Audio Plug-Ins Guide

Audio and MIDI
Plug-ins Guide
Legal Notices
© 2024 Avid Technology, Inc., (“Avid”), all rights reserved. This guide may not be duplicated in whole or in part without the written consent of Avid.
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Avid Pro Tools Audio and MIDI Plug-ins Guide • Created 5/28/2024 • Guide Part Number 9329-66559-00 REV A 06/24
Contents
Contents iii
1 Introduction to Audio Plug-Ins 17
Plug-In Formats 17
Avid Audio Plug-Ins 17
Using Plug-Ins in Pro Tools 20
System Requirements and Compatibility for Plug-Ins 20
Conventions Used in This Guide 21
Resources 21
Avid Link 22
Installing and Authorizing Avid Plug-Ins 23
Installing Plug-ins Bundled with Pro Tools Subscriptions 23
Installing Plug-Ins from the Avid Marketplace 23
Installing Plug-Ins for Pro Tools and Media Composer 23
About iLok 24
Authorizing Avid Audio Plug-Ins 24
Removing Plug-Ins 25
Adjusting Plug-In Controls 25
Dragging Plug-In Controls 25
Editing Control Values 27
Dragging in Graphic Displays 27
Toggle Controls 28
Adjusting Controls with Fine Resolution 28
Resetting Controls to Default Values 28
General Effects Controls 28
MIDI Control 29
2 EQ Plug-Ins 31
304E Equalizer 31
304E Equalizer Controls 31
EQ III 32
iii
EQ III Configurations 32
Adjusting EQ III Controls 33
EQ III I/O Controls 35
EQ III EQ Band Controls 36
1-Band EQ III 37
7-Band EQ III 41
Eleven Effects Graphic EQ 49
Graphic EQ 49
Focusrite D2 50
D2 Configurations 50
D2 Controls 51
Using D2 in Stereo 53
Pultec Plug-Ins 55
Pultec EQP-1A 55
Pultec EQH-2 56
Pultec MEQ-5 56
Pultec Tips and Tricks 57
3 Dynamics Plug-Ins 58
304C Compressor 58
304C Compressor Controls 59
304C Compressor Tips and Tricks 60
BF-2A 61
BF-2A Controls 61
Using the BF-2A Side-Chain Filter 62
BF-2A Tips and Tricks 63
BF-3A 63
BF-3A Controls 64
BF-3A Tips and Tricks 64
BF76 64
BF76 Controls 65
BF76 Tips and Tricks 66
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Channel Strip 67
Channel Strip Sections and Panes 67
Channel Strip Input Section 69
Channel Strip Output Section 71
Channel Strip FX Chain 72
Channel Strip Dynamics Section 72
Channel Strip EQ / Filters Section 79
Dynamics III 84
Dynamics III Common Controls 84
Compressor / Limiter III 87
Expander / Gate III 91
Expander / Gate III Threshold Control 92
De-Esser III 93
Dynamics III Side-Chain Input 96
Eleven Effects Gray Compressor 102
Gray Compressor 102
Fairchild Plug-Ins 103
Fairchild 660 103
Fairchild 670 104
Focusrite D3 106
D3 Compressor 106
D3 Limiter 106
D3 Side-Chain Processing 107
Using D3 in Stereo 107
D3 Common Controls 107
D3 Compressor Controls 109
D3 Limiter Controls 110
Using the Side-Chain Input in D3 111
Impact 112
Impact Controls 112
Using the Impact Compressor 113
Maxim 115
v
About Peak Limiting 115
How Maxim Differs From Conventional Limiters 116
Maxim Controls and Meters 116
Using Maxim 118
Maxim and Mastering 119
Pro Compressor 120
Pro Compressor Metering 120
Pro Compressor Input Section 121
Pro Compressor Output Section 121
Pro Compressor Dynamics Graph 122
Pro Compressor Controls 126
Pro Compressor Side-Chain Processing 129
Source 129
Side-Chain Listen Mode 130
Side-Chain Filter On/Off 131
Side-Chain Filter 131
Pro Expander 132
Pro Expander Metering 132
Pro Expander Input Section 133
Pro Expander Output Section 134
Pro Expander Dynamics Graph 135
Pro Expander Controls 138
Pro Expander Side-Chain Processing 139
Pro Limiter 143
Pro Limiter Metering 143
Pro Limiter Input Section 144
Pro Limiter Output Section 145
Pro Limiter Controls 146
Pro Limiter Loudness Numeric Displays 148
Pro Limiter Histogram and Loudness Meters 149
AudioSuite Processing with Pro Limiter 150
AudioSuite Processing with Pro Limiter Loudness Analyzer 150
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Pro Multiband Dynamics 152
FFT Display and Controls 152
Input and Output Gain Controls 155
Source Linking 156
Mini Multichannel Gain Reduction and Output Meters 156
Band Pane Controls and Indicators 157
Side-Chain Processing 161
Multiband Splitter Plug-In 162
Pro Multiband Dynamics and Multiband Splitter Plug-In Sends 163
Purple Audio MC77 164
Purple Audio MC77 Controls 164
Smack! 165
Smack! Controls and Meters 165
Using the Smack! Side-Chain Input 170
4 Pitch and Time Shift Plug-Ins 171

Pitch II 171
Pitch II Controls 172
Pitch Shift Legacy 174
Pitch Shift Legacy Controls 175
Time Shift 176
Time Shift Controls 177
AudioSuite Input Modes and Time Shift 182
AudioSuite Preview and Time Shift 182
Time Shift as AudioSuite TCE Plug-In Preference 182
Processing Audio Using Time Shift 182
Post Production Pull Up and Pull Down Tasks with Time Shift 184
Vari-Fi 184
Vari-Fi Controls 184
X-Form 185
X-Form Displays and Controls Overview 186
X-Form AudioSuite Input Modes 191
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AudioSuite TCE Plug-In Preference 191
Processing Audio Using X-Form 191
Using X-Form for Post Production Pull Up and Pull Down Tasks 192
5 Reverb Plug-Ins 193

D-Verb 193
D-Verb Controls 193
Selections for D-Verb AudioSuite Processing 195
Eleven Effects Reverb Plug-ins 195
Black Spring 195
Studio Reverb 196
Reverb One 196
About Reverb 197
Reverb One Controls 198
Reverb One Graphs 202
Other Reverb One Controls 204
ReVibe II 205
Using ReVibe II 205
Dragging in the Graphic Display to Adjust Controls 206
ReVibe II Input and Output Meters 207
ReVibe II Controls 207
ReVibe II Decay EQ Graph 213
ReVibe II Decay Color Graph 214
ReVibe II Contour Display 215
ReVibe II Room Types 215
Space 221
Space Feature Highlights 222
Space Overview 223
Impulse Response (IR) and Space 227
Space Presets 230
Space Snapshots 231
Space Controls and Displays 231
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Space Display Area 232
Space IR Browser 236
Space Primary Controls 239
Space Group Selectors and Controls 239
Using Space 244
Space IR Library Categories 245
6 Delay Plug-Ins 246

Mod Delay III 246
Mod Delay III Controls 246
Selections for Mod Delay III AudioSuite Processing 249
Moogerfooger Analog Delay 249
Moogerfooger Analog Delay Controls 250
Moogerfooger Analog Delay Tips and Tricks 251
Reel Tape Delay 251
Reel Tape Common Controls 252
Reel Tape Delay Controls 252
Synchronizing Reel Tape Delay to Session Tempo 254
Reel Tape Delay Presets 254
Tel-Ray Variable Delay 255
Tel-Ray Controls 256
Tel-Ray Tips and Tricks 257
TimeAdjuster 257
TimeAdjuster Controls 257
Using TimeAdjuster for Manual Delay Compensation 258
When to Compensate for Delays 259
7 Modulation Plug-Ins 260

Eleven Effects Modulation Plug-Ins 260
Black Shiny Wah 260
C1 Chorus/Vibrato 261
Flanger 262
Orange Phaser 263
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Roto Speaker 264
Vibe Phaser 264
Moogerfooger Lowpass Filter 265
Moogerfooger Lowpass Filter Controls 267
Moogerfooger Lowpass Filter Tips and Tricks 268
Moogerfooger 12-Stage Phaser 268
Moogerfooger 12-Stage Phaser Controls 269
Moogerfooger 12-Stage Phaser Tips and Tricks 271
Moogerfooger Ring Modulator 271
Moogerfooger Ring Modulator Controls 272
Moogerfooger Ring Modulator Tips and Tricks 273
Reel Tape Flanger 273
Reel Tape Flanger Controls 274
Synchronizing Reel Tape Flanger to Session Tempo 276
Reel Tape Flanger Tips 277
Reel Tape Flanger Presets 277
Sci-Fi 278
Sci-Fi Controls 278
Voce Plug-Ins 280
Voce Chorus/Vibrato 280
Voce Spin 281
8 Harmonic Plug-Ins 286

Aphex Aural Exciter Type III 286
Meters 287
Rotary Controls 288
Using Aural Exciter III 291
Aphex Big Bottom Pro 293
Meters 294
Rotary Controls 294
Switches 295
x
Using Big Bottom Pro 296
Eleven 296
Eleven Input Calibration and QuickStart 298
Using Eleven 302
Eleven Tips and Suggestions 321
Eleven Signal Flow Notes 322
Eleven MK II 323
Eleven MK II Plug-In Features 323
Eleven MK II Cab Plug-In 324
Eleven MK II Input Calibration and QuickStart 324
Using Eleven MK II 329
Eleven MK II Tips and Suggestions 347
Eleven MK II Signal Flow Notes 348
Eleven Effects Harmonic Plug-ins 349
Black Op Distortion 349
DC Distortion 350
Green JRC Overdrive 350
Tri-Knob Fuzz 351
White Boost 352
Lo-Fi 352
Lo-Fi Controls 353
Pro Subharmonic 354
Metering 355
Input 356
Output 356
Dynamic Frequency Display 357
High Pass and Low Pass Filter Controls 358
Subharmonic Frequency Range 359
Lower, Upper, and Direct Gain Controls 360
Drive 361
Mix Controls 361
Surround Send Controls 363
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Tuning Subharmonics with MIDI 363
Recti-Fi 365
Recti-Fi Controls 365
Reel Tape Saturation 367
Reel Tape Saturation Controls 369
Reel Tape Saturation Tips 369
Reel Tape Saturation Presets 370
SansAmp PSA-1 370
PSA-1 Controls 371
PSA-1 Tips and Tricks 371
9 Dither Plug-Ins 372

Dither 372
Dither Controls 372
POW-r Dither 373
POW-r Dither Controls 373
10 Sound Field Plug-Ins 375

AutoPan 375
AutoPan Controls 375
Using AutoPan 382
Down Mixer 383
Source 385
Downmix 385
11 Instrument Plug-Ins 386

Click II 386
Click II Controls and Displays 386
ReWire Requirements 387
ReWire 388
12 MIDI Plug-ins 390

Avid Note Stack 391
Notes 1–9 391
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Avid Pitch Control 392
Range 392
Control 393
Avid Velocity Control 393
Range 394
Control 394
Audiomodern Riffer 395
QuickStart 395
Tracks Setup 396
More to Explore 398
Top Bar Controls 399
Info Tab 399
Piano Roll 399
Keyboard Mode 399
Tempo 399
Undo 399
Panic 399
Learn 399
Presets Tab 399
Settings Tab 399
Resize Window 399
Piano Roll 400
Keyboard Mode 400
Randomization 400
Reverse, Lock, Erase 401
Playback Motion 401
Infinity Mode 401
Lock Notes and Lock Steps 402
Lock Notes 402
Lock Steps 402
Reset 402
Piano Roll Controls 402
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Number of Steps 402
Quantization 403
Shuffle 403
Steps and Loop 403
Root Note 403
Tie Notes 403
Sustain Notes 403
C – B Keys 403
Riff Start and End 404
Octave Buttons 404
MIDI Export 404
Scales 404
Keyboard Button 405
Randomizing the Pitch 405
Pitch Tab 405
Pitch Range 405
Set the Same Pitch Value to All Steps 406
Randomizing the Duration 406
Duration Tab 406
Duration Range 406
Randomizing the Velocity 406
Velocity Tab 406
Velocity Range 406
Randomizing the Density 406
Density Tab 407
Density Range 407
Automation 407
MIDI Learn 407
MIDI CC Mappings 407
Clear Mappings 407
Saving Presets 407
Quick-Load Slots 408
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Presets Tab 408
Settings 410
Mixed in Key MIDI Plug-ins 411
Modalics EONarp 412
MIDI Signal Flow 413
Top Toolbar 413
Pattern Snapshots 413
Editing Toolbar 413
Q (Quantize Notes) 413
Snap 413
Pattern/Step Length 413
ARP 414
Brush Module 414
Undo/Redo 414
Clear 414
Arp Grid 414
Chords 415
Input 415
Retrigger 415
Q Value 415
MIDI File Player 415
Transpose 415
Octave 415
Pitch Innovations Groove Shaper 416
Header Controls 416
Shape Blocks 417
Sequencer Settings 419
Shape Settings 420
Magic Dice 421
Bottom Toolbar 421
Settings 422
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13 Other Plug-Ins 424
InTune 424
InTune Controls and Displays 425
InTune Presets 428
Using InTune 429
MasterMeter 430
MasterMeter Overview 430
Using MasterMeter 435
MasterMeter Controls and Displays 435
Signal Generator 438
Signal Generator Controls 438
AudioSuite Processing with Signal Generator 439
SoundReplacer 439
Audio Replacement Techniques 440
SoundReplacer Controls 440
Using SoundReplacer 443
Getting Optimum Results with SoundReplacer 444
Using the Audio Files Folder for Frequently Used SoundReplacer Files 446
Time Compression / Expansion 447
Time Compression/ Expansion Controls 447
Trim 448
Trim Controls 449
14 Other AudioSuite Plug-In Utilities 450

