon in the Envelope
section.
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.
Resulting phased bass line
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.
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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 62: Voce Plug-Ins
The Voce plug-ins provide a pair of vintage modulation effect plug-ins that are available in AAX,
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.
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.”
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
Voce Chorus/Vibrato
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.
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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.
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!
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
How Voce Spin Works
Don Leslie invented the rotating speaker in 1937.
His design is simple and elegant: an internal 40watt 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
Direct
Sound
Lower speaker assembly
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Rotation
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.
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.
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.
Spin Presets
122 Model 122 speaker, medium pulleys.
122 (Small Pulley) Small pulleys (fastest
rotation).
122 (Large Pulley) Large pulleys (slowest
rotation).
122 (Wide Stereo) Middle pulleys, wide stereo
microphone placement.
122 (Mono) Middle pulleys, one mic each top and
bottom.
21H Model 21H speaker.
Foam Drum Middle pulleys, microphones close.
Memphis Lower drum slow motor unplugged, microphones close.
Steppenwolf Lower drum only, loose belts, micro-
phones close.
Rover (Slow to Fast) Guitar rotating speaker,
maximum speed differential.
Rover (Slow to Medium) Guitar rotating speaker,
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 320.
You may also Alt-click (Windows) or Optionclick (Mac) anywhere to toggle between Chorale
and Tremolo speeds.
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.
Off No rotation, but still through the crossover and
speakers (wherever the speakers comes to rest relative to the microphones!).
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Voce Spin Additional Controls
Accessing Voce Spin Controls
Though the Voce Spin plug-in window contains
only the Chorale/Off/Tremolo control, the following parameters are also available:
Accessing Voce Spin Controls On-Screen
• Input Trim
All Voce Spin parameters can be adjusted onscreen by editing Pro Tools breakpoint automation
data.
• Speed Switch
• Rotor Balance
• Upper Slow Speed
To access additional Voce Spin parameters onscreen:
1
Click the Plug-In Automation button in the
Plug-In window to open the Plug-In Automation window.
2
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.
3
Click OK to close the Plug-In Automation window.
4
In the Edit window, do one of the following:
• Upper Accel Rate
• Upper Decel Rate
• Upper Mic Angle
• Lower Fast Speed
• Lower Slow Speed
• Lower Accel Rate
• Lower Decel Rate
• Lower Mic Angle
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), 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.
• Click the Track View selector and select the automation control you just enabled from the Voce
Spin sub-menu.
• 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.
5
You can access these additional controls through
Pro Tools plug-in automation, and/or from a compatible control surface.
Edit the breakpoint automation for the enabled
control.
Accessing Voce Spin Controls on a 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.
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Audio Plug-Ins Guide
To access additional Voce controls from a control
surface:
1
Focus the Voce Spin plug-in on your control
surface. All available parameters are mapped to
encoders, faders, and switches.
2
Adjust the control currently targeting the desired parameter.
3
If necessary, use the previous/next Page controls to access additional controls.
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
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.
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.
Spin into Moogerfooger Lowpass Filter
Try using the amplitude modulation effects of Spin
as an LFO driving the Moogerfooger Lowpass Filter!
Generator Leakage
Of all the sounds to pass through a Leslie, no sound
has been amplified more often than the sound of B3 Organ generator leakage. Even with no notes
keyed, a small amount of B-3 sound leaks out.
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Part VIII: Harmonic Plug-Ins
Chapter 63: 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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325
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 clip-
ping 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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Audio Plug-Ins Guide
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 64: 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 fre-
quency 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 additional high frequency harmonics in the signal
(0.0–12.0 dB).
Phase Toggle the Phase control to change the po-
larity 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 65: 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 antialiasing 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 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.
The Sample Rate control resamples the audio signal at another sample rate.
On
Bit Depth
For a much grittier sound, disable the Anti-Alias
filter. The Anti-Alias button is lit when the filter is
enabled.
The Bit Depth control lets you truncate the bit
depth of the incoming signal from 16 bits all the
way down to 1 bit.
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329
AIR Lo Fi LFO Section Controls
The LFO controls let you apply a Low Frequency
Oscillator to modulate the Sample Rate.
Rate
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:
• 16 (sixteenth note)
Wave
Select from the following waveforms for the LFO.
Name
Description
Sine
Provides a sine wave
Tri
Provides a triangle wave
Saw
Provides a saw-tooth wave
Square
Provides a square wave
Morse
Provides a Morse code-like
rhythmic effect
S&H
Provides Sample and Hold (S&H)
modulation
Random
Provides random modulation
• 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)
Depth
The Depth control lets you adjust the amount of
modulation applied to the Sample Rate.
• 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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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 high-frequency
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 66: Lo-Fi
Lo-Fi is provides “retro,” down-processing
effects in AAX (DSP, Native, and AudioSuite),
TDM, RTAS, and AudioSuite formats.
Lo-Fi 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 (AAX)
Lo-Fi can be used as either a real-time plug-in
(AAX, TDM, or RTAS) or as a non-real-time AudioSuite plug-in.
With Pro Tools|HD systems, the multichannel
TDM version of the Lo-Fi plug-in is not
supported at 192 kHz, use the multi-mono
TDM or Native version instead.
Lo-Fi (TDM and RTAS)
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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.
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.
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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 sam-
ple data bits in an effort to fit the audio into the selected bit resolution. This imparts a characteristically raunchy sound to the audio that 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 67: Recti-Fi
Recti-Fi provides additive harmonic processing effects through waveform rectification. Recti-Fi is
available in AAX, 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,
Recti-Fi (AAX)
Recti-Fi (TDM and RTAS)
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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 waveform 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 PreFilter 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 PostFilter 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 68: Reel Tape Saturation
Reel Tape Saturation is part of the Reel Tape suite
of tape-simulation effects plug-ins that are available in AAX, 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
Output controls the output signal level of the plugin after processing. Output is adjustable from
–12 dB to +12 dB, with a default value of 0 dB.
Reel Tape Saturation
Tape Machine
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.
The Tape Machine control lets you select one of
three tape machine types emulated by the plug-in,
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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Tape Formula
Bias
The Tape Formula control lets you select either of
two magnetic tape formulations emulated by the
plug-in, each with its own saturation characteristics:
The Bias control simulates the effect of under- or
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.
Classic Emulates the characteristics of
Ampex 456, exhibiting a more pronounced saturation effect.
Hi Output Emulates the characteristics of
Cal Adjust
Quantegy GP9, exhibiting a more subtle saturation
effect.
Cal Adjust simulates the effect of three common
calibration levels on the modeled tape machine and
magnetic tape formulations.
Reel Tape Saturation
Controls
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.
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.
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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.
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 69: SansAmp PSA-1
SansAmp PSA-1 is a guitar amp simulator plug-in that is available in AAX, 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 loworder 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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345
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 mid-boost 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
Pre-Amp 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 70: 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 353.
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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 fullscale 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 358.
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Broadband Noise Reduction
Controls
BNR provides audio processing controls, controls
Contour Line controls, and controls for navigating
the Spectral Graph.
BNR Audio Processing Controls
Response
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.
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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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.
Smoothing
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.
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 High-Shelf
EQ can be enabled and disabled by clicking the
Enable button.
High-Shelf EQ
BNR Contour Line Controls
Learn
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.
Learn button
Learn First Audio Mode Learn First Audio mode
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.
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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Fit
Auto Fit
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.
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).
Fit button
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.
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.
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.
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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Move Breakpoints Up/Down/Left/Right buttons
BNR Spectral Graph Navigation
Controls
Scroll Left/Right
These buttons scroll the Spectral Graph to the left
or right, respectively.
breakpoints up or down, respectively. Alt-Start
key-clicking (Windows) or Control-Option-clicking (Mac) the Arrow keys on your computer keyboard performs the same function.
Undo
Clicking the Undo button undoes the last edit to the
Spectral Graph Display. The Undo button does not
undo changes made to slider positions.
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
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.
Undo button
Using Broadband Noise
Reduction
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.
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
Chapter 70: DINR
355
To use Broadband Noise Reduction:
1
From the Insert pop-up on the track with the
noise, select BNR. The Broadband Noise Reduction window appears.
2
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
Do one of the following:
• 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.
• 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.
356
4
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.
5
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.
6
Adjust the NR amount slider to reduce the noise
by the desired amount. To compare the
audio with and without noise reduction, click
Bypass.
Audio Plug-Ins Guide
7
To fine-tune the effects of the noise reduction,
adjust the Response, Release, and Smoothing
sliders to achieve optimal results.
8
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 358.
9
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.
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.
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 358.
To generate a Contour Line for audio that lacks a
noise-only portion:
1
In the Edit window, select a segment of audio
with a relatively low amount of signal and a
high amount of noise.
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.
5
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.
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, Controlclick (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.
8
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 358.
Noise components on the Spectral Graph
Chapter 70: DINR
357
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.
Dragging a breakpoint
2
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.
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:
1
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.
2
Select Options > Loop Playback.
3
Begin playback.
Moving selected breakpoints
3
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.
4
To create a new breakpoint, click on the Contour Line.
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.
To edit the Contour Line:
1
358
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.
Audio Plug-Ins Guide
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:
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.
Preparing to Render a Clip
To prepare a clip to process with the BNR
AudioSuite plug-in:
1
Select the desired clips in the target tracks or the
Clip List. Only tracks and clips that are selected
will be processed.
2
From the Pro Tools AudioSuite menu, choose
BNR.
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.
4
Click Fit or Super Fit to create a Contour Line
that matches the noise signature.
5
Click Preview to begin playback of the selected
material.
6
Adjust BNR controls and fine-tune the noise reduction using the techniques explained above
(See “Using Broadband Noise Reduction” on
page 355.)
7
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.
8
If unwanted artifacts are generated by the noise
reduction process, click Post-processing. For
best results, set the Response and Release controls to zero.
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.
BNR AudioSuite
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359
Processing a Clip
To begin processing:
1
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.
2
Render the selected clip by doing one of the
following:
• To render the selected clip only in the track in
which it appears, choose Playlist from the Selection Reference pop-up.
• To render the selected clip in the Audio Clip List
only, choose Clip List from the Selection Reference pop-up.
3
Determine which occurrences of the selected
clip you want to render by doing one of the
following:
• 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).
• If you do not want to update every occurrence of
the selected clip, disable Use In Playlist.
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.