DC Offset Removal 450
Duplicate 450
Gain 451
Invert 452
Normalize 452
Reverse 453
Index 456
xvi
1 Introduction to Audio Plug-Ins
Plug-ins are special-purpose software components that provide additional signal processing and other
functionality to Pro Tools® software. These include plug-ins that come with your Pro Tools system, as well as
many other plug-ins that can be purchased or rented from Avid® separately. This guide documents all 64-
bit AAX audio plug-ins available from Avid for Pro Tools, VENUE®, and Media Composer®.
Additional plug-ins are available from third-party developers. For more information, visit
www.avid.com/plugins.
Plug-In Formats
AAX (Avid Audio Extension) plug-ins provide real-time plug-in processing using host-based (“Native”) or
DSP-based (Pro Tools | Carbon and HDX systems only) processing. The AAX plug-in format also supports
AudioSuite non-real-time, file-based rendered processing. AAX plug-in files use the “.aaxplugin” file suffix.
There are three plug-in formats used in Pro Tools:

l AudioSuite™: non-real-time, file-based processing
l AAX Native: real-time, host-based plug-ins
l AAX DSP: real-time, DSP-based plug-ins (Pro Tools | Carbon and HDX systems only)
Avid Audio Plug-Ins

Avid includes a comprehensive set of sound processing, effects, and utility plug-ins with all Pro Tools
systems. Other Avid plug-ins are available for purchase or rental from the Avid store (visit shop.avid.com) or
the Avid Marketplace (in Avid Link, choose Marketplace or in Pro Tools, choose Avid Link >
Marketplace).
Avid Audio Plug-Ins Included with Pro Tools

Pro Tools includes a suite of digital signal processing effects, including EQ, dynamics, delay, and other
essential audio processing tools. The following plug-ins are included with Pro Tools:
EQ
l Channel Strip (see “Dynamics”)
l EQ III
l 1 Band
l 7 Band
Dynamics
l BF76 Compressor
l Channel Strip
l Dynamics III
l Compressor / Limiter
l Expander / Gate
l De-Esser
l Maxim™
17
Pitch and Time Shift
l Pitch II
l Pitch Shift Legacy
l Time Shift
l Vari-Fi™
Reverb
l D-Verb
Delay
l Mod Delay III
l TimeAdjuster
Modulation
l Sci-Fi™
Harmonic
l Eleven® Lite
l Lo-Fi™
l Recti-Fi™
l SansAmp PSA-1
Dither
l Dither
l POW-r Dither
Sound Field
l AutoPan™
l Down Mixer
Instrument
l Click II
l ReWire
Other
l DC Offset Removal (AudioSuite only)
l Duplicate (AudioSuite only)
l Gain (AudioSuite only)
l Invert (AudioSuite only)
l Normalize (AudioSuite only)
l Reverse (AudioSuite only)
l Signal Generator
l Time Compression/Expansion
18
l InTune™
l MasterMeter™
l Trim
Additional Avid Audio Plug-Ins
The following plug-ins are available separately for purchase and rental:
l 304E equalizer
l 304C compressor
l Aphex Aural Exciter® Type III
l Aphex Big Bottom Pro®
l BF-2A
l BF-3A
l Eleven® guitar amplifier modeling plug-in
l Eleven® Mk II guitar amplifier modeling plug-in
l Eleven Effects
l Fairchild 660 and 670
l Focusrite d2/d3
l Impact®
l Moogerfooger plug-ins
l Moogerfooger Analog Delay
l Moogerfooger Ring Modulator
l Moogerfooger 12-Stage Phaser
l Moogerfooger Lowpass Filter
l Pro Compressor
l Pro Expander
l Pro Limiter
l Pro Multiband Dynamics
l Pro Subharmonic
l Purple Audio MC77
l Reel Tape™ plug-ins:
l Reel Tape Saturation
l Reel Tape Delay
l Reel Tape Flanger
l Reverb One™
l ReVibe® II
l Smack!™
l SoundReplacer™
l Space™
l Tel-Ray Variable Delay
19
l Voce Spin
l Voce Chorus/Vibrato
l X-Form®
MIDI Plug-Ins in Pro Tools
Pro Tools includes a set of MIDI plug-ins from Avid and select third-party developers. The following plug-ins
are included and installed with Pro Tools:
Avid MIDI Plug-ins

l Note Stack
l Pitch Control
l Velocity Control
Third-party MIDI Plug-ins
l Audiomodern Riffer 3
l Mixed in Key Human Lite and Captain Chords Lite
l Modalics EON-Arp
l Pitch Innovations Groove Shaper Lite
Using Plug-Ins in Pro Tools

See Pro Tools online Help (Help > Pro Tools Help) or the Pro Tools Reference Guide (Help >
Pro Tools Reference Guide) for general information on working with plug-ins in Pro Tools, including:
l Inserting plug-ins on tracks
l Clip indicators
l Plug-in Window controls
l Adjusting plug-in controls
l Automating plug-ins
l Using side-chain inputs
l Working with instrument plug-ins
l Using plug-in presets
System Requirements and Compatibility for Plug-Ins

To use Pro Tools plug-ins, you need the following:
l An Avid-qualified system running Pro Tools software.
l An iLok USB key (iLok) for plug-ins that can be purchased or rented that do not support computer-
based authorization or iLok Cloud.
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 systems, refer to the documentation that came with
your plug-in.
20
Conventions Used in This Guide
Pro Tools documentation uses the following conventions to indicate menu choices, keyboard commands,
and mouse commands:
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:
g User Tips are helpful hints for getting the most from your Pro Tools system.
c Important Notices include information that could affect your Pro Tools project data or the
performance of your Pro Tools system.
b Shortcuts show you useful keyboard or mouse shortcuts.

n Cross References point to related sections in this guide and other Avid documentation.
How to Use this PDF Guide
This PDF provides the following useful features:
l The Bookmarks on the left serve as a continuously visible table of contents. Click on a subject
heading to jump to that page.
l Click a + symbol to expand that heading to show subheadings. Click the – symbol to collapse a
subheading.
l The Table of Contents provides active links to their pages. Select the hand cursor, allow it to hover
over the heading until it turns into a finger. Then click to locate to that subject and page.
l All cross references in blue are active links. Click to follow the reference.
l Select Find from the Edit menu to search for a subject.
l When viewing this PDF on an iPad, it is recommended that you open the file using Apple Books to take
advantage of active links within the document. When viewing the PDF in Safari, touch the screen,
then touch Open in “Books”.
Resources
The Avid website (www.avid.com) is your best online source for information to help you get the most out of
your Avid system.
Account Activation and Product Registration

Activate your product to access downloads in your Avid account (or quickly create an account if you do not
have one). Register your purchase online, download software, updates, documentation, and other
resources.
t www.avid.com/account
21
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 Avid user community on the User Conference.
t www.avid.com/support
Training and Education
Study on your own using courses available online, find out how you can learn in a classroom setting at an
Avid-certified training center, or view video tutorials and webinars.
t www.avid.com/learning
Videos and Tutorials
Visit the Avid YouTube channel to find playlists and videos that show how to use and learn Pro Tools.
t Avid YouTube Channel (all playlists and videos)
t Pro Tools Tech Tips (playlist)
t Pro Tools Quick Tips (playlist for the Pro Tools Quick Reference Guide, available from the Dashboard)
Products and Developers
Learn about Avid products, download demo software, or learn about our Development Partners and their
plug-ins, applications, and hardware.
t www.avid.com/products
Avid Link
Use Avid Link™ to manage your Avid account, licensing and subscriptions, as well as access a number of
other services. Avid Link will help you keep all of your plug-ins up to date. Login with your Avid Master
Account username and password. For more information, visit www.avid.com/avid-link.
22
Installing and Authorizing Avid Plug-Ins
A core set of audio plug-ins is installed automatically with your version of Pro Tools. No additional steps are
required to authorize these plug-ins for use on your Pro Tools system. You may also be entitled to additional
non-core plug-ins included with your Pro Tools subscription. For any such additional plug-ins, use the Avid
Link application to install and authorize these plug-ins (for more information, visit www.avid.com/avid-link).
Installers for additional plug-ins can be purchased or rented from the Avid store (visit shop.avid.com) or the
Avid Marketplace (in Avid Link, choose Marketplace or in Pro Tools, choose Avid Link >
Marketplace) can also be downloaded, installed, and authorized using Avid Link or from your online Avid
account. These plug-ins are authorized using an iLok license that can be saved to an iLok USB key, iLok
Cloud, or your computer.
This topic includes information about the following:

l "Installing Plug-ins Bundled with Pro Tools Subscriptions" below
l "Installing Plug-Ins from the Avid Marketplace" below
l "Installing Plug-Ins for Pro Tools and Media Composer" below
l "About iLok" on the next page
l "Authorizing Avid Audio Plug-Ins" on the next page
l "Removing Plug-Ins" on page 25
Installing Plug-ins Bundled with Pro Tools Subscriptions

If you are a new Pro Tools software subscriber, start by downloading and installing the Avid Link app. Avid
Link lets you manage all of your Avid products and licenses. For more information, refer to the Avid Link
documentation online.
g In the Avid Link preferences, enable the Install Apps Silently option to have all plug-ins to which you
are entitled installed automatically. You will be prompted to enter the administrator username and
password for your computer. Additionally, all silently installed applications (such as Pro Tools) and
plug-ins with be automatically authorized with your iLok Cloud account. If you want to move any of
these authorizations to a physical iLok or to your local computer, do so using the iLok License
Manager application.
Installing Plug-Ins from the Avid Marketplace

All plug-ins purchased from the Avid Marketplace through Pro Tools are installed silently and — following
activation — are available for use without having to restart Pro Tools.
To install a plug-in from the Avid Marketplace:
1. Launch Pro Tools (or Avid Link).

a. Choose either of the following:
b. Avid Link > Marketplace in Pro Tools.
2. Select Avid Marketplace in the Avid Link app.
3. Find the plug-in you want in the Avid Marketplace and follow the on-screen instructions.
Installing Plug-Ins for Pro Tools and Media Composer

g For information on installing plug-ins for VENUE, refer to your VENUE documentation.
23
To install a plug-in:
1. Log in to your Avid Master Account online (my.avid.com).

2. Download the appropriate installer for your operating system (Mac or Windows) from your Avid
Master Account. After downloading, make sure the installer is uncompressed (.pkg or .exe).
3. Locate and double-click the plug-in installer.
4. Follow the on-screen instructions to complete the installation.
About iLok
The plug-ins documented in this guide can be authorized using an iLok USB or USB-C key from PACE Anti-
Piracy. Plug-ins may also be authorized to iLok Cloud or to your computer using the iLok License Manager
application.
iLok USB-C key (4th generation)
iLok USB key (3rd generation)
An iLok can hold hundreds of authorizations for all of your iLok-enabled software. After a software license is
placed on an iLok, you can use the iLok to authorize that software on any computer.
An iLok USB key is not supplied with plug-ins or software options. You can use the iLok included with certain
Pro Tools systems, or purchase one separately. For more information, visit the iLok website (www.iLok.com).
Authorizing Avid Audio Plug-Ins

When you purchase a boxed version of an Avid Audio plug-in, you receive an activation code on an
activation card. If you purchase or rent a plug-in through the Avid Marketplace, the license is deposited
directly to your iLok account and you can skip the following procedure (see "Installing Plug-Ins from the
Avid Marketplace" on the previous page).
g If you open a session that uses plug-ins that are not installed and authorized on your system, you are
prompted to rent or purchase those missing plug-ins though the Avid Marketplace if available.
To authorize Avid Audio plug-ins:
1. If you don’t already have an iLok account, visit www.ilok.com to sign up for an account.
2. Visit avid.com/redemption and log into your Avid account (if you don’t already have an Avid account,
click “Create Your Account”).
3. Enter your activation code and your iLok.com User ID.
4. Follow the on-screen instructions to deposit your license into your iLok.com account.
24
5. Once the activation process is complete, the download links for your Avid audio plug-in will be
available in the My Products section of your Avid account.
6. Download and install the plug-in that you purchased.
7. If you are authorizing a plug-in using an iLok, make sure your iLok is connected to an available USB
port on your computer.
8. Launch Pro Tools (or Media Composer) and follow the on-screen instructions to transfer the plug-in
license to your iLok, iLok Cloud, or your computer and authorize the plug-in.
Removing Plug-Ins
If you need to remove a plug-in from your Pro Tools system, follow the instructions below for your computer
platform.
Removing Plug-Ins on Mac

To remove a plug-in:
1. Locate and open the Plug-Ins folder on your Startup drive (Library/Application
Support/Avid/Audio/Plug-Ins).
2. Do one of the following:
t Drag the plug-in to the Plug-Ins (Unused) folder.
t Drag the plug-in to the Trash and empty the Trash.
Removing Plug-Ins on Windows
To remove a plug-in:
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.
Adjusting Plug-In Controls

You can adjust plug-in controls by dragging on-screen controls, by editing control values, or by dragging in
graphic displays.

l "Dragging Plug-In Controls" below
l "Editing Control Values" on page 27
l "Dragging in Graphic Displays" on page 27
l "Toggle Controls" on page 28
l "Adjusting Controls with Fine Resolution" on page 28
l "Resetting Controls to Default Values" on page 28
l "General Effects Controls" on page 28
l "MIDI Control" on page 29
Dragging Plug-In Controls

Rotary Controls
Some plug-ins have rotary controls that can be adjusted by dragging over them horizontally or vertically.
25
To adjust a rotary control:
1. Click on the control.

2. Do any of the following:
t Drag up or to the right to increment the control.
t Drag down or to the left to decrement the control.
Adjusting a rotary control by dragging (EQ III)
Slider Controls
Some plug-ins have slider controls that can be adjusted by dragging horizontally.
Some sliders are bipolar, meaning that their zero position is in the center of the slider’s range. Dragging to
the right of center yields a positive value, and dragging to the left of center yields a negative value.
To adjust a slider control:
1. Click on the control.

t Drag to the right to increment the control.
t Drag to the left to decrement the control.
Adjusting a slider control by dragging (ReVibe II)
26
Editing Control Values
Some controls have text boxes that display the current control value. You can edit the control value directly.
To edit control values:
1. Click in the text box corresponding to the control that you want to adjust.
t Type a new value. The following illustration show the value field for the EQ III plug-in.
For controls that support values in kilohertz, typing “k” after a numeric value will multiply the
value by 1000.
t Increment the value by scrolling up with a mouse or scroll wheel, or press the Up Arrow key.
t Decrement the value by scrolling down with a mouse or scroll wheel, or press the Down Arrow
key.
t Press Enter on the numeric keyboard to input the value and remain in keyboard editing mode.
t Press Return (Mac) or Enter (Windows) on the alpha keyboard to enter the value and leave
keyboard editing mode.
To move forward through control text boxes in a plug-in:
t Press the Tab key.
To move backward through control text boxes in a plug-in:
t Press Shift+ Tab.
Dragging in Graphic Displays
Some plug-ins have graphic displays with control points that you can drag to adjust the corresponding
controls.
Dragging a control point (EQ III)
Dragging a control point (ReVibe II)
27
Toggle Controls
Some plug-ins have toggles that let you set different effects modes or even bypass the plug-in.
To change the state of a toggle control:

t Click the toggle switch or button.
Left: Clicking a toggle switch (BBD Delay). Right: Clicking a toggle button (BBD Delay).
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.
Resetting Controls to Default Values

You can reset any on-screen control to its default value by Option-clicking (Mac) or Alt-clicking (Windows)
directly on the control or on its corresponding text box.
General Effects Controls

Line/Inst, Input, Mix, and Output
Some plug-ins provide Line/Inst, Input, Mix, and/or Output controls at the bottom of the Plug-in window
(though not all of the plug-ins provide all four of these controls).
Input, Mix, and Output controls
Line/Inst
Lets you set the input gain as appropriate to your source signal. If the source is an instrument level signal
(like an electric guitar), select Inst. If the source is a line level signal (such as a drum loop), select Line.
Input
Lets you adjust the audio signal input gain into the plug-in from –20 to 12 dB.
Mix
Lets you mix the Wet (processed) signal and the Dry (source) signal. Set the Mix control to 0% for all dry
signal, to 50% for equal wet and dry signal, and to 100% for all wet signal.
Output
Lets you adjust the audio signal output gain from the plug-in from –20 to 12 dB.
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Bypass Toggle and LED
All of the “stomp box” plug-ins provide a Bypass toggle. This has the same effect as the standard Bypass in
the Plug-in window header. These plug-ins also provide a Bypass LED to show the current bypass state. The
LED is lit when the effect is active, and is unlit when the effect is bypassed.
Bypass toggle and LED
Sync
The time-based effects (such as Chorus, Delay, and Flanger) can be set to synchronize with the Session
tempo (including tempos set with the Tap Tempo button). Simply enable the Sync parameter in the effect
you want to synchronize to the Session tempo.
When the Sync control on these effects is set to a rhythmic subdivision of the incoming tempo, the effect
locks to it. When Sync is set to Off, or the Rate or Delay control is moved, manual control takes over, and
the rate of modulation or delay can be set by hand.
MIDI Control
Avid effects plug-ins support MIDI Control Change (CC) messages, meaning that plug-in parameters can
be controlled remotely by any connected CC-capable MIDI device.
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.
MIDI control assignments are saved and restored with the Pro Tools session in which they are defined.
Settings files (presets) for Avid effects plug-ins do not store or recall MIDI Learn assignments.
g Once you set up a session with your preferred plug-in configuration, save it as a Pro Tools session
template. The template will save your MIDI control assignments and you can create future sessions
based on this template.
g VENUE and Media Composer do not support MIDI.

To map a MIDI controller to a plug-in control:
1. Make sure your external MIDI device is connected to your system, and recognized by your MIDI
Studio Setup (Windows) or Audio MIDI Setup (Mac).
2. Create a MIDI track.
3. Set the input of the MIDI track to accept input from your external MIDI device.
4. Set the output of the MIDI track to the plug-in you want to control.
29
Assigning a MIDI Track Output to a plug-in
5. Right-click on any control in the plug-in, and do one of the following:

t Click Learn, then move a control on your MIDI controller. Pro Tools maps whichever control
you touch to that plug-in parameter.
t If you know the MIDI CC value of your foot controller or other device, select it from the Assign
menu.
Right-clicking a plug-in control to Learn MIDI CC
g On Mac, you can Control-click a parameter to show the MIDI Assign menu. However, note that you
won’t be able to use the Control key modifier to “clutch” a Grouped control.
To clear a MIDI assignment:

t Right-click the control and choose Forget.
MIDI Control and Multi-Mono Plug-ins
Multi-mono plug-ins report virtual MIDI nodes for each channel. For example, a multi-mono plug-in inserted
on a stereo track reports two separate MIDI nodes, one for the left channel and one for the right (Black /
Shiny Wah 1 and Black / Shiny Wah 2).
As long as the channels of the mulit-mono plug-in are linked, any one MIDI node will control the plug-in on
all channels. However, if the channels are unlinked, only the corresponding MIDI node will control the multi-
mono instance of that plug-in on that channel. For example, if a multi-mono instance of Black / Shiny Wah
is inserted on a stereo track, and it is unlinked, the MIDI node Black / Shiny Wah 1 controls the multi-mono
instance on the left channel only, and Black / Shiny Wah 2 controls the right.
30
2 EQ Plug-Ins
Equalization (EQ) lets you shape the frequency spectrum of the sound. A simple example of equalization are
the bass and treble controls on many stereo systems. You can use these controls to boost (make louder) or
attenuate (make quieter) the low and high frequencies of the audio. You can use EQ to sharpen drums,
emphasize vocals, and even to cut out unwanted noise.

l "304E Equalizer" below
l "EQ III" on the next page
l "Eleven Effects Graphic EQ" on page 49
l "Focusrite D2" on page 50
l "Pultec Plug-Ins" on page 55
304E Equalizer
304E equalizer is an EQ plug-in that is available in DSP, Native, and AudioSuite formats.
304E supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
304E operates as a mono, multi-mono, or stereo plug-in.
304E offers simple controls with incredibly warm, musical results.
304E Equalizer Controls

Operation of 304E equalizer is simple and to the point.
Bass
Adjusts low frequencies ±11.
Mid and Mid Freq
Adjusts mid frequencies, from 500Hz to 3.5KHz, ±11.
Treble
Adjusts high frequencies ±11.
Gain
The Gain control allows you to adjust the output level ±11.
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EQ III
The EQ III plug-in provides high-quality 1-Band and 7-Band EQ for adjusting the frequency spectrum of
audio material.
EQ III is available in DSP, Native, and AudioSuite formats.
EQ III supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
EQ III operates as a mono, multi-mono, 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.
EQ III Configurations
The EQ III plug-in appears as two separate choices in the plug-in insert selector and in the AudioSuite menu:
l EQ3 1-Band
l EQ3 7-Band
1-Band EQ
The 1-Band EQ has its own window, with six selectable filter types for a single band of EQ.
7-Band EQ
The 7-Band EQ has its own window, with up to seven separate bands, each with it its own set of filter types.
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Adjusting EQ III Controls
In addition to dragging controls and typing control values, there are other ways to adjust EQ III controls.
Inverting Filter Gain

(Peak EQ Bands Only) 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

You can adjust the following 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.
Gain
Dragging a control point up increases the Gain setting. Dragging a control point down decreases the Gain
setting.
Control-dragging (Mac) or Start-dragging (Windows) a control point up decreases the Q setting. Control-
dragging (Mac) or Start-dragging (Windows) a control point down increases the Q setting.
33
Dragging a control point in the Frequency Graph display
Using EQ III in Band-Pass Mode

You can temporarily set any EQ III control to Band-Pass monitoring mode. Band-Pass mode cuts monitoring
frequencies above and below the Frequency setting, leaving a narrow band of mid-range frequencies. It is
especially useful for adjusting limited bandwidth in order to solo and fine-tune each individual filter before
reverting the control to notch filter or peaking filter type operations.
g Band-Pass mode does not affect EQ III Gain controls.

To switch an EQ III control to Band-Pass mode:
t Hold Control+Shift (Mac) or Start+Shift (Windows), and drag any rotary control or control point
horizontally or vertically.
EQ III interactive graph displaying Band-Pass mode
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.
Sets the width of the narrow band of mid-range frequencies centered around the Frequency setting.
To switch an EQ III control out of Band-Pass mode:

t Release Control+Shift (Mac) or Start+Shift (Windows).
34
Controlling EQ III from a Control Surface
EQ III can be controlled from any supported control surface, including EUCON-compatible control surfaces,
D-Control, D-Command, C|24, and 003. Refer to the guide that came with the control surface for details.
EQ III I/O Controls

Certain Input and Output controls are found on all EQ III configurations, except where noted otherwise.
I/O controls and meters for 7-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 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.
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 Only) The plasma-style Input and Output meters show peak signal levels before and after EQ
processing, and indicate them as follows:
l Green
Indicates nominal levels
l Yellow
Indicates pre-clipping levels, starting at –6 dB below full scale
l 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.
35
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, the HPF, LPF, LF, and HF sections have EQ Type selectors to toggle between the two
available filter types in each section.
EQ Type Selectors (7-Band EQ)
Band Enable Button

(7-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.
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.
36
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.
Q Control
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.
Q control
1-Band EQ III
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.
37
Frequency Graph display
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.
Band Controls
The individual EQ types have some combination of the following controls, as noted below.
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 III Types

High Pass Filter
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.
38
1-Band EQ set to High Pass Filter
Notch Filter
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.
1-Band EQ set to Notch Filter
High Shelf EQ
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.
1-Band EQ set to High Shelf EQ
39
Low Shelf EQ
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.
1-Band EQ set to Low Shelf EQ
Peak EQ
The Peak EQ boosts or cuts a band of frequencies centered around the Frequency setting. The width of the
affected band is determined by the Q setting.
1-Band EQ set to Peak EQ
Low Pass Filter
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.
40
1-Band EQ set to Low Pass Filter
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.
41
7-Band EQ III High Pass/Low 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.
High Pass Filter
Attenuates all frequencies below the Frequency setting at the selected slope while letting all frequencies
above pass through.
Low Notch EQ
Attenuates a narrow band of frequencies centered around the Frequency setting. The width of the
attenuated band is determined by the Q setting.
High Pass filter (left) and Low 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:
Control Value
Frequency Range 20 Hz to 8 kHz

Frequency Default 20 Hz
HPF Slope Values 6, 12, 18, or 24 dB/oct
Low Notch Q Range 0.1 to 10.0
Low Notch Q Default 1.0
7-Band EQ III Low Pass/High Notch

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.
Low Pass Filter
Attenuates all frequencies above the Frequency setting at the selected slope while letting all frequencies
below pass through.
42
High Notch EQ
Attenuates a narrow band of frequencies centered around the Frequency setting. The width of the
attenuated band is determined by the Q setting.
Low Pass filter (left) and High Notch EQ (right)
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

Frequency Default 20 kHz
HPF Slope Values 6, 12, 18, or 24 dB/oct
High Notch Q Range 0.1 to 10.0
High Notch Q Default 1.0
7-Band EQ III Low Shelf/Low Peak

The Low Shelf/Peak band is switchable between low shelf EQ and low peak EQ functions. By default, this
band is set to Low Shelf.
Low Shelf EQ
Boosts or cuts frequencies at and below the Frequency setting. The amount of boost or cut is determined by
the Gain setting. The Q setting determines the shape of the shelving curve.
Low Peak EQ
Boosts or cuts a band of frequencies centered around the Frequency setting. The width of the affected
band is determined by the Q setting.
43
Low Shelf EQ (left) and Low Peak EQ (right)
The Low Shelf and Low Peak Gain controls and their corresponding graph elements are displayed on-screen
in red. The following control values are available:
Control Value
Frequency Range 20 Hz to 500 Hz

Low Shelf Q Range 0.1 to 2.0
Low Peak Q Range 0.1 to 10.0
Q Default 1.0
Low Shelf Gain Range –12 dB to +12 dB
Low Peak Gain Range –18 dB to +18 dB
7-Band EQ III Low Mid Peak

The Low Mid Peak band boosts or cuts frequencies centered around the Frequency setting. The width of the
44
Low 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:
Control Value

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
7-Band EQ III Mid Peak

The Mid Peak band boosts or cuts frequencies centered around the Frequency setting. The width of the
Mid Peak EQ
The Mid Gain control and its corresponding graph elements are displayed on-screen in yellow. The following
control values are available:
Control Value

45
Control Value
Mid Peak Q Range 0.1 to 10.0

Mid Peak Q Default 1.0
Mid Peak Gain Range –18 dB to +18 dB
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
High Mid Peak EQ
The High Mid Gain control and its corresponding graph elements are displayed on-screen in green. The
following control values are available:
Control Value

Mid Peak Q Range 0.1 to 10.0
Mid Peak Q Default 1.0
Mid Peak Gain Range –18 dB to +18 dB
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
46
High Shelf EQ (left) and High Peak EQ (right)
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