BNR AudioSuite does not allow destructive
processing, so the Overwrite Files option is
not available in the File mode pop-up menu.
360
4
From the Destination Track pop-up, choose the
destination for the replacement audio.
5
Click Render.
Audio Plug-Ins Guide
Part X: Dither Plug-Ins
Chapter 71: Dither
Dither is a dither-generation plug-in that is
available in AAX, 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 Chapter 72, “POW-r
Dither.”
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 devices
with a maximum resolution of 16 bits, such as
DAT and CD recorders.
18-bit Recommended for output to digital devices
with a maximum resolution of 18 bits.
For more information on using Dither, see
the Pro Tools Reference Guide.
Chapter 71: Dither
363
20-bit Recommended for output to digital devices
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 72: POW-r Dither
POW-r Dither is a dither-generation plug-in that is
available in AAX, 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 71, “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 devices
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 POW-r
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 POWr Dither be used only as the last insert in the
signal chain (especially when using Type 1
Noise Shaping).
Chapter 72: POW-r Dither
365
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 fullspectrum, 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 73: 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
Stereo Width Trim Section Controls
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”).
The Trim controls adjust the perceived center/source of the generated stereo signal.
Level The Level control sets the volume of the per-
ceived center of the stereo signal.
Pan The Pan control sets the position left-to-right
of the perceived center of the stereo signal
Chapter 73: AIR Stereo Width
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370
Audio Plug-Ins Guide
Chapter 74: Down Mixer
Avid Down Mixer is an AAX plug-in (Native) that
can be used to automatically mix greater-than-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
Avid Down Mixer plug-in, Stereo to Mono shown
Down Mixer supports 44.1 kHz, 48 kHz,
88.2 kHz, 96 kHz, 176.4 kHz and 192 kHz sample
rates.
Down Mixer supports the following greater-thanstereo multichannel formats:
• LCR
Source
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.
• LCRS
Mute
• 5.1
When enabled, the Mute button mutes the channel
input to the Down Mixer.
• 7.1 SDDS
• 7.1
Phase
When enabled, the Phase button inverts the phase
of the channel input to the Down Mixer.
Chapter 74: Down Mixer
371
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-thanstereo 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.
372
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 75: 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 75: SignalTools
373
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
374
Audio Plug-Ins Guide
SignalTools Display Options
Both SignalTools plug-ins offer two display options: Phase Meter Display and Leq(A) Meter Display.
To choose a display option:

Click the corresponding button in the Options
section of the plug-in window.
Selecting SurroundScope channels for phase
metering
PhaseScope With PhaseScope, the left and right
channels are always compared.
SignalTools display options
SignalTools Phase Meter
Display
The Phase Meter indicates the phase coherency of
two channels of a multi-channel signal.
SignalTools Phase Meter
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.
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.
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).
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
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.
Chapter 75: SignalTools
375
Selecting Channels for Leq(A) Metering
SurroundScope With SurroundScope, you can se-
lect 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.
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.
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.
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
376
SignalTools level meter types button
choose the length of time the signal is measured
before an average value is calculated. Settings
range from 1 second to 2 minutes.
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.
versions of the Avid SurroundScope plug-in and
uses the same “true” RMS metering scale.
Audio Plug-Ins Guide
RMS (Root Mean Square) was used in previous
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 indi-
cation. 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).
Chapter 75: SignalTools
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Audio Plug-Ins Guide
Chapter 76: TL AutoPan
TL AutoPan is an automatic panning plug-in that is
available in AAX, 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
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379
TL AutoPan Panner Controls
The Panner section provides different controls for
different output channel configurations.
TL AutoPan in mono-to-stereo and mono-to-LCR
formats provide controls common to all output
configurations: Output, Width, and Manual.
TL AutoPan mono-to-quad and mono-to-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.
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.
Output
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
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.
Manual
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 affected by the setting
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Audio Plug-Ins Guide
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.
Place
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.
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.
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
Panner section, mono-to-5.0, clockwise path selected
The Spread slider is only available with mono-toquad 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 382). 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 384). The Envelope Detector can be triggered
by the panned audio signal, or by a side-chain input
(see “Using the Side-Chain Input” on page 385).
which the yellow Sound Location indicator travels
and indicates the panning field.
Path
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-to-quad and
mono-to-5.0 formats.
Path selectors, left to right path selected
Panning Source buttons
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381
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 381).
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
When the Panner section is set to Envelope
(ENV), the controls in the LFO section have
no effect on panning.
Rate
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.
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 384).
LFO Triggers
Single When the Single trigger is selected, the
LFO will cycle thru the waveform once only and
then stop.
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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 AutoPan,
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 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. 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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383
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 plugin window.
Side-Chain Input selector enabled
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Audio Plug-Ins Guide
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 “fly-around” 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.
2
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.
3
In the TL AutoPan Plug-In window, enable
Link To Tempo. This sets the LFO rate to follow
the session tempo.
4
Select the desired duration from the Duration
selector. For example, select 2 Beats.
5
Select the desired waveform for the LFO from
the Waveform selector. For example, select 4
Step Triangle.
6
Enable Beat Clock for the LFO Trigger. This
ensures that the LFO is synchronized to the
beat.
7
Play back the session to hear the panning effect.
3
In the TL AutoPan Plug-In window, select a
clockwise or counter-clockwise Path as desired.
4
Adjust the Spread and Width sliders as desired.
Try automating Spread and Width to alter the
positioning of the panned sound.
5
Try automating the Manual control instead of
using the LFO to create a more erratic panning of the “mosquito” sound.
6
Post Production Panning
7
To pan a mono track to 5.0 with TL AutoPan:
1
2
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.
Adjust the Rate slider as desired.
Try automating Rate to alter the speed of the
panned sound over time.
(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.
From the LFO Waveform selector, select Half
Sine.
Play back the session to hear the “mosquito”
flying around your head.
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.
Select a 5.0 output path from the track’s Output
selector.
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Audio Plug-Ins Guide
Part XII: Instrument Plug-Ins
Chapter 77: 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 plugins.
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.
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389
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.
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 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.
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.
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Audio Plug-Ins Guide
For more information on using the Copy and
Clear buttons, see “Creating a Drum Pattern
Using Boom” on page 395 and “Playing
with Patterns in Boom” on page 395.
Boom Instrument Section
Controls
Boom Global Controls
The Global Controls affect all instruments at once.
Each of Boom’s 10 instruments has a set of controls that set its pan position, volume level, tuning
(pitch), and decay (length).
Global controls
Shuffle Adds a variable amount of rhythmic swing
to the currently playing pattern.
Volume Controls the plug-in’s overall output vol-
ume.
Dynamics Scales the difference in volume between the pattern sequencer’s three possible Velocity levels.
Boom instrument strips
Boom Transport Controls
Pan Sets the current instrument’s pan position 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 instru-
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.
ment in varying ways.
Sample Selector Sets the current instrument’s
sample (10 samples available for each instrument).
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391
Boom Kit Selector
Boom Speed Switches
Kit Selector menu
The Kit Selector menu gives you access to the 10
preset drum kits in Boom.
The available kits are as follows:
392
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.
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.
Audio Plug-Ins Guide
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.
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.
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.
Patterns can be selected using MIDI notes
at any time, regardless of Edit Mode switch
position.
Boom Event Bar
The Event Bar is where most of the work of creating and playing patterns in Boom is done.
Info display
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.
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393
Boom Setup Page
Click the Setup button to view the Setup page. It
has two parameters you can set that change
Boom’s behavior.
Inserting Boom 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.
To insert an instrument plug-in on an Instrument
track:
1
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:
Beat Boom starts playing the selected pattern from
the step that corresponds with the incoming MIDI
note’s place in the current bar.
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.
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Audio Plug-Ins Guide
Create a new stereo Instrument track (recommended) in your Pro Tools session by doing the
following:
• Choose Track > New.
• Select 1 new Stereo Instrument track in Ticks.
• Click Create.
2
Click the Pro Tools Track Insert selector and select an instrument.
3
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.
See the Pro Tools Reference Guide for
instructions on how to use the MIDI sequencer in Pro Tools.
Creating a Drum Pattern
Using Boom
Saving a Boom Pattern as a
Preset
This section will get you started on the process of
creating beats with Boom. First, you’ll need an
empty pattern to edit.
You can save Boom settings, (in this case, your
drum patterns and instrument settings) as plug-in
settings files (.tfx), or presets.
To clear a pattern:
1
Set the Edit Mode switch to Pattern Select
mode.
2
Click one of the Event switches to select the
pattern you want to clear. For now, click the
Event switch marked 1.
3
Click the Clear button above the Matrix display.
All notes in the selected pattern will be cleared.
Plug-Ins Settings Select Button
To save a Boom pattern:
1
Click the Plug-In Settings Select button to open
the Plug-In Settings dialog.
2
Choose “Save Settings As...”, choose a name
for your preset, and click “Save”.
To create a new pattern:
1
Set the Edit Mode switch to Pattern Edit mode.
2
Press play on Boom’s transport.
3
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.
4
In the Event Bar, try out various rhythms by toggling the Event switches on and off.
5
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.
6
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 switch between patterns on-the-fly:
1
Press Play on Boom’s transport.
2
Set the Edit Mode switch to Pattern Select
mode.
3
Click the Event switch marked 2. A different
pattern will start to play, and its notes will appear in the Matrix display.
4
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.
Continue adding parts until you’re satisfied
with the pattern.
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395
To copy a pattern:
1
2
3
4
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.
Click Event switch 2. The Event Bar returns to
its normal state, and pattern 1 will be copied to
pattern 2.
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.
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.
To create a simple fill:
1
2
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.
Event switches 13–16, toggled on to create a roll
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Audio Plug-Ins Guide
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 cur-
rent 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 pre-existing MIDI sequences.
MIDI Note
Instrument Played
C1
Kick
C#1
Rim
D1
Snare
D#1
Clap
E1
Snare
F1
Lo Tom
F#1
Clsd Hat
G1
Lo Tom
G#1
Clsd Hat
A1
Hi Tom
A#1
Open Hat
B1
Hi Tom
C2
Hi Tom
C#2
Crash
D2
Hi Tom
D#2
Ride
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.
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
1
Record-enable the Instrument track on which
Boom is inserted.
2
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.
3
If you press and hold a note, the corresponding
pattern plays until you release the note.
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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397
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.
3
Click the Event switch that corresponds with
the pattern you want to start the chain with.
4
Hold Control on the computer keyboard and
click the Event switches that correspond with
each pattern you want to add to the chain.
5
Boom will play the patterns you’ve chosen in
the order they were added to the chain.
6
To remove a pattern from the chain, hold Control and click its Event switch.
7
To go back to playing patterns individually,
click any Event switch without holding down
Control.
To create a pattern chain with MIDI notes
398
1
Press and hold a MIDI note between C3 and D3
on your MIDI controller, triggering a pattern.
2
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.
3
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.
Audio Plug-Ins Guide
4
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.
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.
Assigning MIDI Controllers to
Boom Controls
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 realtime control from a MIDI keyboard or control surface. See Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
Chapter 78: 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.
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:
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.

• Portamento
• Velocity-sensitive gain and detuning
• Time-slice switching using envelope triggering
or MIDI beat clock
• Voice-stacking
• Side-chain input for control using an external
audio source
• Supports sample rates up to 192 kHz
• Online help
Reso Features
Reso features include:
Reso uses a resonator, which adds harmonic
overtones to the source audio through a short signal delay line with a feedback loop.
• Harmonic resonance generation
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.
• Multi-voice detuning

• Up to 62 voices of polyphony (on Pro Tools|HD
Accel systems)
• Resonant low pass filter
• Editable ADSR envelope generator
Bruno/Reso Features
Bruno Features
• Portamento
• Velocity-sensitive resonance, damping, gain,
and detuning
Bruno features include:
• Harmonic switching using envelope triggering
or MIDI beat clock
• Time-slice tone generation with adjustable
crossfade
• Voice-stacking
• Polyphony: Up to 62 voices of polyphony (on
Pro Tools|HD Accel systems)
• Multi-voice detuning
• Side-chain input for control using an external
audio source
• Supports sample rates up to 192 kHz
• Online help
• Editable ADSR envelope generator
Chapter 78: Bruno and Reso
399
Bruno/Reso DSP
Requirements
Bruno and Reso each require one full DSP chip on
a Pro Tools|HD card.
3
Play Bruno/Reso with the on-screen keyboard
or by MIDI control (see “Playing Bruno/Reso”
on page 400).
4
Adjust Bruno/Reso controls to get the effect
you want.
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 62voice 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 systems 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).
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.