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
Frequency Graph Display

(7-Band EQ Only) The Frequency Graph display in the 7-Band EQ shows a color-coded control dot that
corresponds to the color of the Gain control for each band. The filter shape of each band is similarly color-
coded. The white frequency response curve shows the contribution of each of the enabled filters to the
overall EQ curve.
47
Frequency Graph display for the 7-Band EQ
48
Eleven Effects Graphic EQ
Graphic EQ
Graphic EQ is useful for simple frequency sculpting. Use Graphic EQ before other effects, such as
Distortion, to shape the sonic results of additional processing.
Graphic EQ is available as a Mono or Stereo plug-in.
Frequency Band Gain
Graphic EQ provides gain controls for five frequency bands: 100 Hz, 370 Hz, 800 Hz, 2 kHz, and
3.25 kHz.
The 370 Hz, 800 Hz, and 2 kHz bands can be boosted by up to 18 dB and attenuated down by as much
as –18 dB.
The 100 Hz and 3.25 kHz bands can be boosted by up to 12 dB and attenuated down by as much as –
12 dB.
Output Gain
The Output gain control lets you boost the plug-in output by as much as +6 dB or attenuate the output by
as much as – 20 dB.
49
Focusrite D2
Focusrite D2 is a high-quality digital equalizer plug-in for Pro Tools. Developed in cooperation with
Focusrite, the D2 is based on the acclaimed Red Range 2™ dual EQ, designed by Rupert Neve. It provides up
to six simultaneous bands of EQ, including: high-pass, low-shelf, low-mid peak, high-mid peak, high-shelf,
and low-pass filters. D2 includes a highly accurate Cartesian graph that displays EQ curves in real-time as
EQ controls are adjusted.
D2 is available in DSP, Native, and AudioSuite formats.
D2 supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
D2 operates as a mono, multi-mono, or stereo plug-in.
D2 Configurations
There are three configurations of the Focusrite D2 plug-in.
1–2 Band EQ
D2 1-2 Band can use up to two filters simultaneously, depending on which you enable. The high-pass, low-
shelf, and low-pass filters each utilize the entire module and cannot be used in combination with another
filter. The low-mid peak, high-mid peak, or high-shelf filters can be used in combination with each other (up
to two bands total).
4-Band EQ
D2 4-Band can use up to four filters simultaneously. Any combination of filters can be engaged, up to a
total of four bands.
6-Band EQ
D2 6-Band can use up to six filters simultaneously. Any combination of filters can be engaged, up to a total
of six bands. By default, the low-pass and high-pass filters are in Bypass mode when the 6-Band EQ is first
opened.
50
D2 Controls
The lower left corner of D2 provides input controls, output controls, and meters.
Input Level
Input Level allows you to attenuate signal input level to the D2. The range of this control is from –18 dB to
+12 dB.
When you use D2 in stereo, each channel has its own separate Input Level knob. To adjust input levels for
both channels simultaneously, select the Link button, then drag either knob.
Output Level
Output Level allows you to adjust the overall output gain. The range of this control is from –18 dB to +12 dB.
When you use the D2 plug-in in stereo, each channel has its own separate output level knob. To adjust
output levels for both channels simultaneously, select the Link button.
Meters
The D2 high-resolution plasma-style meters indicate signal levels and detect clipping at the input,
algorithm, or output stage. When D2 is used in stereo, two meters appear, one for each channel.
A Clip Indicator is located above each meter. It indicates clipping by increasing its brightness as successive
samples are clipped. Click the Clip Indicator to clear it. Option-clicking (Mac) or Alt-clicking (Windows)
clears both channels when D2 is used in stereo.
The following metering indications are used:

l Green = nominal levels
l Yellow = pre-clipping at –6 dB below full scale signal
l Red = full scale signal (clipping)
Frequency Display
The frequency display is a visual representation of the current EQ settings. As you adjust the controls of any
currently active filter, the display plots the changes to the EQ curve in real-time. If you are using D2 in
stereo, the frequency display shows the EQ curve for the right channel in red and the left channel in blue.
Cartesian Graph
51
g To reset all D2 controls to their default settings, Option-click (Mac) or Alt-click (Windows) the
frequency display. To reset controls for both channels when in Stereo mode, Option-Shift-click (Mac)
or Alt-Shift-click (Windows) the frequency display.
EQ Filter Controls
Each of the six different EQ filters has its own controls and its own icon. The icons act as three-state
switches for enabling, disabling, or bypassing the specific filter. The current state of a filter is indicated by
its color:
l White = enabled. In this state the filter is active, audible, and using available DSP resources.
l Black = disabled. In this state the filter is not using any DSP resources and has no effect on audio.
l Gray = bypassed. In this state the filter is not active, but is still using available DSP resources. The
effect of the filter is not audible.
High-Pass Filter
The 18 dB/octave High-Pass Filter provides a rotary control for adjusting the corner (cutoff) frequency,
variable from 20 Hz to 6.4 kHz.
Low-Shelf Filter
The Low-Shelf Filter provides two rotary controls: The upper rotary control adjusts the corner frequency,
variable from 33 Hz to 460 Hz. The lower rotary control adjusts the filter’s amplitude gain or attenuation.
Amplitude range is ±15 dB from unity.
Low-Mid Peak Filter
The Low-Mid Peak Filter provides three rotary controls. The upper rotary control adjusts the center
frequency, variable from 33 Hz to 6.4 kHz. The lower left rotary control adjusts the filter’s amplitude gain or
attenuation. Amplitude range is ±15 dB from unity (utilizing a reciprocal curve for both gain and
attenuation). The lower right rotary control adjusts filter “Q” which is variable from 0.7 to 4.0.
High-Mid Peak Filter
The High-Mid Peak Filter provides three rotary controls. The upper rotary control adjusts the center
frequency, variable from 120 Hz to 18 kHz. The lower left rotary control adjusts the filter’s amplitude gain or
attenuation. Amplitude range is ±15 dB from unity (utilizing a reciprocal curve for both gain and
attenuation). The lower right rotary control adjusts filter “Q” which is variable from 0.7 to 4.0.
High-Shelf Filter
The High-Shelf Filter provides two rotary controls: The upper rotary control adjusts the corner frequency,
variable from 3.3 kHz to 18 kHz. The lower rotary control adjusts the filter’s amplitude gain or attenuation.
Amplitude range is ±15 dB from unity.
52
Low-Pass Filter
The 18 dB/octave Low-Pass Filter provides a rotary control for adjusting the filter’s cutoff frequency,
variable from 100 Hz to 18 kHz.
Enabling, Disabling and Bypassing EQ Filters

You can enable, disable, or bypass specific EQ filters by clicking them.
To disable a filter:
t Control-click (Mac) or Start-click (Windows) the EQ Filter icon. When disabled, the icon is black.
To re-enable a filter:
t Click the EQ filter icon. When enabled, the icon is white.
To bypass a filter:
t Click the EQ filter icon a second time. When bypassed, the icon is gray.
g If you are using all available bands of the 1–2 Band or 4–Band EQ and want to change filter types, you
must disable one filter before you can enable a different one.
Using D2 in Stereo
Because Focusrite D2 has a single set of Filter control knobs, when it is used in stereo, you must select which
channel, left or right, you want to edit.
Left Channel and Right Channel Buttons

The Left Channel and Right Channel buttons are used to select which controls are active.
Left Channel, Right Channel, and Link buttons
Link Button
The Link button lets you adjust controls for both channels simultaneously. By default, Link mode is enabled
so that you can maintain parity between channels.
You can also use Link mode to help you maintain a relative offset between control settings on the two
channels.
53
To maintain an offset between channels:
1. Deselect the Link button.

2. Select a channel button, left or right, and adjust the controls for that channel.
3. Select Link mode and adjust the same controls for the opposite channel. D2 will maintain the relative
offset between the two channels.
g To copy the control settings of the active channel to the opposite channel, Option-click (Mac) or Alt-
click (Windows) while linking channels.
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Pultec Plug-Ins
The Pultec plug-ins are a set of EQ plug-ins that are available in DSP, Native, and AudioSuite formats. The
following plug-ins are included:
l Pultec EQP-1A
l Pultec EQH-2
l Pultec MEQ-5
The Pultec plug-ins support 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
The Pultec plug-ins operate as mono, multi-mono, or stereo plug-ins.
g Pultec plug-ins support EQ Curve in Pro Tools 2018.1 and later.

Pultec EQP-1A
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.
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. 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 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.
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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
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.
The Pultec EQH-2 offers three equalization sections: low frequency boost and attenuation, midrange boost
only, and 10k attenuation. Like its EQP-1A sibling, it features high-quality transformers and a tube gain
stage. But unlike the EQP-1A, the tube stage in the EQH-2 is a push-pull design. As a result, the EQH-2 offers
a beefier tone.
Pultec EQH-2 Controls

Low Frequency Section
Adjust low frequencies using the top row of Boost and Atten knobs and the CPS (cycles per second) switch.
All low-frequency equalization is a gentle shelving type, 6 dB per octave.
High Frequency Boost Section
Boost mid and high frequencies using the KCS (kilocycles per second) and Boost knobs on the second row.
High Frequency Attenuate Section
Cut high frequencies using the 10k Atten knob located at the right side of the plug-in.
Pultec MEQ-5
The Pultec MEQ-5 is the most unique equalizer in the Pultec family. It is particularly useful on individual
tracks during mixdown.
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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.
Low Frequency Peak
Boost low frequencies (200, 300, 500, 700, 1000 Hz) using the upper left controls.
Mid Frequency Peak
Boost mid-frequencies (1.5k, 2k, 3k, 4k, 5k) using the controls at the upper right.
Wide-Range Dip
Cut frequencies using Dip controls on the bottom row.
Pultec Tips and Tricks

“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.
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.
Our ears are 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.
“Q” and Boost

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.
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 (such as Guitar 1A and 1B or 2A and 2B) for further examples of this technique.
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3 Dynamics Plug-Ins
Dynamics (Compression, Limiting, Expansion, Gating, and other) processing lets you smooth the dynamics
of your audio to control output levels. Compression can keep loud parts from getting too loud, to make
vocals sound more intimate, or to keep cymbals from sounding too shrill. Use a limiter to keep peaks in the
audio signal from exceeding a certain threshold without affecting audio that doesn’t exceed that level.

l "304C Compressor" below
l "BF-2A" on page 61
l "BF-3A" on page 63
l "BF76" on page 64
l "Channel Strip" on page 67
l "Dynamics III" on page 84
l "Eleven Effects Gray Compressor" on page 102
l "Fairchild Plug-Ins" on page 103
l "Focusrite D3" on page 106
l "Maxim" on page 115
l "Pro Compressor" on page 120
l "Pro Expander" on page 132
l "Pro Limiter" on page 143
l "Pro Multiband Dynamics" on page 152
l "Purple Audio MC77" on page 164
304C Compressor
304C compressor is a dynamics processing plug-in that is available in DSP, Native, and AudioSuite formats.
304C supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
304C operates as a mono, multi-mono, or stereo plug-in.
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304C 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.
304C Compressor Controls

304C compressor provides the following controls:
Input Gain
Adjusts the input level to the compressor.
Compression
Affects the gain to the side-chain of the compressor. Use it along with Slope to adjust the amount of
compression.
Output Gain
Provides makeup gain after compression.
Slope
Similar to the compression ratio controls found on other compressors. However, on the 304C, 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 (use 5 to create severe pumping effects).
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
Sets the time during which signal returns to normal after compression. With longer release times, the
compression is less noticeable.
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304C 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.
304C 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 304C compressed sound is louder than it really is—but without the
strange psychoacoustic effect of “deadness” that other compressors suffer from.
Overshoot
At fast Attack settings, it is possible to make the 304C “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.
Attack and Release Times

It may be difficult to understand the interactions between the Attack and Release controls, because the
304C compressor behaves very differently than typical compressors. Experimentation is the best option,
but an explanation may help you understand what’s going on.
304C 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, 304C 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.
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BF-2A
BF-2A is a vintage-style compressor plug-in that is available in DSP, Native, and AudioSuite formats.
BF-2A supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
BF-2A operates as a mono, multi-mono, or stereo plug-in.
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. The BF-2A has been meticulously crafted
to capture every nuance of the legendary LA-2A tube studio compressor, providing the most authentic
vintage compression sound available.
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 Electro-Optical 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 electroluminescent 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.
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:
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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.
Meter
Both Gain Reduction and Output metering are provided. The Meter knob operates as follows:
l 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.
l When set to Output, the needle indicates the output level of the signal. The meter is calibrated with
0 VU indicating –18 dBFS.
Using the BF-2A Side-Chain Filter

The BF-2A provides an extra, a side-chain filter, that does not have a control on the plug-in interface, but
that can be accessed on-screen 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.
l A setting of zero means that the filter is not applied to the side chain signal.
l A setting of 100 means that all frequencies below 250 Hz are filtered out of the side chain signal.
To access the side-chain filter on-screen:
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 double-click a
control in the list).
3. Click OK to close the plug-in automation window.
4. In the Edit window, do one of the following:
t Click the Track View selector and select Side-Chain Filter from the BF-2A sub-menu.
t Reveal an Automation lane for the track, click the Automation Type selector and select Side-
Chain 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).
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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 control.
g For more information on plug-in automation, see the Pro Tools Reference Guide
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.
BF-3A
BF-3A is a vintage-style compressor plug-in that is available in DSP, Native, and AudioSuite formats.
BF-3A supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
BF-3A operates as a mono, multi-mono, or stereo plug-in.
BF-3A is based on the classic LA-3A that adds a smoothness and sonic texture that makes sounds jump
right out of the mix. Designed and manufactured in the late 1960s, the original LA-3A shares many
components in common with the LA-2A compressor. Just like the LA-2A, the heart of the LA-3A is the T4B
Electro-Optical Attenuator. This is a device that converts audio to light and back and is largely responsible
for the compression character of the unit.
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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.
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:
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.
Meter
Both Gain Reduction and Output metering are provided. The Meter knob operates as follows:
l 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.
l 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