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
Inserting Bruno/Reso onto an
Audio Track
Notes played with the on-screen keyboard are triggered at a MIDI velocity of 92.
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 play Bruno/Reso with the on-screen keyboard:
To add Bruno/Reso as a track Insert:
400
1
Click the Insert selector on the desired track and
select Bruno or Reso.
2
Click Play on the Pro Tools Transport to start
audio playback.
Audio Plug-Ins Guide
1
Open the plug-in window for Bruno/Reso.
2
Click Play on the Pro Tools Transport to start
audio playback.
3
Click the on-screen keyboard. Bruno/Reso will
only produce sound while audio plays on the
source track.
To latch keys on the on-screen keyboard:
1
Click the Latch bar, then click multiple keys.
Chords can be played in this way.
2
To turn off a latched key, click it a second time.
3
To turn off key latching entirely, click the Latch
bar a second time.
To play Bruno/Reso with a MIDI controller:
1
Start audio playback.
2
Play your MIDI keyboard while audio plays.
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.
2
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.
3
Start Pro Tools playback.
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
Click the track’s MIDI Output selector and select Bruno or Reso.
Using an External Key Input
with Bruno/Reso
(Side-Chain Processing)
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.
4
Record-enable the Instrument or MIDI track.
5
Test your MIDI connection by playing notes on
your MIDI keyboard. The corresponding notes
should highlight on Bruno/Reso’s on-screen
keyboard.
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).
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.
Chapter 78: Bruno and Reso
401
To use a key input for side-chain processing:
1
Click the Key Input selector and choose the input or bus with the audio you want to use to trigger the plug-in.
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.
Selecting a Key Input
2
Click the Key Input button (the button with the
key icon above it) to activate external sidechain processing.
3
Begin playback. The plug-in uses the input or
bus that you chose as a side-chain input to trigger the effect.
4
To hear the audio source you have selected to
control side-chain input, click the Key Listen
button (the button below the Ear icon).
Remember to disable Key Listen to resume
normal plug-in monitoring.
5
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.
By carefully choosing the type of source audio, the
crossfade length, and the type of switching, you
can create unique and complex sound
textures.
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Audio Plug-Ins Guide
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 wave-sequenced 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 401.
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 78: Bruno and Reso
403
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.
Spread
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).
404
Audio Plug-Ins Guide
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 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 millisec-
onds 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 millisec-
onds 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 78: Bruno and Reso
405
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.
406
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.
Reso
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 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 78: Bruno and Reso
407
Reso Resonance Controls
Reso Damping Controls
Resonance Amount
Damping 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.
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 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.
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.
Resonance Velocity
Damping 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.
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.
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.
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
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.
Resonance Velocity has an effect only when
you play Reso with a velocity-sensitive MIDI
controller.
408
Audio Plug-Ins Guide
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.
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.
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 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 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 401.
Threshold
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.
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 78: Bruno and Reso
409
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.
Amplitude controls
Gain Amount
Mix
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).
Mix adjusts the mix of the processed audio with
the original, unprocessed audio.
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).
Spread
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.
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Audio Plug-Ins Guide
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 millisec-
onds 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.
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
Reso Pitch Controls
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
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 used on synthesizers.
Detune Velocity
Detune Velocity controls how MIDI key velocity
affects voice detuning. This gives you touch-sensitive control over voice detuning when you play
Reso with a MIDI keyboard.
Chapter 78: Bruno and Reso
411
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 cutoff
Mono (Monophonic)
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.
412
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
Chapter 78: Bruno and Reso
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Audio Plug-Ins Guide
Chapter 79: 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:
1
Select Options > Click to enable the Click option
(or enable the Metronome button in the Transport).
Click plug-in
2
Create new a mono Auxiliary Input track and
insert the Click plug-in.
Click Controls
3
Select a click sound preset.
4
Choose Setup > Click/Countoff and set the Click
and Countoff options as desired.
MIDI In LED Illuminates each time the Click plugin 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.
Unaccented Controls the output level of the unac-
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.
cented beats of the audio click.
Chapter 79: Click
415
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.
See the Pro Tools Reference Guide for more
information on configuring Click options.
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Audio Plug-Ins Guide
Chapter 80: 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.
Chapter 80: DB-33
417
DB-33 Organ Page Controls
Scanner Vibrato
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.
Like many vintage organs, DB-33 features vibrato/chorus to animate the organ sound.
Tonewheels
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.
Scanner Vibrato controls
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:
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Wave Shape
Description
Tonewheels control
V1, V2, V3
The Tonewheels control sets the condition of the
tonewheels, and even provides a couple of nonstandard choices.
Vibrato (pitch modulation) effect.
Higher numbered modes offer
stronger modulation
C1, C2, C3
Chorus (timbral modulation)
effect. Higher numbered modes
offer stronger modulation
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
Audio Plug-Ins Guide
Drawbars
Percussion
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.
DB-33’s percussion feature adds a short burst of
additional harmonics at the beginning of each note
played.
Percussion controls
On/Off Turns the percussion feature on and off.
Drawbars
From left to right, each drawbar controls a different part of the harmonic spectrum, ranging from
low fundamentals to high harmonics.
When a drawbar is pulled out (downward), the volume of the corresponding harmonic is increased.
When pushed in, (upward) it is decreased.
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.
Loud/Soft Sets the volume of the added harmonic
burst.
Short/Long Sets the length of the harmonic burst.
3rd/2nd Sets the harmonic that is added, either the
3rd or 2nd harmonic.
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 420. for more information.
Key Click control
Turn the knob to set the level of click, from zero to
full.
Master Level control
Turn this knob to set the overall output volume.
This control is set, by default, the MIDI CC 7.
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419
Rotation Speed Switch
Input
This control switches the speed of rotation for the
rotating speaker cabinet.
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.
Rotation Speed switch
Increasing either of these inputs to very high levels
drives the tube pre-amp harder, sometimes leading
to pleasant (or not-so-pleasant) distortion.
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.
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.
Cabinet page overview
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Audio Plug-Ins Guide
Input controls
External Sets the volume of incoming signal when
DB-33 is used as an insert effect.
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
Mics
The Tube Pre-amp section offers control over the
preamplifier that precedes the rotating speaker
cabinet.
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.
Tube Pre-Amp controls
Character Effects the tonal balance of the signal.
Mics controls
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.
Drum/Horn Controls the mix between the low-end
speaker (drum) and the high-end speaker (horn).
Drive Sets the amount of gain in the pre-amp, rang-
sponse 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.
ing 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 trebleboosted tone.
Spread Sets the angle, and thus, the stereo re-
Chapter 80: DB-33
421
Speed Control
The Speed Control section affects the rotating
speaker cabinet’s speed of rotation, and the time it
takes to change between speed modes.
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:
1
Create a new stereo Instrument track (recommended) in your Pro Tools session by doing the
following:
Speed controls
• Choose Track > New.
Rotation Speed Switch A duplicate of the switch
• Select 1 new Stereo Instrument track in Ticks.
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.
Acc/Dec Sets the amount of time it takes for the ro-
tating speaker to move from one speed to another.
DB-33 Info Display and
Organ/Cabinet Switches
• Click Create.
2
Click the Pro Tools Track Insert selector and select an instrument.
3
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.
See the Pro Tools Reference Guide for
instructions on how to use the MIDI
sequencer in Pro Tools.
Assigning MIDI Controllers to
DB-33 Controls
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.
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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 realtime control from a MIDI keyboard or control surface. See Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
Chapter 81: 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 plugins.
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.
Chapter 81: Mini Grand
423
Mini Grand Main Controls
Model This knob selects between seven different
piano models that range from dark and mellow
(Atmo) to bright and aggressive (Dance).
Room This control switches between various room
ambiences. Effects range from natural
reverbs to special effects.
Room control
Model knob
Mix The Mix knob blends the desired amount of
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.
Tuning scale
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Audio Plug-Ins Guide
Mix knob
Level Turn this knob to control the overall output
volume of Mini Grand.
Level control
Mini Grand Info Display and
Setup Button
Model
Description
Real
Natural and dynamic, little processing
Bright
Controlled and percussive, lowered bass and low-mids
Hard
Loud, pointed, with accentuated
highs, lowered bass and lowmids
Dance
Loud, aggressive and gritty,
strong upper mid and treble presence, scooped lows
Info display (left) and Setup Button (right)
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
Room
Description
Soft
Mellow, sweet reverb
Shaping Mini Grand’s Sound
Bright
Heavier early reflections, accentuated highs
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.
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
Model
Description
Atmo
Quiet and dark, accentuated lowmids and muted highs
Soft
Mellow and rounded, with accentuated lows and low-mids, and
mellow highs
Ballad
Dynamic, but understated, warm
lows
See the Pro Tools Reference Guide for instructions on how to access and save Presets within Mini Grand.
Chapter 81: Mini Grand
425
Inserting Mini Grand on a
Track
Assigning MIDI Controllers to
Mini Grand Controls
To use an instrument plug-in to its best advantage,
insert it on a stereo Instrument track in your
Pro Tools session.
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 realtime control from a MIDI keyboard or control surface. See Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
To insert an instrument plug-in on an Instrument
track:
1
Create a new stereo Instrument track (recommended) in your Pro Tools session by doing the
following:
• Choose Track > New.
• Select 1 new Stereo Instrument track in Ticks.
• Click Create.
2
Click the Pro Tools Track Insert selector and select an instrument.
3
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.
See the Pro Tools Reference Guide for
instructions on how to use the MIDI sequencer in Pro Tools.
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Audio Plug-Ins Guide
Chapter 82: 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

Chapter 82: Structure Free
427
Smart Knobs The Smart Knobs are special con-
Info Display The Info display above the Keyboard
trols 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.
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.
Smart Knobs
To display a control’s current value:

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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Audio Plug-Ins Guide
Info display
Editing a Patch Comment
To edit a patch comment:
1
Select a patch.
2
Double-click into the Info display.
3
Type in your comment.
4
Press Enter.
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 sub-pages.
Controls” on page 433 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.
Add Patch The Add Patch submenu lets you add a
Patch list
Structure Free Patch Module
Controls
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.
Remove Patch The Remove Patch entry unloads
the selected patch removing it from 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 Structure
Free Patches. See “Structure Free Browser Page
Remove All Patches The Remove All Patches entry 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.
Copy Patch The Copy Patch entry copies the se-
lected patch to the clipboard.
Chapter 82: Structure Free
429
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 Parameter entry inserts only the parameter settings of the
copied patch to the selected patch.
Selected Patch copies the samples of the selected
patch to disk.
Loading a Patch from the Structure Free Patch
Menu
All Patches copies the samples of all patches of
To load a patch from the Patch menu:
Session copies the samples of all patches of all
the Structure Free instance to disk.
Structure Free instances in your session to disk.
1
Go to the Patch menu and click Load Patch.
2
In the following file dialog, locate and select a
patch.
Adding Additional Structure
Free Patches
3
Click OK.
Additional factory patches for Structure Free can
be downloaded from Avid’s website
(www.avid.com).
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 Samples to Session Folder function to transfer the loaded samples
430
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.
Audio Plug-Ins Guide
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:
1
Download the Structure Free patches from the
Avid website (www.avid.com). After downloading, make sure the patches are uncompressed.
2
Drag the uncompressed downloaded patches
into the Structure QuickStart folder, located on
your computer at the following location:
• Applications/Digidesign/Structure/Structure
QuickStart (Mac)
• 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.
Selecting the Edit 1 sub-page
Pitch Bend Down Sets the downward pitch bend
range for the patch in semitones.
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.
Chapter 82: Structure Free
431
FX Send On Activates the Effect Send for the
patch.
Amp Envelope Section
Attack Softens the attack phase of Instruments by
Structure Free Edit 2 Sub-Page Controls
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.
Filter Section
Hold Adjusts the length of the Amp envelope’s
Filter Type Selects a filter type.
Hold time at the end of the attack phase.
Cutoff Adjusts the filter cutoff frequency.
Decay Shortens the played instrument hits by ap-
FX Send Level Adjusts the level sent from the
patch to the Effect Send.
plying an amplitude decay after the hold time.
Envelope Level Adjusts how strongly the filter en-
Filter Envelope Section
Sustain Adjusts the level of the sustain segment.
The envelope’s signal remains at this level as long
as the note is held.
Attack Sets the time needed for the filter envelope
Release Adjusts the time for the release segment
velope modulates filter cutoff.
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 re-
lease 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.
Amplifier Section
Vel Sens (Velocity Sensitivity) Adjusts the envelope velocity sensitivity (range in dB between lowest and highest velocity).
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Audio Plug-Ins Guide
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
The Browser page provides the following controls:
Patch Activates the displaying of only patches.
Parts Activates the displaying of only parts.
Sample Activates the displaying of only samples.
Show All Activates the displaying of all file types.
Previous Directory Navigates to the previous
folder.
Next Directory Navigates to the next folder.
Directory Up Navigates one folder level up.
Show Favorites Shows your Favorite folders.
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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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:
1
Create a new stereo Instrument track in
Pro Tools.
2
Click the track’s Insert selector and choose
Structure Free from the list.
To load a patch from the browser:

Drag a patch into the Patch list to load it.
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.
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.
2
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.
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 Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”.
The default Sine Wave patch
3
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Audio Plug-Ins Guide
Next, load a Structure Free patch.
Loading a Structure Free Patch
To load a patch from the Browser:
1
Click the Browser tab in the Parameter panel to
display the Browser page.
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
2
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:
To find missing samples for a patch:
1
Click the Patch menu and select Find Missing
Samples from the menu.
2
In the following dialog, navigate to the new
sample location and click OK.
Windows Program Files\Digidesign\
Structure\Structure QuickStart
Full Recursive Search 3Searches for missing sam-
Mac OS X /Applications/Digidesign/
Using Structure Free Smart
Knobs
Structure/Structure QuickStart
3
Drag the Patch named 01 Six String Guitar.patch onto the Sine Wave Patch to load it and
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.
4
After loading, the multi-purpose display shows
a short description of the Patch, and the Parameter panel above displays its Patch parameters.
5
Play some notes and chords. Adjust the Patch
volume using the horizontal fader on the Patch
module in the Patch list.
ples in the specified folder and all its subfolders.
The Smart Knobs are special controls which can be
assigned to one or more Structure Free parameters
in the currently selected patch.
To use Structure Free Smart Knobs:
1
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.
2
Set the Smart Knob for Delay Mix to 30%.
3
Set the Smart Knob for Chorus Mix to 65%.
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435
4
Play some notes and chords. Set the other Smart
Knobs at will.
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:
1
Load Patch 04 Electronic Drum Kit.patch, and
play with it on your keyboard.
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.
3
Click the second Key Switch C#0, or play the
corresponding key to add dirt to the kit’s sound.
Key Switches
436
4
Try out the other Key Switches.
5
The synth pad patch has Key Switches too.
Audio Plug-Ins Guide
Assigning MIDI Controllers to
Structure Controls
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 realtime control from a MIDI keyboard or control surface. See Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
Chapter 83: 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 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
TL Drum Rehab is a mono plug-in only. It
cannot be used on multi-channel tracks
(stereo or greater).
• Custom file format (DRP)
• Tracking, compression, and quantization
• Triggering sensitivity and filtering controls
• Random sample selection
• No Latency mode
TL Drum Rehab plug-in
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437
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.
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 443).
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.
2
Make a short selection to set TL Drum Rehab’s
parameters. For example, make a two bar selection.
3
In TL Drum Rehab’s Trigger panel (see “TL
Drum Trigger Panel Display and Controls” on
page 443), select Kick from the Detector Mode
pop-up menu (see “TL Drum Rehab Detector
Mode Menu” on page 444).
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 448).
To edit sample layers and adjust the sound of samples, use the Samples panel (“Samples Panel Display and Controls” on page 452).
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 438)
uses TL Drum Rehab to replace the kick drum
sound on a mono kick drum track in real-time.
• The second example (“Replacing and Quantizing a High Hat Using TL Drum Rehab” on
page 441) describes a more complicated procedure, using TL Drum Rehab’s Expert panel to
replace a high hat track and quantize the replacement samples.
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Audio Plug-Ins Guide
Detector Mode pop-up menu (left) and Trigger Panel
button (right)
4
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.
6
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 447).
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.
5
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.
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 455),
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 444.
Detected triggers
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439
9
Do one of the following:
10
• To load a DRP file into TL Drum Rehab, double
click the desired DRP file in the Library
browser.
• 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 445).
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 447.)
You can also use the Ducking control to mask
track’s audio with the triggered sample (see
“Playback Controls” on page 447).
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.
12
Auto-Audition enabled
Selected sample
Once you are satisfied with the result, do one of
the following:
• 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
For more information on working with
committed triggers, see “TL Drum Rehab
Commit Button” on page 449.
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:
• 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.
Committed triggers
4
• Enable Listen mode.
• Play back the selection to detect triggers.
3
In the Expert panel, click Commit All.
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.
Commit All button (left) and the Expert panel button
(right)
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 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.
Uncommitted trigger (left) and an ignored trigger
(right)
5
Disable Listen mode.
6
For no latency on playback, enabled No Latency mode (see “Triggering Controls” on
page 447).
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441
7
Select the desired quantize resolution from the
Quantize To pop-up menu (see “TL
Drum Rehab Quantize To Menu” on page 452).
8
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).
9
Adjust TL Drum Rehab’s playback controls as
desired (see “Playback Controls” on page 447).
10
Once you are satisfied with the result, do one of
the following:
• 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 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 455.
TL Drum Rehab Main Library browser
TL Drum Rehab Main Window
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 443).
TL Drum Rehab Controls and
Displays Overview
Expert Panel Lets you precisely edit the placement
When using TL Drum Rehab, most operations take
place in one of two displays: the Main window and
the Library Browser.
Samples Panel Lets you view and manage drum
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 442.
of triggers (see “TL Drum Rehab Expert Panel
Display and Controls” on page 448).
samples loaded into TL Drum Rehab (“Samples
Panel Display and Controls” on page 452).
Preferences Panel Lets you edit TL Drum Rehab’s preferences (see “TL Drum Rehab Preferences Panel Display and Controls” on page 454).
A Note About TL Drum Rehab Control Sliders
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.
TL Drum Rehab Main window
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Audio Plug-Ins Guide
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.
A/B Blend slide, a global control
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 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.
To access the trigger panel:

Click the Trigger Panel button.
Quantize slider, a unique control
Not all sliders are active controls in every panel.
For example, the last slider in the Trigger panel is
grayed out.
Trigger panel button
TL Drum Rehab Waveform Display
Inactive slider
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.
Detected triggers
Detected amplitudes
Waveform display with detected triggers and
amplitudes
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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 447).
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:

Selecting the detection algorithm from the Detector
Mode pop-up menu
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Audio Plug-Ins Guide
In the Library browser, locate and double-click
the DRP file you want to load. All samples in
the DRP file are loaded into their assigned
Zones and Clips.
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 447.
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 452).
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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445
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 447.
Use the Velocity Map to select a Zone for loading
a sample (see “Loading a Sample into a Zone” on
page 446) and also to adjust the crossfade between
Zones (see “Adjusting the Crossfade Between
Zones” on page 447). Using multiple Velocity
Zones lets you layer samples by dynamics for
more realistic drum sample playback. Use the leftmost 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 452).
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Audio Plug-Ins Guide
Loading a Sample into a Zone
To load a sample into a Zone:
1
Click the Zone in the Velocity Map where you
want to load a sample. The selected Zone is indicated by a white triangle.
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).
3
Double-click the audio file (WAV, AIF, or SD2)
you want to load into the selected Zone.
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 456.
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.
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 449). For most uses of TL Drum Rehab, Listen is enabled.
No Latency Enable the No Latency button to play
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.
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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447
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.
A/B Blend Controls the mix between samples
loaded into Positions A and B in the Samples panel
(see “Position A/B” on page 452). 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.
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Audio Plug-Ins Guide
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.
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 447).
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 443). 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 enabled (see “Tempo
Changes” on page 454), 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.
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.
To commit detected triggers:
Waveform display in Expert mode with detected
triggers and amplitudes, and Tempo events
1
Listen for triggers (see “Triggering Controls”
on page 447).
2
Select the Expert panel.
3
Do one of the following:
• Click Commit All to commit all detected triggers.
• Click Commit and then click only the triggers
you want to commit. Committed triggers are indicated by a red arrow.
Committing specific detected triggers
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449
To play back only committed triggers:
1
Deselect Listen.
2
Start playback.
To edit the position of a committed trigger:
1
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.
Editing the location of a committed trigger
2
While still holding down the mouse, move the
trigger left or right until it is at the desired location.
3
Release the mouse.
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 change the amplitude of a committed trigger:

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.
To uncommit triggers, do one of the following:

In the Expert panel, click Uncommit All to uncommit all triggers.