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.
BF76
BF76 is a vintage-style compressor plug-in that is available in DSP, Native, and AudioSuite formats.
BF76 supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
BF76 operates as a mono, multi-mono, or stereo plug-in.
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Modeled after the solid-state 1176 studio compressor, BF76 preserves every sonic subtlety of this classic
piece of studio gear.
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.
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 controls 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 60 and 1,100 milliseconds.
Ratio
The Ratio Push switches select the compression ratio from 4:1 to 20:1.
Meter
The Meter Push switches affect the metering.

l GR shows the amount of gain reduction.
l –18 and –24 show the output level (calibrated so that 0VU indicates –18dB FS and –24dB FS
respectively).
l The “Off” switch turns off the meter.
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BF76 Tips and Tricks
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.
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 a 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 a 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.
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Channel Strip
Avid Channel Strip is available in DSP, Native, and AudioSuite formats. Channel Strip provides EQ,
Dynamics, Filter, and Gain effects. Channel Strip processing algorithms are based on the Euphonix System
5 console channel strip effects.
Channel Strip, Compressor / Limiter tab shown
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.
g Greater-than-stereo formats are only available with Pro Tools Ultimate and Studio.
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 setting 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 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.
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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 plug-in to
use less screen space. These sections are shown by default.
To hide (or show) the Dynamics or EQ / Filters section of the plug-in window:
t Click the Show / Hide triangle to the left of the section you want to show or hide.
Channel Strip, Dynamics section hidden
Disabling or Enabling Channel Strip Effects

You can independently disable effects in the Dynamics and EQ / 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.
Dynamics section, Exp / Gate disabled
To enable effects in the Dynamics or EQ / Filters section:

t 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:
t 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.
l 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).
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l 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.
l When enabled for either of the Filter effects, Listen solos the enabled Filter band and inverts the
Filter. This allows you to hear only hear the portion of the audio signal that is being removed by the
filter.
To enable (or disable) Listen on the Side Chain effect, EQ band, or a Filter effect:
t 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.
Channel Strip, Chain Listen mode enabled
g 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.
Channel Strip Input Section

The Input section provides input metering, and controls for trimming the input signal and inverting its phase.
It can also be toggled to show post-processing gain reduction meters.
Input section (5.1 channel format shown)
Input Trim Control

The Input Trim control sets the input gain of the plug-in before EQ processing, letting you make up gain or
prevent clipping at the plug-in input stage.
To Trim the input signal, do one of the following:

t Click in the Input Trim field and type a Trim value (–36.0 dB to +36.0 dB).
t Click Trim and drag up or down to adjust the Input Trim setting.
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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:

t Click the Phase Invert button so that it is highlighted. Click it again so that it is not highlighted to
disable it.
Input Meters
The Input meters show peak signal levels before processing:
l Dark Blue
Indicates nominal levels from –INF to –12 dB.
l Light Blue
Indicates pre-clipping levels, from –12 dB to 0 dB.
l White
Indicates full scale levels from 0 dB to +6 dB.
Gain Reduction Meters
The Input meter can be switched to show Gain Reduction metering for the processed signal from 0 dB to –
36 dB.
The Gain Reduction meters are usually displayed in yellow. When the Knee setting for either or both the
Expander and the Compressor is greater than 0 dB, the Gain Reduction meter displays the amount of the
Knee level in amber over the meter’s usual yellow display.
To toggle between the Gain Reduction and Input meters:

t Click the Input / Gain Reduction toggle in the top right-hand corner of the Input section.
Toggling between Input and Gain Reduction meters
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Channel Strip Output Section
The Output section provides output metering and controls for adjusting the level of the output signal.
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 "Channel Strip FX Chain"
on the next page.
To adjust the Output Volume, do one of the following:

t Click in the Output Volume field and type a value (–INF dB to +12 dB).
t Click VOL and drag up or down to adjust the Output Volume setting.
Output Meters
The Output meters show peak signal levels after processing:
l Dark Blue
l Light Blue
l White
Indicates full scale levels from 0 dB to +6 dB (which can result in distortion and clipping).
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Channel Strip 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.
To set the FX Chain:
1. Click the FX Chain show / hide button to reveal the Process Order options.
Showing the FX Chain Process Order
2. Click an effects chain ordering option to select it. The available options include:
– EQ > FILT > DYN
– 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.
Bypassing or Unbypassing Individual Effects Modules
In the FX Chain display, you can deselect or select individual effects modules to bypass or unbypass the
effect.
FX Chain, FILT bypassed
To bypass an effect module:

t Click the module so that it is not highlighted.
To unbypass an effect module:
t Click the module so that it is highlighted.
Channel Strip Dynamics Section
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.
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Dynamics section, All tab shown
Dynamics Graph
Dynamics graph display
The Dynamics Graph display—used with Expander / Gate and Compressor / Limiter processing—shows a
curve that represents the level of the input signal (on the horizontal x–axis) and the amount of gain
reduction applied (on the vertical y–axis). The display shows two vertical lines representing the Threshold
setting for the Expander / Gate and Compressor / Limiter, respectively.
The Dynamics Graph display also features an animated red ball in the gain transfer curve display. This ball
shows the amount of input gain (x-axis) and gain reduction (y-axis) being applied to the incoming signal at
any given moment. To indicate overshoots (when an incoming signal peak is too fast for the current
compression setting), the cursor temporarily leaves the gain transfer curve.
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Use this graph as a visual guideline to see how much dynamics processing you are applying to the incoming
audio signal.
Dynamics Graph Gain Reduction Resolution
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:

t Click the Graph Resolution toggle.
Using the Dynamics Graph to Adjust Controls
You can drag in the Dynamics Graph display to adjust the corresponding Expander / Gate and Compressor
/ Limiter controls. The cursor updates to show which control is being adjusted:
l Expander / Gate Ratio
l Expander / Gate Knee
l Expander / Gate Threshold
l Gate Depth
l Hysteresis
l Compressor / Limiter Ratio
l Compressor / Limiter Knee
l Compressor / Limiter Threshold
l Limiter Depth
g For the Expander / Gate and Compressor Limiter effects, adjusting a control in the Dynamics Graph
display automatically shows the pane that includes the adjusted control if it is not already shown
(except when the All tab is shown).
Expander / Gate Controls
Dynamics section, Expander / Gate tab
Threshold
The Threshold (Thresh) control sets the level below which an input signal must fall to trigger expansion or
gating. Signals that fall below the threshold will be reduced in gain. Signals that are above it will be
unaffected.
The Dynamics Graph display shows the threshold as a vertical line.
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Attack
The Attack control sets the attack time, or the rate at which gain is reduced after the input signal crosses
the threshold. Use this along with the Ratio setting to control how soft the Expander’s gain reduction curve
is.
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.
Depth
The Depth control sets the depth of the Expander / Gate when closed. Setting the gate to higher range
levels allows more and more of the gated audio that falls below the threshold to peek through the gate at all
times.
Hold
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.
Release
The Release control sets how long it takes for the gate to close after the input signal falls below the
threshold level and the hold time has passed.
Knee
The Knee control sets the rate at which the Expander / Gate reaches full effect once the threshold has been
exceeded.
Hysteresis
The Hysteresis (Hyst) control lets you adjust whether or not the gate rapidly opens and closes when the
input signal is fluctuating near the Threshold. This can help prevent undesirably rapid gating of the signal.
Hysteresis applies a differential threshold for signals that are rising, as opposed to signals that are falling,
to reduce the chance of the gate opening when you don’t want it to. For example, if the threshold is set to –
30 and Hysteresis (Hyst) is set to 10, the gate opens when the signal reaches –20 and closes when signal
reaches the set threshold of –30. This control is only available when Ratio is set to Gate, otherwise it is
grayed out.
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Compressor / Limiter Controls
Dynamics section, Compressor / Limiter tab
Threshold
The Threshold control sets the level that an input signal must exceed to trigger compression or limiting.
Signals that exceed this level will be compressed. Signals that are below it will be unaffected.
Attack
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.
At the limiter setting (LMTR), for every decibel that the incoming signal goes over the set Threshold, 1 dB of
gain reduction is applied.
Compressor / Limiter Ratio set to LMTR
Once the Ratio control passes the LMTR setting, it provides negative ratio settings from –20.0:1 to 0:1.
Compressor / Limiter Ratio set to a negative value
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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.
Release
The Release control sets the length of time it takes for the Compressor / Limiter to be fully deactivated
after the input signal drops below the threshold.
Release times should be set long enough that if signal levels repeatedly rise above the threshold, the gain
reduction “recovers” smoothly. If the release time is too short, the gain can rapidly fluctuate as the
compressor repeatedly tries to recover from the gain reduction. If the release time is too long, a loud section
of the audio material could cause gain reduction that continues through soft sections of program material
without recovering.
Knee
The Knee control sets the rate at which the compressor reaches full compression once the threshold has
been exceeded.
As you increase this control, it goes from applying “hard-knee” compression to “soft-knee” compression:
l With hard-knee compression, compression begins when the input signal exceeds the threshold. This
can sound abrupt and is ideal for limiting.
l With soft-knee compression, gentle compression begins and increases gradually as the input signal
approaches the threshold, and reaches full compression after exceeding the threshold. This creates
smoother compression.
Gain
The Gain control lets you boost overall output gain to compensate for heavily compressed or limited
signals.
Side Chain Processing Controls
Dynamics section, Side Chain tab
Dynamics processors typically use the detected amplitude of their input signal to trigger gain reduction.
This split-off signal is known as the side-chain. Compressor / Limiter and Expander / Gate processing
features external key capabilities and filters for the side-chain.
With external key side-chain processing, you trigger dynamics processing using an external signal (such as
a separate reference track or audio source) instead of the input signal. This external source is known as the
key input.
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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.
l 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.
l Key
If Key is selected, the plug-in uses the amplitude of a separate reference track or external audio
source to trigger dynamics processing. The reference track used is selected using the Plug-In Key
Input selector in the Plug-In window header. With greater-than-stereo multichannel processing, the
key signal triggers dynamics processing for all processed audio channels equally.
l 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.
Detection
The Detection options include Peak or Avg (Average).

l Peak
Select the Peak option to apply side-chain processing according to the detected peak amplitude.
l Average
Select the Average option to apply side-chain processing according to the detected average
amplitude.
Filter Frequency
The Filter Frequency control lets you set the frequency for the selected Filter Type.
l Filter Type
Four Filter Type options are available for side-chain processing:
– Low Pass
Select the Low Pass option to apply a low pass filter to the side-chain processing at the
selected frequency.
– High Pass
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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
The Side Chain Processing Graph display shows the frequency curve for the selected Filter Type at the
selected Filter Frequency.
Channel Strip EQ / Filters Section

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 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:

t Click the Graph Resolution toggle.
Clicking and Dragging in the Frequency Graph to Adjust Controls
You can adjust the following EQ controls by dragging the control points directly in the Frequency Graph
display:
g Control-Shift-click (Mac) or Start-Shift-click (Windows) the control point for any frequency band to
solo that frequency band. Drag left or right to change the frequency value.
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l Frequency
Dragging a control point to the right increases the Frequency setting. Dragging a control point to the
left decreases the Frequency setting.
g 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.
l Gain
Dragging a control point up increases the Gain setting. Dragging a control point down decreases the
Gain setting.
g Option-Shift-click (Mac) or Alt-Shift-click (Windows) an EQ control point to invert its Gain

setting.
l Q
Click within the curve of an EQ control point and drag up or down to increase or decrease the Q
setting.
g 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.
Common Controls
Each Frequency band EQ and Filter provides the following common controls:
l Enable / disable
l Listen mode
Enable / Disable
Click the Enable / Disable button to enable or each band.
Low Frequency EQ Controls
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.
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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.
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.
Low Mid Frequency EQ Controls
EQ / Filters section, Low Mid Frequency tab
The LMF tab provides controls for the low mid frequency band of the EQ. This band is a peak EQ.
Frequency
The Frequency control lets you set the center frequency for the peak low mid frequency band.
Gain
The Gain control lets you boost or attenuate the corresponding frequencies for the low mid frequency
band.
The Q control changes the width of the low mid peak EQ band. Higher Q values represent narrower
bandwidths. Lower Q values represent wider bandwidths.
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High Mid Frequency EQ Controls
EQ / Filters section, High Mid Frequency tab
The HMF tab provides controls for the high mid frequency band of the EQ. This band is a peak EQ.
Frequency
The Frequency control lets you set the center frequency for the peak high mid frequency band.
Gain
The Gain control lets you boost or attenuate the corresponding frequencies for the high mid frequency
band.
The Q control changes the width of the high mid peak EQ band. Higher Q values represent narrower
bandwidths. Lower Q values represent wider bandwidths.
High Frequency EQ Controls
EQ / Filters section, High Frequency tab
The High Frequency EQ tab provides controls for the high frequency band of the EQ.
Filter Type
The High Frequency band can be set to be a Peak or High Shelf EQ.
Frequency
The Frequency control lets you set the center frequency for the high frequency band (Peak or Shelf EQ).
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Gain
The Gain control lets you boost or attenuate the corresponding frequencies for the high frequency band.
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.
Filter 1 and Filter 2 Controls
EQ / Filters section, Filter 1 tab shown
The Filter 1 and Filter 2 tabs provide the same set of controls for each filter.
Filter Type
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 selected Filter Type (from 20 Hz to
21.0 kHz).
Slope
When the Filter Type is set to Low Pass or High Pass, the Slope control is available. The Slope control lets
you set the slope for the filter from the selected Frequency to –INF (12 dB/O or 24 dB/O).
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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Dynamics III
Dynamics III is a suite of three dynamics plug-ins available in DSP, Native, and AudioSuite formats:
l Compressor / Limiter (see "Compressor / Limiter III" on page 87)
l Expander / Gate (see "Expander / Gate III" on page 91)
l De-Esser (see "De-Esser III" on page 93)
Dynamics III supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
The Compressor / Limiter and Expander / Gate modules support mono, stereo, and greater-than-stereo
multichannel formats up to 7.1.
The De-Esser module supports 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 Common Controls

The Levels section, the LFE Enable button, and the Dynamics Graph display of the user interface are shared
between the Compressor / Limiter, Expander / Gate, and De-Esser plug-ins.
Dynamics III Levels Section

The indicators and controls in the Dynamic III Levels section let you track input, output, and gain reduction
levels, as well as work with phase invert and the threshold setting.
g See "De-Esser III Level Meters" on page 94 for more information on De-Esser III Input / Output Level
controls.
I/O Meter display (stereo instance shown)
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The Input (In) and Output (Out) meters show peak signal levels before and after dynamics processing:
l Green
Indicates nominal levels.
l Yellow
Indicates pre-clipping levels, starting at –6 dB below full scale.
l 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.
g 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.
g When Side-Chain Listen is enabled, the Output meter only displays the levels of the side-chain signal.
See "Dynamics III Side-Chain Listen" on page 97.
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
Gain Reduction Meter
The Gain Reduction (GR) meter indicates the amount the input signal is attenuated (in dB) and shows
different colors during dynamics processing:
l Light Orange
Indicates that gain reduction is within the “knee” and has not reached the full ratio of compression.
l 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.
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Dynamics III LFE Enable
(Pro Tools Ultimate and Studio Only)
The LFE Enable button (located in the Options section) is on by default, and enables plug-in processing of
the LFE (low frequency effects) channel on a multichannel track formatted for 5.1, 6.1, or 7.1 surround
formats. To disable LFE processing, deselect this button.
LFE Enable button (Compressor / Limiter III shown)
g 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 plug-ins—shows a
curve that represents the level of the input signal (on the horizontal x–axis) and the level of the output signal
(on the vertical y–axis). The orange vertical line represents the threshold.
Use this graph as a visual guideline to see how much dynamics processing your are applying.
Dynamics graph display
The Compressor / Limiter and Expander / Gate plug-ins 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.
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Gain transfer curve and cursor showing amount of compression
To indicate overshoots (when an incoming signal peak is too fast for the current compression setting) the
cursor temporarily leaves the gain transfer curve.
The cursor changes color to indicate the amount of compression applied, as shown in the following table:
Cursor Color Compression Amount
white no compression
light orange below full ratio
dark orange full ratio amount
g See "De-Esser III Frequency Graph Display" on page 95 for information on using the De-Esser 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 100.
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.
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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
can be 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.
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 84 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.
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 86.
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.
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Threshold arrow on input meter
The Dynamics Graph display also shows the threshold as an orange vertical line.
Threshold indicator on Dynamics Graph display
This control ranges from –60 dB (lowest gain) to 0 dB (highest gain).
Compressor / Limiter III Ratio Control

The Ratio control sets the compression ratio, or the amount of compression applied as the input signal
exceeds the threshold. For example, a 2:1 compression ratio means that an input level that is 2 dB above the
threshold will be attenuated, resulting in an output level that is 1 dB over the threshold.
This control ranges from 1:1 (no compression) to 100:1 (hard limiting).
Compressor / Limiter III Attack Control

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.
This control ranges from 10 µs (fastest attack time) to 300 ms (slowest attack time).
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Compressor / Limiter III Release Control
The Release control sets the length of time it takes for the Compressor / Limiter to be fully deactivated after
the input signal drops below the threshold.
without recovering.
This control ranges from 5 ms (fastest release time) to 4 seconds (slowest release time).
Compressor / Limiter III Knee Control

been exceeded.
Graph examples of hard knee (left) and soft knee (right) compression
For example, a Knee setting of 10 dB would be the gain range over which the ratio gradually increased to
the set ratio amount.
The Gain Reduction meter displays light orange while gain reduction has not exceeded the knee setting,
and switches to dark orange when gain reduction reaches the full ratio.
This control ranges from 0 db (hardest response) to 30 db (softest response).
Compressor / Limiter III Gain Control

The Gain control lets you boost overall output gain to compensate for heavily compressed or limited signals.
This control ranges from 0 dB (no gain boost) to +40 dB (loudest gain boost), with the default value at 0 dB.
g For more information on the LFE channel, refer to the Pro Tools Reference Guide.
Compressor / Limiter 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 side-chain between the internal input signal or an external key input,
and tailor the equalization of the side-chain signal so that the triggering of dynamics processing becomes
frequency-sensitive. See "Dynamics III Side-Chain Input" on page 96.
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Expander / Gate III
The Expander / Gate plug-in applies expansion or gating to audio material, depending on the ratio setting.
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.
Expander / Gate III Input / Output Level Meters

The Input and Output meters show peak signal levels before and after dynamics processing. See "Dynamics
III Levels Section " on page 84 for more information.
Expander / Gate III Dynamics Graph Display

The Dynamics Graph display lets you visually see how much expansion or gating you are applying to your
audio material. See "Dynamics III Graph Display" on page 86.
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.
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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Expander / Gate III Threshold Control
The Threshold (Thresh) control sets the level below which an input signal must fall to trigger expansion or
gating. Signals that fall below the threshold will be reduced in gain. Signals that are above it will be
unaffected.
An orange arrow on the Input meter indicates the current threshold, and can also be dragged up or down to
adjust the threshold setting.
Threshold arrow on Input meter
The Dynamics Graph display also shows the threshold as an orange vertical line.
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 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 threshold. Use this along with the Ratio setting to control how soft the Expander’s gain reduction curve
is.
This control ranges from 10 µs (fastest attack time) to 300 ms (slowest attack time).
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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.
This control ranges from 5 ms (shortest hold) to 4 seconds (longest hold).
Expander / Gate III Release Control

The Release control sets how long it takes for the gate to close after the input signal falls below the
threshold level and the hold time has passed.
This control ranges from 5 ms (fastest release time) to 4 seconds (slowest release time).
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 side-chain between the internal input signal or an external key input,
and tailor the equalization of the side-chain signal so that the triggering of dynamics processing becomes
frequency-sensitive. See "Dynamics III Side-Chain Input" on page 96.
De-Esser III
The De-Esser reduces sibilants and other high frequency noises that can occur in vocals, voice-overs, 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, frequency-dependent 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.
Using De-Essing Effectively
To use de-essing most effectively, insert the De-Esser after compressor or limiter plug-ins.
The Frequency control should be set to remove sibilants (typically the 4–10 kHz range) and not other parts
of the signal. This helps prevent de-essing from changing the original character of the audio material in an
undesired manner.
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Similarly, the Range control should be set to a level low enough so that de-essing is triggered only by
sibilants. If the Range is set too high, a loud, non-sibilant section of audio material could cause unwanted
gain reduction or cause sibilants to be over-attenuated.
To improve de-essing of material that has both very loud and very soft passages, automate the Range
control so that it is lower on soft sections.
g 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.
De-Esser III I/O Meter display
The Input and Output meters show peak signal levels before and after dynamics processing:
l Green
Indicates nominal levels.
l Yellow
Indicates pre-clipping levels, starting at –6 dB below full scale.
l 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:
l Light Orange
Indicates that gain reduction is being applied, but has not reached the maximum level set by the
Range control.
l Dark Orange
Indicates that gain reduction has reached the maximum level set by the Range control.
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De-Esser III Frequency Control
The Frequency (Freq) control sets the frequency band in which the De-Esser operates. When HF Only is
disabled, gain is reduced in frequencies within the specified range. When HF Only is enabled, the gain of
frequencies above the specified value will be reduced.
This control ranges from 500 Hz (lowest frequency) to 16 kHz (highest frequency).
De-Esser III Range Control

The Range control defines the maximum amount of gain reduction possible when a signal is detected at the
frequency set by the Frequency control.
This control ranges from –40 dB (maximum 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.
De-Esser III Listen Control

When enabled, the Listen button lets you monitor the sibilant peaks used by the De-Esser as a side-chain to
trigger compression. This is useful for listening only to the sibilance for fine-tuning De-Esser controls. To
monitor the whole output signal without this filtering, deselect the Listen button.
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.
De-Esser graph display
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Dynamics III Side-Chain Input
(Compressor / Limiter and Expander / Gate Only)
This split-off signal is known as the side-chain. The Compressor / Limiter and Expander / Gate plug-ins
feature external key capabilities and filters for the side-chain.
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.
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
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.
External Key button
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Dynamics III Side-Chain Listen
When enabled, this control lets you listen to the internal or external side-chain input by itself, as well as
monitor its levels with the Output meter. This is especially useful for fine-tuning the plug-in’s filter settings
or external key input.
Side-Chain Listen button
g Side-Chain Listen is not saved with other plug-in presets.

Dynamics III Side-Chain HF and LF Filter Enable Buttons
The HF Filter Enable and LF Filter Enable buttons toggle the corresponding filter in or out of the side-chain.
When this button is highlighted, the filter is applied to the side-chain signal. When this button is dark gray,
the filter is bypassed and available for activation.
HF and LF Filter Enable buttons
Dynamics III Side-Chain High-Frequency (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.
l 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.
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Band-Pass filter
l 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
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
20 Hz to 20 kHz.
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.
l 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.
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l 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.
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
20 Hz to 20 kHz.
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Using Dynamics III Key Input for Side-Chain Processing
To use a filtered or unfiltered external key input to trigger dynamics processing:
1. Click the Key Input selector and select the input or bus carrying the audio from the reference track or
external audio source.
Selecting a Key Input
2. Click External Key to activate external side-chain 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).
Side-Chain Listen
4. To filter the key input so that only specific frequencies trigger the plug-in, use the HF and LF controls
to select a frequency range.
5. 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.
Using a Filtered Input Signal for Side-Chain Processing with Dynamics III
To use the filtered input signal to trigger dynamics processing:
1. Ensure the Key Input selector is set to No Key Input.
2. Ensure that the External Key button is disabled (dark gray).
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3. To listen to the signal that will be used to control side-chain input, click Side-Chain Listen to enable it
(highlighted).
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 a 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.
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Eleven Effects Gray Compressor
Gray Compressor
Inspired by a well-loved ’70s solid-state compressor pedal, Gray Compressor can add singing sustain to
leads or lend power and girth to chunky rhythm guitar parts.
Gray Compressor is available as a Mono (or Multi-Mono) plug-in only.
Sustain
The Sustain control lets you adjust the threshold, and thus, the amount of compression, in the
Compressor effect.
Level
The Level control sets the overall output volume of the effect.
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Fairchild Plug-Ins
The Fairchild plug-ins are a pair of vintage compressor plug-ins that are available in DSP, Native, and
AudioSuite formats.
The Fairchild plug-ins support 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
The Fairchild plug-ins operate as mono, multi-mono, or stereo plug-ins.
Fairchild 660
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.
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.
Fairchild 660 Controls

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.
Input Gain
Input Gain sets the input level to the unit and the compression threshold, just like the Input control on an
1176. Full clockwise is loudest.
Threshold
Threshold adjusts the gain to the sidechain, just like the Peak Reduction control on an LA-2A.
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Time Constant
Selects the attack and release times for the compressor. One is fastest, and six is slowest. Seven and eight
are custom presets.
Fairchild 660 Tips and Tricks

5,6,7,8…
The Fairchild manual documents Time Constant settings 5 and 6 as user presets—although you have to go
inside with a soldering iron to change them. We used the “factory default” values.
Bonus Settings
Settings 7 and 8 do not exist on real-world units—well, at least most of them. These settings are taken from
a real-world Fairchild modification invented by Dave Amels many years before he designed the plug-in
version.
What do they do? Settings 7 and 8 offer versions of Time Constant 2 with a gentler release useful for
compressing vocals and other program material where you desire more subtlety in the compression action.
Give them a try—you’ve already heard them on hit songs on the radio.
Pump It Up
With a carefully adjusted Input Gain and Threshold, you can use Time Constant 1 to achieve a cool
pumping effect on drums. The sound gets darker and fuller, and sits beautifully in a mix.
Fairchild 670
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!
The internal design of the Fairchild 670 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.
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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 the previous page 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.
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Focusrite D3
Focusrite D3 is a high-quality dynamics processor plug-in. Developed in cooperation with Focusrite, the D3
is based on the highly acclaimed Red Range 3™ dual mono/stereo compressor & limiter designed by Rupert
Neve.
D3 is available in DSP, Native, and AudioSuite formats.
D3 supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
D3 operates as a mono, multi-mono, or stereo plug-in.
D3 features include:
l Compressor+Limiter. This configuration allows you to use both the compressor and the limiter at the
same time. The Compressor+Limiter plug-in requires twice as much DSP as the Compressor / Limiter.
l Compressor / Limiter. This configuration allows you to use either the compressor or the limiter—but
not both at the same time. The Compressor / Limiter plug-in uses half as much DSP as the
Compressor+Limiter. It is provided so that you can conserve DSP, since you may not need both
compression and limiting at the same time.
The Compressor / Limiter defaults to the compressor being enabled and the limiter disabled.
D3 Compressor
The D3 compressor reduces 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 D3 applies as input levels increase is referred to as the
compression ratio. This parameter is adjustable. If you set the compression ratio to 2:1, for example, for
each 2 dB that the signal exceeds the threshold, the output level will be reduced to 1 dB above the threshold.
With a compression ratio of 4:1, an 8 dB increase in input will produce only a 2 dB increase in output.
D3 Limiter
The D3 limiter operates as a fast-attack compressor with a high compression ratio. It does not attack
instantaneously or look ahead in order to attack ahead of time, but instead uses a very fast, 1-millisecond
attack time. As such, the D3 is not a “brick wall” limiter, but limits the overall dynamic range of signals in a
sonically-pleasing way.
Like the Compressor, the Limiter is activated when the signal exceeds the user-selected threshold. The
Limiter then compresses any signal above the selected threshold down to the threshold limit that you have
set.
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To enable the limiter:
1. Disable the compressor by Control-clicking (Mac) or Start-clicking (Windows) its icon.