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
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Audio Plug-Ins Guide
TL Drum Rehab Ignore Button
TL Drum Rehab Add Button
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.
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.
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.
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
Ignoring specific detected triggers
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.
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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.
Samples Panel Display and
Controls
The Samples panel lets you load, view, shape, and
organize samples for playback.
To access the Samples panel:

Click the Samples Panel button.
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.
Samples panel button
Position A/B
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 greater degree of
realism by adding variety to the sound. For exam452
Audio Plug-Ins Guide
ple, 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 add a sample to a Clip:
Play
In the Samples panel, click Play to audition the
currently loaded sample for the selected Zone and
Clip.
Clear
1
In the Samples panel, select the Velocity Zone
to which you want to add a sample.
In the Samples panel, click Clear to clear the currently loaded sample for the selected Zone and
Clip.
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.
Velocity Map
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
Repeat steps 2–3 as desired.
5
From the Clip Playback Mode pop-up menu, select Cycle or Random to determine whether the
clips playback in sequence or in random order.
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 445).
Invert
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.
Sample Name Display
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 444).
# of Zones
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 445).
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.
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 445).
Sample Name display
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TL Drum Rehab Samples Panel
Display Waveshaping Controls
To access the Preferences panel:

Click the Preferences Panel button.
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 characteristics 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%.
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.
Freq Adjusts the frequency of the EQ for all clips
in the current position (A or B). The EQ Frequency
slider has a range of 10 Hz to 15 kHz.
Q Adjusts the Q of the EQ for all clips in the cur-
rent position (A or B). The Q slider has a range of
0.1 to 6.0.
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.
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Audio Plug-Ins Guide
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.
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 449).
This preference is set to Hide by default.
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.
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 456.
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 currently
loaded into Clips and Zones as a new DRP file.
Save DRP File Saves any edits to the currently
loaded DRP file.
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 456).
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.
Library
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.
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455
Edit
Use the Edit pop-up menu to Add or Remove Favorites, and organize your Favorites in folders.
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 se-
Loading Samples and Saving
Custom DRP Files in
TL Drum Rehab
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.
lected Favorites folder.
Loading samples and saving a custom DRP file:
Auto-Audition
1
Insert TL Drum Rehab on a mono audio track.
Enable Auto-Audition to hear drum samples in the
Library browser automatically when you click
them. Use the slider to adjust the audition volume.
2
In the Library browser, select File > Show All
Volumes. The Library browser displays the root
level of your computer.
3
Auto-Audition
Help
The Help button at the top of the Main window
turns TL Drum Rehab Help Balloons on or off.
TL Drum Rehab Help Balloons
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Audio Plug-Ins Guide
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.
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.
4
To audition a file before importing it, enable
Auto-Audition and click the sample name in the
Library browser.
8
In the Library browser, double-click the desired
audio file (WAV, AIF, or SD2) to load it into the
selected Zone.
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 452.)
5
Select the Samples panel (see “Samples Panel
Display and Controls” on page 452).
6
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 453.)
7
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 left-most
Zone first. (See “TL Drum Rehab Velocity Map
and Velocity Zones” on page 445.)
10
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.
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.
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457
458
12
Select File > Save New DRP File. The new DRP
file appears highlighted at the top of the browser
list as “Drum Samples.drp.”
13
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.”
14
Press Enter (if you do not press Enter, the new
DRP will not be saved). The new DRP file appears in the current directory.
15
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.
Audio Plug-Ins Guide
Chapter 84: 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.
2
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.
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.
2
Create a new audio, Auxiliary Input, or Instrument track.
3
Insert TL Metro on the new track.
4
Ensure that Options > Click is enabled.
Factory Presets
TL Metro provides a number of factory presets that
provide a range of sounds.
To audition a preset:
1
Select the desired preset from the Plug-In Librarian menu.
2
Click Play in TL Metro.
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459
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 461.
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 462.
Play Button
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.
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.
• Accented Quarter Note = Beat 1
• Quarter Note = Beat 2
• Eighth Note = Beat 3
• Sixteenth Note = Beat 4
Tap Button
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
The LCD style information display in TL Metro
displays the following:
• Triplet = Beat 5
Synchronizing TL Metro to
Pro Tools
TL Metro can be synchronized to the Pro Tools
Transport and Tempo using the Link Status selector.
• 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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461
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 459).
• 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.
To make any preset the default when TL Metro is
instantiated:
1
2
From the Plug-In Librarian menu, select the desired preset.
From the Plug-In Settings menu, select Set As
User Default.
3
From the Plug-In Settings menu, select Settings
Preferences > Set Plug-In Default To > User
Setting.
For more information on using plug-in presets in Pro Tools, see the Pro Tools Reference Guide.
Linked to Transport and Tempo
Importing Custom Samples to
TL Metro
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.
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>.”
Ensure that MIDI is correctly configured for TL
Metro in Pro Tools (see “Configuring Pro Tools
for Use with TL Metro” on page 459).
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.
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.
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.
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Audio Plug-Ins Guide
To import a sound:
1
Click the Sound Library button to display the
sample menu.
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.
Chapter 84: TL Metro
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Audio Plug-Ins Guide
Chapter 85: 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 85: Vacuum
465
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.
(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 os-
cillation 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.
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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Audio Plug-Ins Guide
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
Cutoff Sets the frequency at which the given filter
Mixer section
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.
VTO 1 and VTO 2 Sets the relative volume of the
Slope Sets the curve of the filter slope. At higher
two oscillators. One (or even both) oscillators can
be reduced to silence, if needed.
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.
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.
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-wave-like
tone. At lower values, the filter simply cuts off the
specified frequencies.
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467
Env 1 Controls the amount that the filter cutoff fre-
quency 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 playing
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,
lope’s modulation to reach its highest point when a
MIDI note is received.
Decay The amount of time it takes for the enve-
lope’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 enve-
changing the tonal quality of the current filter from
soft to aggressive and distorted.
lope’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 ve-
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 469. for more information.
Env One controls
468
Attack The amount of time it takes for the enve-
Audio Plug-Ins Guide
locity 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.
As the Vel control is moved to the right, more and
more modulation occurs relative to incoming note
velocity.
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 Sets which parameter is modulated.
The choices are:
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 occurs.
At its center, no modulation will occur. To the
right, modulation increases, and to the left, modulation also increases, but with reversed polarity.
Modulation Routing controls
Source Sets what signal is used to modulate the
chosen parameter. The choices are:
Vacuum Age Controls
The Age section lets you explore the tonal effects
of aging internal circuitry and years of dust and
dirt.
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
Age controls
MW
The position of the Mod wheel
Drift Adds a variable amount of pitch drift to the
AT
The amount of Aftertouch, if supported by your MIDI controller
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, emulat-
ing the worn and dusty contacts often found on
older synths.
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469
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
Mode Sets the direction of the arpeggiator.
Vol Sets the overall volume.
Mode
Description
Shape Adds a variable amount of tube saturation
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.
to the final output signal.
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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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:
Pitch and Mod Wheel controls
Off No Glide.
Pch and Mod Onscreen wheels move along with
Held Only apply glide when more than one note is
incoming pitch bend and modulation MIDI messages. They can also be clicked and dragged like
other controls.
held at once.
On apply glide to every note.
Dest Sets what parameter is modulated when the
Pitch Bend Range
Mod wheel is moved upward.
This sets the range of the Pitch wheel, in semitones.
Dest
Description
Off
No modulation occurs
Envelope Retrigger
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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471
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:
1
Create a new stereo Instrument track (recommended) in your Pro Tools session by doing the
following:
• Choose Track > New.
• Select 1 new Stereo Instrument track in Ticks.
• Click Create.
2
Click the Pro Tools Track Insert selector and select an Avid Virtual Instruments.
3
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.
See the Pro Tools Reference Guide for
instructions on how to use the MIDI
sequencer in Pro Tools.
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Audio Plug-Ins Guide
Assigning MIDI Controllers to
Vacuum Controls
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 realtime control from a MIDI keyboard or control surface. See Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
Chapter 86: 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
Chapter 86: Xpand!2
473
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.
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.
Patch is another name for the plug-in settings.
Refer to the Pro Tools Reference Guide for information on working with RTAS plug-ins.
Xpand!2
Level control
Global Controls
Xpand!2 Smart Knobs
Xpand!2
The upper section of
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.
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.
The Part selector switches (A, B, C, or D) give access to the Smart Knob parameters for the selected
part.
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Audio Plug-Ins Guide
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 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
MIDI Channel selector
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.
For details on how to assign MIDI controllers, see Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
Category Selector
To view Parts organized by categories, click the
Category selector.
On/Off
Category selector
Activate or deactivate the Part by clicking its
On/Off button. When the Part is activated the Part
character in its center is lit.
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 Name
Part Selector
Click the Part selector to select the Part, so that its
Smart Knobs are displayed.
Level
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.
Part selector
Level slider
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475
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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Audio Plug-Ins Guide
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.
Chapter 86: Xpand!2
477
To define the key range for the current Part
1
Right-click the Upper/Lower key range limit
control and choose Learn.
2
Then press the appropriate key on your MIDI
keyboard.
To adjust the key range for the current Part

Click the Upper/Lower key range limit control
and drag up or down to increase or decrease its
value.
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).
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.
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.
Xpand!2 Arp Controls
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.
Rate
Arp parameters
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Audio Plug-Ins Guide
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.
Xpand!2 Mod Patch Edit
Controls
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.
Mod Wheel Destination
Click the Destination button (the lower Mod
Wheel button) to select a destination for the modulation from the pop-up menu.
Mod Wheel Destination button
Mod parameters
The following destinations for mod wheel action
are provided:
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.
Mod Wheel
Destination
Description
Pitch
Affects the Part’s pitch.
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.
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 “Random.” If the pop-up is set to
Const the movements of the modulation wheel will
directly modulate the destination without a time
varying waveform.
Mod Wheel Shape button
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479
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.
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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Audio Plug-Ins Guide
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.
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.
3
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.
See the Pro Tools Reference Guide for instructions on how to use the MIDI sequencer
in Pro Tools.
Pressure Depth control
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
Assigning MIDI Controllers to
Xpand!2 Controls
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 realtime control from a MIDI keyboard or control surface. See Chapter 88, “Using the MIDI Learn
Function on Avid Virtual Instruments.”
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:
1
Create a new stereo Instrument track (recommended) in your Pro Tools session by doing the
following:
• Choose Track > New.
• Select 1 new Stereo Instrument track in Ticks.
• Click Create.
2
Click the Pro Tools Track Insert selector and select an Avid Virtual Instruments.
Chapter 86: Xpand!2
481
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Chapter 87: 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 sample-accurate 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 plug-ins
• 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.
Chapter 87: ReWire
483
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
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.
• 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/24-bit 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.)
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:
1
Make sure that the ReWire client application is
installed properly and that you have restarted
your computer.
2
In Pro Tools, choose Track > New and specify
one Instrument track (or audio or Auxiliary Input track), and click Create.
Chapter 87: ReWire
485
3
In the Mix window, click the Insert selector on
the track and assign the ReWire RTAS client
plug-in to the track insert.
6
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.
4
Configure the ReWire client application to play
the sounds you want.
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.
Selecting the ReWire client device to receive MIDI
from Pro Tools (Instrument track shown)
7
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).
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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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.
2
From the track’s MIDI Input selector, select the
ReWire device that you want to record.
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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487
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:
1
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).
2
Start playback in Pro Tools.
3
Switch to the ReWire client application. Notice
that the corresponding parameter changes according the MIDI CC data from Pro Tools.
Adjusting a parameter in a ReWire client application
(Reason’s SubTractor shown)
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.
7
When you are done adjusting the parameter, return to Pro Tools and stop recording.
8
Record disable the MIDI track.
9
From the MIDI Track View selector in the Edit
window, select the view for the CC data you just
recorded.
MIDI CC data recorded from a ReWire client
application
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Quitting ReWire Client
Applications
When quitting Pro Tools sessions that integrate
ReWire client applications, quit the client application first, then quit Pro Tools.
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.
1
In the Pro Tools Timeline, select the time range
that you want to loop.
2
Begin playback by pressing the Spacebar or
clicking the Play button in the Transport.
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.
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.
• Draw the automation in the Edit window.
For information on drawing automation, see
the Pro Tools Reference Guide.
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Chapter 88: 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 realtime control from a MIDI keyboard or control surface. MIDI assignments are saved with the session.
Some MIDI continuous controllers are pre-assigned and cannot be learned, as follows:
Assigning a MIDI Controller
To assign a MIDI controller to an Avid Virtual
Instrument parameter, 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.