2. Click the Limiter icon.
Compressor and Limiter icons
D3 Side-Chain Processing
Compressors and limiters generally use the detected amplitude of the input signal as a control source.
Other signals can also be used as a control source by using a key input. With de-essing, for example, a
frequency-modified version of the input signal is used as a trigger. This is known as side-chain processing.
Side-chain processing allows the D3 compression or limiting to be controlled by another independent audio
signal. In this way you can compress or limit one track’s audio using the dynamics of a different track’s
audio.
Using D3 in Stereo
In stereo configurations, all D3 controls except the Input Level affect both channels of the stereo signal. The
D3 RMS detector (which derives the control signal that drives the dynamics processing) uses a composite of
the two channels. Because of this, when stereo processing occurs, there is no image shift when signal levels
differ between the two channels (since the composite control signal drives processing for both channels).
D3 Common Controls
The following D3 controls are common to both the compressor and limiter.
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Input Level
Input Level attenuates signal input level to the compressor or limiter. The range of this control is from –
30 dB to 0 dB.
When you use the stereo version of the D3 plug-in, each channel has its own separate Input Level knob. To
adjust input levels for both channels simultaneously, Shift-drag. Option-Shift clicking (Mac) or Alt-Shift-
clicking (Windows) either Input Level knob resets both channels to 0 dB.
Output Level
Output Level adjusts the overall output gain. Because large amounts of compression can restrict dynamic
range, the Output Level knob is useful for compensating for heavily compressed signals and making up the
resulting difference in level.
When you use the stereo version of the D3 plug-in, this single knob controls the master output for both
channels. The range of this control is from –12 dB of attenuation to +18 dB of gain.
External Key and Key Listen

The side-chain is the split-off signal used by a plug-in’s detector to trigger dynamics processing. External
Key lets you designate an external source (known as the key input) for the side-chain, while Key Listen lets
you monitor the key input.
External Key and Key Listen toggles
g See "Using the Side-Chain Input in D3" on page 111 for detailed information on external side-chain
processing.
Meters
The meters indicate gain reduction (the top meter) and output level (the bottom meter). The Gain Reduction
meter indicates the amount of gain reduction in dB. The Output Level meter indicates the output signal level
in dB.
In Stereo mode, two Output Level meters appear, one for each channel. However, a single Gain Reduction
meter is used for both channels, since the D3 RMS detector uses a composite control signal.
A red Clip Indicator appears to the right of the output meter(s). Clicking on the Clip Indicator clears it.
Option-clicking (Mac) or Alt-clicking (Windows) clears both channels when the plug-in is used in stereo.
The following metering indications are used:
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l Green = nominal levels
l Yellow = pre-clipping at –6 dB below full scale signal
l Red = full scale signal (clipping)
D3 Compressor Controls
The Compressor icon, which represents a compression curve, acts as a three-state switch for enabling,
disabling, or bypassing the compressor. Its current state is indicated by the icon’s color.
l White indicates enabled. In this state the compressor is active and using available DSP resources.
l Black indicates disabled. In this state, the compressor is not using DSP resources.
l Gray indicates bypassed. In this state the compressor is not active, but is still using available DSP
resources.
To disable the compressor:
t Control-click (Mac) or Start key-click (Windows) the icon. When the compressor is disabled, the icon
is black.
To re-enable the compressor:
t Click the icon. When the compressor is enabled, the icon is white.
To bypass the compressor:
t Click the icon a second time. When the compressor is bypassed, the icon is gray.
If you are using the Compressor / Limiter plug-in, which allows you to use either the compressor or the
limiter (but not both simultaneously), you must disable one module by Control-clicking (Mac) or Start-
clicking (Windows) the icon before you can enable the other.
Ratio
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. The range of this control is from 1.5:1 (very little compression), to 10:1
(heavy compression, bordering on limiting).
Threshold
Threshold sets the threshold level. Signals that exceed this level will be compressed. Signals that are below it
will be unaffected. The range of this control is from 0 dB to –48 dB. A setting of 0 dB is equivalent to no
compression.
Attack
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 doesn’t decrease the overall volume. The range of this control is from 1.0 ms to
150.0 ms.
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Release
Release controls how long 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 section of the audio material could cause gain reduction that persists through a soft
section. The range of this control is from 25 milliseconds to 2.5 seconds.
Auto Release
Auto Release enables the automatic release function. In this mode the Release control has no effect on
release time. Instead, the D3 uses a release time value that is program dependent and based on the audio
being processed.
D3 Limiter Controls
The Limiter icon, which represents a limiter curve, acts as a three-state switch for enabling, disabling, or
bypassing the limiter. Its current state is indicated by the icon’s color:
l White indicates enabled. In this state, the limiter is active and using available DSP resources.
l Black indicates disabled. In this state, the limiter is not using DSP resources.
l Gray indicates bypassed. In this state, the limiter is not active, but is still using available DSP
resources.
To disable the limiter:

t Control-click (Mac) or Start key-click (Windows) the icon. When the limiter is disabled, the icon is
black.
To re-enable the limiter:
t Click the icon. When the limiter is enabled, the icon is white.
To bypass the limiter:
t Click the icon a second time. When the limiter is bypassed, the icon is gray.
If you are using the Compressor / Limiter plug-in, which allows you to use either the compressor or the
limiter (but not both simultaneously), you must disable one module by Control-clicking (Mac) or Start-
clicking (Windows) the icon before you can enable the other.
Limit LED
When lit, the Limit LED indicates that limiting is being applied. When unlit, limiting is not being applied.
Threshold
This sets the threshold level. Signals that exceed this level will be limited. Signals that are below it will be
unaffected. A setting of 0 dB is equivalent to no limiting. The range of this control is from –24 dB to 0 dB.
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Using the Side-Chain Input in D3
The side-chain is the split-off signal used by a plug-in's detector to trigger dynamics processing. D3 lets
you switch between internal and external side-chain processing.
With external side-chain processing, a plug-in's detector is triggered by an external signal (such as a
separate reference track or audio source) known as the key input.
A typical use for this feature is to use a kick drum track to gate and tighten up a bass track, or a rhythm
guitar track to gate another instrument.
External Key
External Key toggles external side-chain processing on or off. When this button is enabled, the plug-in uses
the amplitude of an external signal (the key input) to trigger compression or limiting. When this button is
disabled, the plug-in uses the amplitude of the input signal to trigger dynamics processing.
Key Listen
Key Listen enables and disables auditioning of the key input controlling the external side-chain. This is
useful for fine-tuning the compressor’s settings to the key input.
Using a Key Input for External Side-Chain Processing

To use a key input to trigger dynamics processing:
1. Click the Key Input selector and select the input or bus carrying the audio from the reference track or
external audio source.
Selecting a key input
2. Click External Key to activate external side-chain processing.

3. To listen to the key input that will be used to control side-chain processing, click Key Listen to
enable it.
4. Begin playback. The plug-in uses the input or bus that you chose as an external key input to trigger
its effect.
5. Adjust the plug-in Threshold control to fine-tune external key input triggering.
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Impact
Impact is available in DSP, Native, and AudioSuite formats. 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 supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Impact supports mono, stereo, and greater-than-stereo multichannel formats up to 7.1.
g Greater-than-stereo formats are only available with Pro Tools Ultimate and Studio. Note that the
LFE channel is not processed in multichannel versions of Impact.
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 Release Control

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.
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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 Make-Up control
determines master output levels for all channels. The range of this control is from 0 dB of attenuation to
+40 dB of gain.
g Applying large amounts of Make-Up gain will boost the level of any noise or hiss present in audio
material, making it more audible.
Impact Ext Control (Side-Chain)

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.
g See "Using the Impact Compressor" below 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
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.
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 control is adjustable in discrete increments. If you set the compression ratio to 2:1,
for example, for each 2 dB that the signal exceeds the threshold, the output level will be reduced to 1 dB
above the threshold. With a compression ratio of 4:1, an 8 dB increase in input will produce only a 2 dB
increase in output.
Side-Chain Processing
Compressors generally use the detected amplitude of their input signal as a control source. However, you
can also use other signals, such as a separate reference track or an external audio signal as a control
source. This is known as side-chain processing.
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Side-chain processing lets you control Impact compression using an independent audio signal. 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.
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).
g Remember to disable Listen to resume normal plug-in monitoring.

6. Adjust Impact’s Threshold control to fine-tune Key Input triggering.
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Maxim
Maxim is a unique and powerful peak-limiting and sound maximizing plug-in that is available in DSP, Native,
and AudioSuite formats. Maxim is ideal for critical mastering applications, as well as standard peak-limiting
tasks.
Maxim offers several critical advantages over traditional hardware-based limiters. Maxim takes full
advantage of the random-access nature of disk-based recording to anticipate peaks in audio material and
preserve their attack transients when performing reduction. This makes Maxim more transparent than
conventional limiters, since it preserves the character of the original audio signal without clipping peaks or
introducing distortion.
Maxim supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Maxim operates as a mono, multi-mono, or stereo plug-in.
Maxim features include:

l “Perfect attack-limiting” and look-ahead analysis accurately preserve transient attacks and the
character of original program material
l A full-color histogram plots input dB history during playback and provides visual feedback for setting
threshold level
l A user-adjustable ceiling lets material be level-optimized for recording
l Dither for noise shaping during final mixdown
l Online Help provides descriptions of each control, shown by holding the pointer over any of the
elements in Maxim’s display.
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.
Limiting has two main uses in the audio production cycle:
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l Adjusting the dynamic range of an entire final mixdown for premastering purposes
l Adjusting the dynamic range of individual instruments for creative purposes
Limiting a Mix
The purpose of applying limiting during final mixdown is to flatten any large peaks remaining in the audio
material to have a higher average signal level in the final mix. By flattening peaks that would otherwise clip,
it is possible to increase the overall level of the rest of the mix. This results in higher average audio levels,
potentially better signal to noise ratio, and a smoother mix.
Limiting Individual Instruments

The primary purpose of applying limiting to individual instruments is to alter their dynamic range in subtle
or not-so-subtle ways. A common application of this type of limiting is to modify the character of drums.
Many engineers do this by applying heavy limiting to flatten the snap of the attack portion of a drum hit. By
adjusting the release time of the limiter it is possible to bring up room tone contained in the decay portion of
the drum sound.
In some cases, this type of limiting can actually change a drum’s character from a very dry sound to a
relatively wet sound if there is enough room tone present. This method is not without its drawbacks,
however, since it can also bring noise levels up in the source audio if present.
How Maxim Differs From Conventional Limiters

Maxim is superior to conventional limiters in several ways. Unlike traditional limiters, Maxim has the ability
to anticipate signal peaks and respond instantaneously with a true zero attack time.
Maxim does this by buffering audio with a 1024-sample 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.
c Maxim introduces a delay that is proportional to the session sample rate. 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.
Maxim Controls and Meters

Maxim Input Level Meter
This meter displays the amplitude of input signals prior to limiting. Unlike conventional meters, Maxim’s
Input meter displays the top 24 dB of dynamic range of audio signals, which is where limiting is typically
performed. This provides you with much greater metering resolution within this range so that you can work
with greater precision.
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.
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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.
By dragging the Threshold slider downwards, you can visually adjust the level at which limiting will occur.
Maxim displays the affected range in orange.
Histogram
Maxim Threshold Slider

This slider sets the threshold level for limiting. Signals that exceed this level will be limited. Signals below it
will be unaffected. Limited signal peaks are attenuated to match the threshold level, so the value that you
set here will determine the amount of reduction applied.
Maxim Output Meter

This meter displays the amplitude of the output signal. The value that appears here represents the
processed signal after the threshold, ceiling, and mixing settings have been applied.
Maxim Ceiling Slider

This slider determines the maximum output level. After limiting is performed you can use this slider to adjust
the final output gain. The value that you set here will be the absolute ceiling level for limited peaks.
Maxim Attenuation Meter

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.
Maxim Release Knob

This knob 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 control has a very
noticeable effect on the character of limiting. In general, if you are using heavy limiting, you should use
proportionally longer release times in order to avoid pumping that may occur when Maxim is forced to jump
back and forth between limited and unlimited signal levels. Lengthening the release time has the effect of
smoothing out these changes in level by introducing a lag in the ramp-up or ramp-down time of
attenuation. Use short release times on material with peaks that are relatively few in number and that do
not occur in close proximity to each other. The Release control has a default value of 1 millisecond.
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Maxim Mix Slider
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
control precisely the amount of the processed effect mixed with the original signal.
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.
Applying dither helps reduce quantization noise that can occur when you are mixing from a 24-bit source to
a 16-bit destination, such as CD-R or DAT. If you are using Maxim on a Master Fader during mixdown,
Maxim’s built-in dither function saves you the trouble and DSP resources of having to use a separate Dither
plug-in.
If Dither is disabled, the Noise Shaping and Bit Resolution controls will have no effect.
Maxim Noise Shaping Control