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.
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.
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MIDI Controller Assignment
Options
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:

492
Control-click or Right-click (Mac) or Rightclick (Windows) a control, and select Invert
Range.
Audio Plug-Ins Guide
Part XIII: Other Plug-Ins
Chapter 89: 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 495)
• Correlation Meter (see “BF Essential Correlation Meter” on page 496)
• Meter Bridge (see “BF Essential Meter Bridge”
on page 496)
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 496)
BF Essential Clip Remover
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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 (Aweighting 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.
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Audio Plug-Ins Guide
BF Essential Noise Meter
Chapter 90: Signal Generator
Signal Generator is a test tone generator plug-in
that is available in AAX, 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 audio 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 decibels.
Values range from a low of –95 dB to a high of
0.0 dB.
Signal These buttons select the waveform.
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 AAX and TDM versions of Signal Generator produce 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.
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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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 plug-in.
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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Chapter 91: 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
SoundReplacer
• Three separate amplitude zones per audio event
for triggering different replacement samples according to performance dynamics
Audio Replacement
Techniques
• Zoomable waveform display for precision
threshold/amplitude zone adjustment
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.
• Crossfading or hard-switching of replacement
audio in different amplitude zones for optimum
realism and flexibility
• Online help
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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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 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.
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.
Waveform display with trigger markers shown
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.
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 amplitudeswitched between the replacement samples, or amplitude crossfaded for seamless transitions.
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Audio Plug-Ins Guide
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
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.
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.
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.
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.
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.
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.
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SoundReplacer Zoomer
SoundReplacer Peak Align
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.
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.
• 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.
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.
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Audio Plug-Ins Guide
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.
For more information on using Peak
Align, see “Getting Optimum Results with
SoundReplacer” on page 505.
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:
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.

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.
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 505.
12 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.
13
To use SoundReplacer:
1
On the source track, select the audio you want to
replace. Only selected audio will be replaced.
2
Choose SoundReplacer from the AudioSuite
menu.
3
Click the Load Sound icon (the icon beneath the
yellow slider) to import the replacement sound
for amplitude zone 1.
4
Locate the desired audio file and click Open.
5
Adjust the amplitude zone slider.
6
Repeat steps 3–5 to load replacement sounds
into amplitude zones 2 and 3.
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 506.
7
8
9
Adjust the Threshold sliders to fine tune audio
replacement triggering.
10 Adjust
the Dynamics slider to fine tune how
SoundReplacer tracks and matches changes in
the source audio’s dynamics.
11
504
• To render the selected clip only in the track in
which it appears, choose Playlist from the Selection Reference pop-up.
• To render the selected clip in the Audio Clip List
only, choose Clip List from the Selection Reference pop-up.
14
Adjust the Mix slider to get the desired balance
between replacement audio and source audio.
Audio Plug-Ins Guide
Determine which occurrences of the selected
clip you want to render by doing one of the following:
• 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).
• If you do not want to update every occurrence of
the selected clip, disable Use In Playlist.
15
To align the amplitude peak in the replacement
file(s) to threshold trigger markers in the source
audio, enable Peak Align.
Click Preview to audition the replacement audio.
Render the selected clip by doing one of the following:
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 popup menu.
Because SoundReplacer does not allow
destructive rendering, the AudioSuite
Overwrite Files option is not available.
16
From the Destination Track pop-up, choose the
destination for the replacement audio.
17
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.
In general:
A slower-peaking kick drum
 Turn on Peak Align if you are replacing drum or
percussion sounds whose peak level occurs at the
initial attack.
The first figure shows a fast-peaking kick drum
whose peak level occurs at its 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 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 Command-drag
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 Sound-Replacer 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 92: 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 display
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.
Ratio Sets the destination length in relation to the
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.
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 milliseconds, 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 subtracting very small portions of audio material and
very quickly crossfading between these alterations
in the waveform of the audio material.)
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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.
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Accuracy Prioritizes the processing resources al-
located 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.
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.
Chapter 93: TL InTune
TL InTune is a professional instrument tuner plugin that is available in AAX, 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 plug-in
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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511
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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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 515.
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 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.
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.
Octave Buttons
Down Octave button
Up Octave button
Octave buttons
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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.
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.
TL InTune Tone Volume
The Tone Volume slider controls the volume of the
test tone audio signal.
TL InTune Information Display
The LCD style information display in TL InTune
displays the following:
• The reference frequency
• The current note to which TL InTune is
tuning
• The number of cents sharp or flat from the current note
Selecting a TL InTune preset
To make any preset the default when TL InTune is
instantiated:
1
2
From the Plug-In Librarian menu, select the desired preset.
From the Plug-In Settings menu, select Set As
User Default.
• The status of any test tones playing
3
From the Plug-In Settings menu, select Settings
Preferences > Set Plug-In Default To > User
Setting.
For more information on using plug-in
presets in Pro Tools, see the Pro Tools
Reference Guide.
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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.
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
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
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.
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.
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.
Exit
In the Tuner Programming screen, click the Exit
button to return to the main TL InTune screen.
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515
Using TL InTune
For best tuning results with guitars, do the
following:
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.
• Use headphones, as loud monitors can modulate
the guitar string.
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.
• Roll your guitar’s tone knobs all the way off to
remove all the highs.
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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• Switch your guitar to its rhythm (neck) pickup,
if it has one.
• 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 94: 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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517
TL Master Meter Overview
This section provides an overview of metering and
mastering, and how TL MasterMeter can help you
produce great sounding mixes.
The following four figures show how the same
complex waveform shown in the previous figure
can be represented in the digital domain.
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
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.
Waveform sampled
Waveform as represented in DAW
Sampling
A waveform can be represented in multiple ways
during the process of sampling, display and reconstruction.
Waveform as reconstructed at the D/A
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
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rate). The range below the Nyquist frequency defines the “legal” range of allowed frequencies as
frequencies in this range 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 is easily
identified: the samples themselves do not represent
the peak value of the waveform. The waveform is
only complete after the reconstruction process.
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519
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.
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!
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.
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, compressed even
more so with multi-band compressors, limited,
normalized, and maximized to get the audio to play
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D/A converter range
Seven consumer CD players were subjected to
tests [Nielsen 2003] 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.
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521
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 31-tap 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.”
TL MasterMeter can be used in two different ways
during a session: Real-Time Metering or Historical
Metering.
Real-Time Metering
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.
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.
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TL MasterMeter Controls and
Displays
TL Master Meter Browsers
Signal Clip Events Browser
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 ceases to
record additional events. Although the meters re-
main 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.
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.
The information in this browser is cleared using
the Clear button, or is cleared automatically whenever the Pro Tools transport is started.
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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523
TL MasterMeter Clear Button
TL MasterMeter Offset 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 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.
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.
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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.
TL MasterMeter Clip Field
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.
Chapter 95: Trim
The Trim plug-in is available in AAX, 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 95: Trim
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Chapter 96: Other AudioSuite Plug-In
Utilities
The following AudioSuite-only utility plug-ins are
installed when you install Pro Tools:
DC Offset Removal
• 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-todigital convertors) that can cause pops and clicks
in edited material.
• Duplicate
• Gain
• Invert
• Normalize
• Reverse
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 steady-state offset
from zero will appear in the waveform. Use the DC
Offset Removal plug-in to
remove it.
DC Offset Removal plug-in
To remove DC offset from an audio clip:
1
Select the clip with DC offset.
2
Choose AudioSuite > Other > DC Offset
Removal.
3
Ensure that Use In Playlist is enabled.
4
Click Render.
Chapter 96: Other AudioSuite Plug-In Utilities
527
Duplicate
Gain
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.
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.
Duplicate plug-in
Gain 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.
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.
Find Level When clicked, displays the peak
The Duplicate plug-in works nondestructively.
You cannot choose to overwrite files.
amplitude value of the current selection.
RMS/Peak Toggle Switches the calibration of gain
To duplicate an audio selection:
1
Select the audio you want to duplicate.
2
Choose AudioSuite > Other > Duplicate.
3
Ensure that Use In Playlist is enabled.
4
Click Render.
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.
To change the gain of an audio clip:
528
Audio Plug-Ins Guide
1
Select the clip whose gain you want to change.
2
Choose AudioSuite > Other > Gain.
3
Adjust the Gain slider as desired.
4
Click Preview to audition your changes.
5
Ensure that Use In Playlist is enabled.
6
Click Render.
Invert
Normalize
The Invert plug-in reverses the polarity of selected
audio. Positive sample amplitude values are made
negative, and all negative amplitudes are made
positive.
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).
This process is useful for altering the phase or polarity relationship of tracks. The Invert plug-in 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.
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).
Invert plug-in
To invert the phase an audio clip (or selection):
1
Select the clip whose phase you want to invert.
2
Choose AudioSuite > Other > Invert.
3
Ensure that Use In Playlist is enabled.
4
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 96: Other AudioSuite Plug-In Utilities
529
RMS/Peak Toggle Switches the calibration of nor-
malizing 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.
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.
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:
Reverse plug-in
Peak On Each Chan/Track Searches for the peak
To reverse an audio clip (or selection):
level on a channel-by-channel or track-by-track
basis.
1
Select the clip you want to reverse.
2
Choose AudioSuite > Other > Reverse.
3
Ensure that Use In Playlist is enabled.
4
Click Render.
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):
530
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.
Audio Plug-Ins Guide
Part XIV: Eleven
Chapter 97: 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.
• Support of up to 8 channel (7.1) operation, in
mono or multi-mono plug-in only.
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.
• Two speaker cabinet models.
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).
Eleven Free Plug-In Features
• Two custom amp models from Avid.
• 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 97: Eleven and Eleven Free
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Audio Plug-Ins Guide
Chapter 98: 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 539.
• 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 98: Eleven Input Calibration and QuickStart
535
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:
1
Do one of the following, depending on your
hardware configuration:
• If you are using an interface that has a DI input
(such as an Mbox Pro), plug your guitar into an
available DI input.
• 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.
2
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.
To connect your guitar to a Pro Tools|HD system:
1
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.)
2
Plug your guitar into an available pre-amp input
and set its source, impedance, and other settings
as needed for your setup.
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.
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:
1
In Pro Tools, choose Setup > Playback Engine
and set your Hardware Buffer to a low enough
setting to reduce monitor latency.
2
On your guitar, select the highest output pickup
or position and set the volume and tone controls
to 10 (maximum).
3
Strum full chords (your loudest expected playing) while watching the Input indicators on your
audio hardware.
4
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).
Set Up a Pro Tools Track
In this step, you’ll create and configure an audio
track to use for the final stage of input calibration.
To set up and check Track level (all systems):
1
Choose Tracks > New, and create one mono
Audio track.
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
Track meter
To prepare your guitar and Pro Tools|HD hardware
for input calibration:
1
On your guitar, select the highest output pickup
or position and set all volume and tone controls
to the maximum.
2
Strum full chords (your loudest expected playing) while watching the Input indicators on your
audio hardware.
3
Adjust your pre-amp input gain until you see a
strong signal on your audio interface Input meters (but not overloading the input).
One audio track for input calibration on Pro Tools
4
Record enable the audio track, or enable its
TrackInput monitoring button (Pro Tools|HD
only).
Chapter 98: Eleven Input Calibration and QuickStart
537
Set Up Eleven
Use Eleven’s Input LED to make final gain adjustments and complete the input calibration process.
Green (Off to –8)
Indicates signal is present, but too low.
Yellow (–8 to –4) Indicates the best level for low
output sources, such as single coil pickups.
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.
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.
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.
4
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.
Eleven’s Input LED (top) and Clip LED (bottom)
2
538
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:
Audio Plug-Ins Guide
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 544.
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:
1
Make sure you already calibrated your input
signal as explained in the previous sections of
this chapter.
2
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.
See “Processing Pre-Recorded Tracks
Through Eleven” on page 553 for more
information.
Librarian menu (left) and the Settings menu (right)
3
Pick any amp and cabinet from the available
types (see “Pairing Amps and Cabinets” on
page 548.)
4
Turn to Chapter 99, “Using Eleven” for specific
details on Eleven’s main controls, and for suggested track setups for recording, jamming, and
mixing.
Use the Settings menu to save, copy, paste,
and manage plug-in settings files. To save a
setting, see “Eleven Settings (Presets)” on
page 543.
Chapter 98: Eleven Input Calibration and QuickStart
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Audio Plug-Ins Guide
Chapter 99: 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 plugin 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
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 542.
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 popup menus and select an item.
When Pro Tools plug-ins are organized by
Category or Manufacturer, Eleven is listed as follows:
Category Harmonic
Manufacturer Digidesign
Chapter 99: Using Eleven
541
You can also click the Previous/Next arrows to
step through Amp, Cabinet, and Mic choices one at
a time.
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 542
About Unused Controls and Worksurfaces
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.
Enabling Switches
To enable or disable a switch or button, such as
Amp Bypass, click it to toggle its setting.
Using MIDI and MIDI Learn
with Eleven
Groups and Linked Plug-In Controls
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).
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.
542
Audio Plug-Ins Guide
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:
1
2
Make sure your external MIDI device is connected to your system, and recognized by your
MIDI Studio Setup (Windows) or Audio MIDI
Setup (Mac).
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 one 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.
• If you know the MIDI CC value of your foot
controller or other device, select it from the Assign menu.
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.
2
Click the Settings menu and choose Save Settings. Name the preset, choose a location, and
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.
To clear a MIDI assignment:

Right-click the control and choose Forget.
Chapter 99: Using Eleven
543
Output
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.
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
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.
Amp Type
Amp Type selects which amplifier model to use
(see “Amp Types” on page 545).
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 561.
Input
The Input knob provides input trim/boost, for tone
and distortion control. The Input range is –18 dB
to +18 dB.
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
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 561
544
Audio Plug-Ins Guide
Cab Type
This selector lets you select which speaker cabinet
model to use (see “Eleven Cabinet Types” on
page 548).
Gate
Noise Gate Threshold
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 545. For details on
where it derives its key (trigger) and applies its
gate, see “Eleven Signal Flow Notes” on
page 561.
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:
1
2
Connect and calibrate your guitar as explained
in “Connect your Guitar and Configure Source
Input” on page 536.
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.
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 *
3
4
To make it easier to hear the effect, begin by setting the Release to its middle (12 o’clock) position.
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).
• ’69 Plexiglas – 100W *
• ’82 Lead 800 – 100W *
• ’85 M-2 Lead *
• ’89 SL-100 Drive *
• ’89 SL-100 Crunch *
• ’89 SL-100 Clean *
5
6
7
Slowly lower the Threshold control until the
Gate opens again to find the cutoff (or, threshold) of the noise.
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.
If you lowered the Release setting as suggested
in step 3, make sure to return it to its maximum
setting (fully clockwise) before continuing.
• ’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 99: Using Eleven
545
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
Treble
Speed Depth
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 with each
546
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 preamp 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 preamp 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.
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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.
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:
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.

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 543).
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Audio Plug-Ins Guide
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
Mic Type
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.
The Mic Type selector lets you choose the microphone to use with the selected cabinet.
Speaker Breakup
(Full version, TDM Only)
Mic Type selector in the Cabinet section
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.
Available Mic Types include the following:
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.
• Condenser 87
• Dynamic 7
• Dynamic 57
• Dynamic 409
• Dynamic 421
• Condenser 67
• 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.
When enabled, Speaker Breakup draws
additional CPU resources.
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549
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 offcenter, in order to emphasize (or de-emphasize)
certain frequencies that the mic picks up.
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.
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.
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 550.
• “Recording Wet: Record Eleven-Processed
Track to Disk” on page 551.
• “Recording Dry and Eleven Simultaneously” on
page 552.
• “Processing Pre-Recorded Tracks Through
Eleven” on page 553
• “Blending Eleven Cabinets and Amps” on
page 554.
Recording Dry: Monitor Through
Eleven
About Mic Placement
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.
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.
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.
Mic Axis switch in the Cabinet section
To record dry and monitor through Eleven:
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Audio Plug-Ins Guide
1
Choose Track > New and configure the New
Track to create one mono Audio Track.
2
Set the track input to the audio interface input
your guitar is plugged in to (such as In 1
(Mono)).
3
Insert Eleven on the track (see “Inserting
Eleven on Tracks” on page 541).
Eleven
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.
Guitar input
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 a new track by doing the following:
• Create one mono Auxiliary Input track.
• Click the Add Row button (+).
• Create one mono audio track.
• Click Create.
3
Audio track for recording dry while hearing Eleven
4
Choose a Settings file (preset), or adjust
Eleven’s parameters to get your tone (see
“Eleven Settings (Presets)” on page 543).
5
Record enable the track, or enable TrackInput
monitoring (Pro Tools HD only) and check your
levels.
6
In the Mix (or Edit) window, configure the Aux
Input by doing the following:
• 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.
When you’re ready, arm the Pro Tools Transport and press Record to record your part.
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.
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4
Configure the audio track by doing the following:
• Click the Input selector and choose Bus 1.
• Record enable the audio track.
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:
Eleven
To record guitar dry and with Eleven live:
Guitar input
Bus output
Bus input
1
Choose Track > New.
2
Configure the New Tracks dialog to create two
mono audio tracks, then click Create.
3
In the Mix (or Edit) window, configure the first
(left-most) new audio track by doing the following:
• 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.
Aux Input
Audio Track
Recording Eleven (printing its output)
5
6
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.
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.
• Click the Insert selector and select Eleven.
• Record enable the audio track.
4
Configure the second audio track by doing the
following:
• Click the Input selector and choose Bus 1.
• Record enable the audio track.
5
Make sure you are not overloading your input
signal by checking levels in all tracks and
Eleven's Input LED.
6
When you’re ready, arm Pro Tools and begin recording.
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):
1
Import and place your audio in a Pro Tools audio track.
To process and re-record tracks through Eleven:
1
Import and place your audio in a Pro Tools
audio track.
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).
• Click the Insert selector and select Eleven.
2
Assign the audio track Output to Bus 1 (or
Bus 1-2 if working with stereo material).
3
Choose Track > New and create one mono
audio track.
3
Create an Aux Input track, and configure it by
doing the following:
4
Configure the new audio track by doing the following:
• 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).
• Click the Insert selector and select Eleven.
• Click the Insert selector and select Eleven.
4
Begin playback and watch Eleven’s Input LED
to check your level. Make sure you’re not clipping Eleven’s input.
5
While listening, adjust Eleven’s Input knob to
increase or decrease input level.
6
After setting your gain structure, do any of the
following:
• Try different Settings files (presets) to get your
basic amp/cab/mic combination.
5
Record enable the new audio track (or enable
TrackInput monitoring if using Pro Tools HD).
6
Begin playback and start listening.
7
While listening, adjust Eleven’s Input knob to
increase or decrease input level.
8
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.
7
Apply other plug-ins, or bus the Aux Input to
another track for additional processing.
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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
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, multi-mic one
cabinet, or both.
Three tracks, selected
5
• Choose Bus 1 from the Input selector of any of
the three selected Aux Inputs.
• 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.
6
Open the Eleven plug-in on the audio track and
click the Cabinet Bypass to bypass Cabinet and
microphone processing.
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.
To blend multiple cabinets:
1
Choose Tracks > New.
2
Configure a new track by doing the following:
• Create one mono Audio Track.
• Click the Add Row button.
• Create three mono Aux Inputs.
• Click Create.
3
In the Mix or Edit window, configure the audio
track by doing the following:
• 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.
4
554
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.
Audio Plug-Ins Guide
Hold Option+Shift (Mac) or Alt+Shift
(Windows) while doing each of the following:
13
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.
14
Do the following to continue:
Amps bypassed/Cabs on
Amp on,
Cab bypassed
• 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 557).
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
9
Click to open the Eleven plug-in window on the
first Aux Input, and do any of the following:
• Choose a cabinet.
• Choose a mic and its position.
• Adjust Speaker Breakup as desired.
10
When you’re done, close the plug-in window
and then unsolo the track.
11
Solo the next Aux Input track, and repeat to
configure its cabinet and mic settings.
12
Repeat for other Aux Input tracks to configure
their cabinet and mic settings.
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 557 for more information.
Chapter 99: Using Eleven
555
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 plugins (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 multiamp setups.
To blend multiple amps:
1
Set up tracks and signal routing as explained in
the previous workflow (see “To blend multiple
cabinets:” on page 554).
2
Remove (or simply bypass) the Eleven plug-in
on the source input/track.
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.
Setup for blending amps
3
Solo the first Aux Input track.
4
Click to open the Eleven plug-in window on the
soloed Aux Input, and do any of the following:
• 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.
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Audio Plug-Ins Guide
5
Solo the next Aux Input track, and repeat to
configure its settings for a different tone.
6
Repeat for other Aux Input tracks to configure
their settings.
7
When you have set your tones, make sure to unsolo all the Aux Inputs.
8
Continue playing so you can hear the combined
tone of all the amps.
9
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.
10
Evaluate the phase relationships of the combined signals and adjust accordingly (see
“Phase Considerations with Blending in
Eleven” on page 557).
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.
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 554. Make sure each Aux Input
has an Eleven plug-in followed by a TimeAdjuster plug-in.
2
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).
3
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).
4
Click the Phase switch on the next channel’s
TimeAdjuster plug-in, listen, then disengage.
5
Repeat for additional Eleven/TimeAdjuster
channels.
6
Try combinations of flipped and non-flipped
Phase settings to find the ideal relationship for
the currently blended amps and cabinets.
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
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557
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
2
Configure your audio track and Aux Inputs as
instructed in “Blending Eleven Cabinets and
Amps” on page 554. 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).
3
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.
4
Try combinations of TimeAdjuster settings with
flipped and non-flipped Phase settings for endlessly variable tonal possibilities.
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 556 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 551.
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.
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559
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 Avid’s classic compressor 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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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:
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 Input knob in the Master section, which affects the signal level before entering the amplifier
model.
 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.
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Part XV: Synchronic
Chapter 100: 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.
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Chapter 101: 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.
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567
Playing Synchronic RTAS
See “Using Synchronic as an AudioSuite
Plug-In” on page 605 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 plug-ins.
Instead, Synchronic works as follows:
1
Load audio files (loops) into Synchronic’s
Sound module, much like you would add sound
to a sampler.
2
Use the Detection Slider to slice up the loops
into rhythmically logical units (beats and subdivisions of the beat).
3
Play back the sliced-up loop in tempo.
Loading a Loop in Synchronic
For detailed information on loading audio
files (loops) into Synchronic see “Importing
a Sound into Synchronic” on page 578.
To load a loop in Synchronic:
1
Insert Synchronic on an Instrument track.
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 571).
Edit/Performance Import
Mode toggle
button
Current
preset
Choose an audio file (loop) for import
4
Sound module, Edit mode
3
To import an audio file (mono, dual mono, or
stereo) into the Waveform display, do one of the
following:
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.
• Drag an audio file from the Workspace to the
Waveform display.
• 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
non-contiguous files.
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569
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 581.
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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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 102,
“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 104, “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 605 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 102,
“Using Synchronic”
Synchronic Performance
Controls
(RTAS Only)
Synchronic provide several types of controls for
real-time performance, including presets, user-assignable 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 572.
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 102, “Using Synchronic.”
Chapter 101: Synchronic Overview
571
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 600.
Keyboard, MIDI module
Synchronic Presets
User Knobs, Playback module
Sound A and B Selectors
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.
Each module has 12 presets available, as follows:
• 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.
Sound A/B selectors, Playback module
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 105,
“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 600.
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 603).
To edit and store a preset:
1
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 576).
2
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 102, “Using Synchronic”).
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.
2
Control-click (Windows) or Command-click
(Mac) the destination preset.
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.

In Edit mode, click a Preset button in the Sound
module, or select a preset from the Preset popup menu for the Playback, Effects, and XFade
modules. In Edit mode, the MIDI module’s presets are not available on-screen.
The settings from the selected preset will be
copied to the destination preset.
Preset pop-up menu, Playback module
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Audio Plug-Ins Guide
Chapter 102: Using Synchronic
This section covers Synchronic controls and
functions.
To type a parameter value:
1
Click on the parameter text that you want to
edit.
Adjusting Synchronic
Parameters
2
Change the value by doing one of the following:
You can adjust Synchronic parameters with a
mouse or a computer 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.
• Type the desired value.
• To increase a value, press the Up Arrow on your
keyboard. To decrease a value, press the Down
Arrow on your keyboard.
3
Do one of the following:
• Press Enter on the numeric keyboard to input the
value and remain in keyboard editing mode.
• 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.
Chapter 102: Using Synchronic
575
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 576) 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 577), 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
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.”
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 nonselected 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.
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 105, “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).
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 Controlclicking (Mac) the Waveform display re-synchronizes playback with the current Pro Tools MIDI
Beat Clock location (according to bars and beats).
Starting and Stopping Playback
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.
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577
Performance/Edit
Mode toggle
Selected
preset
Sound
presets
Interactive Waveform
display (stereo file
loaded and sliced)
Sound Attributes
Slice Detection slider
Import Sound Delete Sound
button
button
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 571).
2
Select the preset where you want to store the audio file (loop).
3
Click the Import button. If there is already a file
loaded into the current preset, you will be
prompted to delete it.
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
non-contiguous files.
5
4
Drag the selected audio to the Waveform display in Synchronic.
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:
1
Make sure the Sound module is set to Edit
mode. (see “Performance and Edit Modes” on
page 571).
2
Select the preset where you want to store the audio file (loop).
3
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.
Waveform display, stereo file, before slicing
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579
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 581).
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 audio
loop contains eighth, sixteenth, or thirty-second
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:
1
Make sure the Sound module is set to Edit
mode. (see “Performance and Edit Modes” on
page 571).
2
Select the preset you want to clear.
3
Click the Delete button.
4
You will be prompted to confirm deletion.
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 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.
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
Chapter 102: Using Synchronic
581
Generate Missing Adds additional slices where
transients appear to be missing. This can be useful
when an imported audio loop contains few transients.
Use Generate Missing to create rhythmically
logical slices on ambient loops or drones, and
apply playback manipulations and
effects for interesting results.
Current Displays the currently selected Slice for
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.
In Keyboard Focus mode, use the Left and
Right arrow keys to increment/decrement the
current slice number.
Audition Modes Are initiated using the Pro Tools
Delete Deletes the currently selected Slice. Op-
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:
tion-click can also be used to delete erroneous
slices.
• 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.
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.
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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.
• Type 2: Sounds most natural for most loops and
is the default setting.
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 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 583) and Playback Edit mode (see
“Synchronic Playback Edit Mode” on page 584)
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
User Knobs
(PB1 and PB2)
Playback module, Performance mode
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583
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
Preset selector
Performance/Edit
Mode toggle
User Knob
assignments
Playback module, Edit mode
Playback User Knobs (PB1 and PB2)
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.
Synchronic Playback Modes
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.
Selecting Playback Mode
Standard Plays each slice without any additional
manipulation and plays each slice from start to
end.
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Stretch Uses a granular time compression/expan-
sion 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 590).
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 589).
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.
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 590).
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
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 583).
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585
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.
Straddle (Straddle 2–5) Plays slices by stepping
Selecting Playback Order
Off Plays the sound slices in their original sequence. 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 re-
verse order. Each slice plays in the specified direction (see “Synchronic Playback Direction” on
page 588), but order of slices is reversed.
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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 gen-
erally emphasize the “and” of the beat.
Syncopate (Syncopate 1–5) Five variations fea-
turing dotted rhythms.
Selecting Playback Duration
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587
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 sixteenthnote 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
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Forward Plays all slices forward (from beginning
Enable (On, Off) Enables or disables Pitch modu-
to end).
lation.
Backward Plays all slices backward (from end to
beginning).
BEG and END Range (–60 to +60) Sets the Begin-
For/Back Plays alternating slices forward then
ning transposition and End transposition values (in
semitones) referenced by both LFOs.
backward.
Low Frequency Oscillator (LFO)
Back/For Plays alternating slices backward then
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.
forward.
F/B Diddle Plays consecutive slices forward, back-
ward, 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.
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 Waveshape
selectors
LFO Duration
selectors
Playback Pitch settings
LFO Waveshape pop-up menu
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589
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 587.
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 modu-
lation.
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 wave-
shape 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 591.
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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591
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 592) mode and Effect Edit mode (see “Synchronic Effect Edit Mode” on page 593).
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Audio Plug-Ins Guide
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 Waveshape
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 591.
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593
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 Waveshape
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.
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.
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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Audio Plug-Ins Guide
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 591.
Noise Mode
Synchronic Filter Effect
There are three different Noise modes: Dark,
White, and Brite; each with two variations: a noise
generator (Osc) or amplitude modulation (AM).
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 Waveshape
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 591.
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595
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 Waveshape
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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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 591.
Delay Mode
To assign a User Knob to control Edit parameters:
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).

Selecting Delay Mode
Assigning Synchronic Effect
User Knobs (FX1 and FX2)
(RTAS Only)
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.
Select the desired Edit parameter assignment
from the User Knob Assignment pop-up menu.
Selecting a User Knob assignment (FX2 assigned to
Delay 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.
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597
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
Performance/Edit
Mode toggle
Manual/Preset
Mode toggle
Presets
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:
1
If necessary, click the Performance/Edit Mode
toggle to switch to Performance mode.
2
Click the Manual/Preset Mode toggle to switch
between Manual and Preset mode.
In Edit mode, the XFade module lets you edit
crossfade modulation effects and the XFade User
Knob assignment.
Preset selector
Performance/Edit
Mode toggle
Manual mode takes precedence over any
XFade preset automation. When switching
out of Manual mode, any XFade automation
previously received will take effect.
Beginning and End
XFade amount (%)
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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599
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 dynamic
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
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 591.
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 600) and MIDI Edit mode (see “Synchronic
MIDI Edit Mode” on page 602).
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 onscreen 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.
To capture and assign a combination to a MIDI
note number:
1
Edit Synchronic's Sound, Playback, Effect, and
XFade parameters and select presets as desired.
2
If necessary, switch the MIDI module to Performance mode (see “Performance and Edit
Modes” on page 571).
3
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 602).
4
Enable or disable the Module Assign Enable
buttons as desired (see “Selective Module Assignment” on page 602).
5
Click the MIDI key on the on-screen keyboard
(or press the corresponding key on your MIDI
keyboard).
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 601).
• 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 onscreen keyboard or with a MIDI controller.
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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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
Enable buttons
(enabled)
Module Assign
Enable button
(disabled)
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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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 105, “Synchronic Plug-In Settings”).
To assign an external MIDI controller:
1
2
If necessary, switch the MIDI module to Edit
mode (see “Performance and Edit Modes” on
page 571).
Select Pitch Wheel or Controller # from the
Source pop-up menu.
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 controller
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 Cross-
fade presets 1–10.
MIDI Controller # field assignment for PB2 User Knob
(set to MIDI controller #1, the modulation wheel)
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603
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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Audio Plug-Ins Guide
Chapter 103: 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.
Chapter 103: Using Synchronic as an AudioSuite Plug-In
605
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 button
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 607 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 577 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 584 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 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 593 for detailed information on Playback module controls in Edit mode.
Synchronic AudioSuite
Workflow
Use the AudioSuite version of Synchronic to work
with and play back audio loops as follows:
1
Load audio files or a selected portion of audio
clips (loops) into the Sound module, much like
you would add sound to a sampler.
2
Use the Detection Slider to slice up the loops
into rhythmically logical units (beats and subdivisions of the beat).
3
Preview the sliced-up loop.
4
When you are finished with a loop, you render it
to a track.
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 578 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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607
To load a selection from Pro Tools into
Synchronic:
1
Choose AudioSuite > Synchronic.
2
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.
See “Slicing Up a Sound in Synchronic” on
page 581 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
3
In the Sound module of Synchronic AudioSuite,
click Load Selection.
• 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:
1
In the Pro Tools Edit window, select the portion
of an audio track where you want to place the
rendered audio.
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.
2
Click Render to print the audio loop to the
selection.
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Audio Plug-Ins Guide
Chapter 104: 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
plug-in parameter in the Plug-In window, then
choose Open Automation dialog from the popup menu.
• Using Pro Tools automation playlists
• Using MIDI
Using Automation Playlists
Pro Tools creates a separate playlist for each plugin 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:
1
Open the Plug-In window for the plug-in you
want to automate.
2
Do one of the following:
• Click the Automation Enable button in the PlugIn window.
• Control-Start-Alt-click (Windows) or Command-Option-Control-click (Mac) the Track
View Selector in the Edit window.
Selecting parameters to automate
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611
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 Plug-In
window by Control-Alt-Start-clicking (Windows) or Command-Control-Option-clicking
(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:
1
In the Automation Enable window, make sure
that plug-in automation is write-enabled.
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 613.
612
3
On the track with Synchronic inserted, choose
an automation mode. For an initial pass, choose
Auto Write.
4
Click Play to begin writing automation, and
move the controls you want to automate.
5
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 600.
Assigning MIDI Notes and
Controller Data to Presets
5
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:
1
If necessary, switch the MIDI module to Edit
mode (see “Performance and Edit Modes” on
page 571).
2
Select Pitch Wheel or Controller # from the
Source pop-up menu for the desired User Knob
assignment.
To assign MIDI notes to combinations of
Synchronic presets:
1
If necessary, switch the MIDI module to Performance mode (see “Performance and Edit
Modes” on page 571).
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 602).
Edit Synchronic’s Sound, Playback, Effect, and
XFade parameters and select presets as desired.
Selecting MIDI control source
3
If you select Controll