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 Bit Resolution Button

These buttons select dither bit resolution. In general, set this control to the maximum bit resolution of your
destination media.
l 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.
l 18-bit is recommended for output to digital devices that have a maximum resolution of 18 bits.
l 20-bit is recommended for output to digital devices that support a full 20-bit recording data path. It
is 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.
Using Maxim
Following are suggestions for using Maxim most effectively.
To use Maxim:
1. Insert Maxim on a 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.
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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 control 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.
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 converters on some DATs and CD players will clip at or slightly below zero.
g 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 produce 16-bit output from a source with a higher bit depth, 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.
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Pro Compressor
Avid Pro Compressor is available in DSP, Native, and AudioSuite formats. Pro Compressor provides dynamic
compression processing. The Avid Pro Compressor processing algorithms are based on the award-winning
Euphonix System 5 console channel strip effects.
Pro Compressor supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Pro Compressor supports mono, stereo, and greater-than-stereo multichannel formats up to 7.1.
In addition to standard knob and fader controls, Pro Compressor also provides a dynamics graph to track
the gain transfer curve for compression, as well as a frequency graph for side-chain filtering. Additionally,
the dynamics graph can be used to graphically edit the Threshold, Ratio, Knee, and Depth settings.
Pro Compressor Metering

Pro Compressor provides combined meters that show both sample peak metering and averaged metering.
Pro Compressor uses sample peak meters over average metering for Input and Output signals. Attenuation
metering uses sample peak metering only.
The Peak Hold value is displayed numerically at the top of the meter and the Peak Hold indicator appears as
a thin orange line in the meter. This provides highly accurate visual metering correlation with the audio
signal. Pro Compressor also displays averaging metering with an integration time of approximately 400 ms.
Input and Output meters use the following color coding:

l Dark Blue
Indicates nominal levels from –90 to –20 dB.
l Light Blue
l Yellow
Attenuation meters show yellow for the entire dynamic range displayed.
Peak Indicators
The Input and Output meters provide graphical representation of transient peaks, as well as graphical and
numerical display of the last, greatest registered peak (Peak Hold). The Attenuation meter provides similar
graphical and numeric representations for the amount of compression applied to the input signal.
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The numerical display for the Peak value turns orange when the signal exceeds 0 dB on the meters. You can
click the numerical display to reset the displayed value.
Peak indicators in the Output meters
Pro Compressor Input Section

The Input section provides input metering and controls for adjusting the level of the input signal.
Input section with Meters and Input Level control
Input Level
The Input Level control sets the input gain of the plug-in before processing, letting you boost or attenuate
gain at the plug-in input stage.
To adjust the level of the input signal, do one of the following:

t Click in the Input Level field to type a value (–36.0 dB to +36.0 dB).
t Click the Input Level control and drag up or down to adjust the Input Level setting.
Input Meters
The Input meters show peak signal levels before processing.
Pro Compressor Output Section

The Output section provides output metering and controls for adjusting the level of the output signal. The
Output meters can also be toggled to show post-processing gain attenuation meters.
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Output section with Meters and Output Level control
Output Level
The Output Level control sets the output level after processing, letting you make up gain or prevent clipping
on the channel where the Pro Compression plug-in is being used.
To adjust the output level, do one of the following:

t Click in the Output Level field and type a value (–INF dB to +12 dB).
t Click the Output Level control and drag up or down to adjust the Output Level setting.
Output Meters
The Output meters show peak signal levels after processing.
Attenuation Meters
The Output meter can be switched to show Attenuation metering for the processed signal from 0 dB to –
36 dB.
To toggle between the Attenuation and Output meters:

t Click the Output / Attenuation toggle in the top right-hand corner of the Output section.
Output / Attenuation toggle (Attenuation meters shown)
Pro Compressor Dynamics Graph

The Dynamics Graph 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 display shows a vertical line
representing the Threshold setting for the Compressor.
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compression setting), the ball temporarily leaves the gain transfer curve.
audio signal.

You can drag in the Dynamics Graph display to adjust the corresponding Compressor controls. The cursor
updates to show which control is being adjusted:
l Threshold
l Ratio
l Knee
l Depth
To adjust the Threshold setting using the Dynamics graph:
t Position the cursor over the vertical Threshold line in the graph and drag left or right to make the
adjustment.
Cursor indicates Threshold adjustment
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To adjust the Ratio setting using the Dynamics graph:
t Position the cursor over the ratio curve in the graph and drag up or down (or left or right) to make the
adjustment.
Cursor indicates Ratio adjustment
To adjust the Knee setting using the Dynamics graph:

t Position the cursor over the “knee” of the curve in the graph and drag up or down (or left or right) to
make the adjustment.
Cursor indicates Knee adjustment
To adjust the Depth setting using the Dynamics graph:

t Position the cursor over the horizontal Depth line in the graph and drag up or down (or left or right) to
Cursor indicates Depth adjustment
Detection Modes
Pro Compressor provides several different detection options for determining how the compressor responds
to the input signal.
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To change the detection mode for the compressor:
t Click a detection mode from the options available above the Dynamics graph.
Detection mode options (Smart mode selected)
Detection options include the following:

l Smart
Select the Smart option for tracks with diverse input signals, or if you are simply not sure what
detector works best with the given audio material. The Smart option analyzes the incoming signal
and interpolates between the different detection modes as needed. This lets you apply a lot of
compression without distortion or pumping.
l RMS
Select the RMS option to apply processing according to the detected RMS (Root Mean Square)
amplitude of the input signal. The RMS option is similar to the Average option, but with a faster
release time.
l Average
Select the Average option to apply processing according to the detected average amplitude of the
input signal.
l Peak
Select the Peak option to apply processing according to the detected peak amplitude of the input
signal.
l Fast
Select the Fast option for tracks with high and short transient signals, such as a snare drum track.
The Fast option is similar to the Peak option, but with faster attack and release times. However, be
careful when using the Fast option as it distorts earlier than the other options. Be sure to configure
the other compressor settings with this in mind.
Attenuation Listen Mode
Attenuation Listen mode lets you isolate the gain reduction part of the processed audio signal. This can help
you hear what parts of the input signal are triggering compression, which, in turn, can help you better
understand the characteristics of the compressor with the current settings.
To enable (or disable) Attenuation Listen mode:

t Click the Attenuation Listen button (the speaker icon at the top right of the dynamics graph) so that
it is highlighted. The button flashes while Attenuation Listen mode is enabled. To disable it, click the
button again so that it is not highlighted.
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Attenuation Listen mode enabled
Pro Compressor Controls

Threshold
The Threshold control sets the level that an input signal must exceed to trigger compression. A signal will
be compressed if its level exceeds this setting. If the signal level falls below this value, no compression will
occur.
Ratio
The Ratio control sets the compression ratio—the amount of compression applied as the input signal
exceeds the threshold. For example, a 2:1 compression ratio means an input level that is 2 dB above the
threshold will be attenuated, resulting in an output level that is 1 dB over the threshold. The compression
ratio ranges from 1.0:1 to 20.0:1.
Once the Ratio control hits 21.0:1, it displays LMTR. The LMTR setting marks the highest “normal”
compression mode before the onset of negative compression values (from –20.0:1 to 0:1).
At the LMTR setting, for every decibel that the incoming signal goes over the set Threshold, 1 dB of gain
reduction is applied.
Compressor Ratio set to LMTR
Once the Ratio control passes the LMTR setting, it provides negative ratio settings from –20.0:1 to 0:1.
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Compressor 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 applied. Consequently, the output signal is
both compressed and made softer. You can use this as a creative effect, or as a kind of ducking effect when
used with an external key input.
Knee
been exceeded.
Attack
The Attack control sets the attack time, or the rate at which gain is reduced after the input signal level
crosses the threshold.
The smaller the value, the faster the attack. The faster the attack, the more rapidly the compressor 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.
g The actual compression attack time is also dependent on the selected Detection mode. Each mode has
its own attack and release times that are calculated in advance of compression processing. If a slower
Detection mode is selected (such as AVG), the fastest possible actual attack time for compression can
only be about 20 ms. The selected Detection mode similarly affects the compressor release time.
Release
The Release control sets the length of time it takes for compression to be fully deactivated after the input
signal drops below the threshold.
without recovering.
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Depth
The Depth control sets the maximum amount of gain reduction applied regardless of the input signal. For
example, if Ratio is set to LMTR (between 20.0:1 and –20.0:1) and Depth is set to Off, 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.
Dry Mix
The Dry Mix control sets the balance between the compressed signal (wet) and the original signal (dry).
The Dry Mix setting determines how much of the original signal is sent to the output rather than the
processed signal. For example, at 30%, the output will be 30% dry and 70% wet. Turn the Dry Mix knob
counterclockwise to 0% to pass only the processed signal (100% wet). Turn the Dry Mix knob clockwise to
100% to pass only the input signal (100% dry).
Makeup
The Makeup control lets you boost overall output gain to compensate for heavily compressed or limited
signals.
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Pro Compressor Side-Chain Processing
This is known as a side-chain signal. Pro Compressor provides filters for side-chain processing and supports
external key side-chain capabilities.
With external key side-chain processing, you can 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.
Source
The Source selector lets you set the source for side-chain processing: Int-Stereo Pairs, Ext-All
(w/LFE), Int All (no LFE), or Int-Front/Rear.
Selecting the Source for side-chain processing
Internal Source—Stereo Pairs

When Int-StereoPairs is selected, the plug-in uses the amplitude of the input signal to trigger
dynamics processing based on stereo input. With greater-than-stereo multichannel processing, the input
signal for each stereo pair affects only those same channels, and likewise mono channels are affected 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.
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External All (with LFE)
When Ext-All (w/LFE) is selected, the plug-in uses the amplitude of a separate reference track or
external audio source to trigger dynamics processing. The reference track used is selected using the Plug-In
Key Input selector in the Plug-In window header. With greater-than-stereo multichannel processing, the key
signal triggers dynamics processing for all processed audio channels equally.
Internal–All (No LFE)

When Int-All (no LFE) 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.
Internal Front/Rear
For LCRS or greater channel formats, when Int Front/Rear is selected, dynamics processing is applied
based on front channel inputs (LCR) and surround channel inputs (S) independently. For .1 formats, the LFE
Side-Chain Listen Mode

Listen mode lets you hear the input signal for the side-chain to the compressor. This can be either the
external key input or the internal side-chain (including the applied filter).
To enable (or disable) Listen mode on the side-chain:

t Click the Listen button in the top-right corner of the Side-Chain section so that it is highlighted. The
icon flashes while Side-Chain Listen mode is enabled. To disable it, click the button again so that it is
not highlighted.
Side-Chain Listen mode enabled
c Attenuation Listen and Side-Chain Listen can be enabled simultaneously, in which case
Attenuation Listen is audible but Side-Chain Listen is not.
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Side-Chain Filter On/Off
You can use the side-chain input with or without filtering by enabling or disabling the Side-Chain Filter
On/Off button.
To enable (or disable) filtering on the Side-Chain:

t Click the Side-Chain Filter On/Off button on the right side of the Side-Chain section so that it is
highlighted. To disable it, click the button again so that it is not highlighted.
Side-Chain Filter enabled
Side-Chain Filter
The side-chain-filter applies only to the side-chain signal feeding the Pro Compressor detection algorithm.
Compression is triggered only when the signal exceeds the Threshold setting at the frequencies passing
through the side-chain filter.
g Note that the side-chain filter does not apply filtering to the compressed signal. Compression is
applied to all frequencies of the input signal when compression is triggered by the side-chain.
Filter Frequency
The Freq control lets you set the center frequency for the selected Filter Type (from 20 Hz to 21.0 kHz).
Filter Q
Filter Type
l Low Pass
Select the Low Pass option to apply a low pass filter to the side-chain processing at the selected
frequency.
l High Pass
Select the High Pass option to apply a high pass filter to the side-chain processing at the selected
frequency.
l Notch
Select the Notch option to apply a notch filter to the side-chain processing at the selected frequency.
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l 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
The Side-Chain Processing Graph display shows the frequency curve for the selected Filter Type at the
Pro Expander
Avid Pro Expander is available in DSP, Native, and AudioSuite formats. Pro Expander provides dynamic
expansion and gating processing. The Avid Pro Expander processing algorithms are based on the award-
winning Euphonix System 5 console channel strip effects.
Pro Expander supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Pro Expander supports mono, stereo, and greater-than-stereo multichannel formats up to 7.1.
In addition to standard knob and fader controls, Pro Expander also provides a dynamics graph to track the
gain transfer curve for dynamic expansion and gating, as well as a frequency graph for side-chain filtering.
Additionally, the dynamics graph can be used to graphically edit the Threshold, Ratio, Knee, and
Depth settings.
Pro Expander Metering

Pro Expander provides combined meters that show both sample peak metering and averaged metering. Pro
Expander uses sample peak meters over average metering for Input and Output signals. Attenuation
metering uses sample peak metering only.
signal. Pro Expander also displays averaging metering with an integration time of approximately 400 ms.

l Dark Blue
Indicates nominal levels from –90 dB to –20 dB.
l Light Blue
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l Yellow
Attenuation meters show dark blue for the entire dynamic range displayed.
Peak Indicators
The Input and Output meters provide graphical representation of transient peaks, as well as graphical and
numerical display of the last, greatest registered peak. The Attenuation meter provides similar graphical
and numeric representations for the amount of compression.
The numerical display for the Peak value turns orange when the signal exceeds 0 dB on the meters. You can
click the numerical display to reset the displayed value.
Peak indicators in the Output meters
Pro Expander Input Section

Input section with Meters and Input Level control
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Input Level

t Click in the Input Level field and type a value (–36.0 dB to +36.0 dB).
Input Meters
The Input meters show peak signal levels before processing.
Pro Expander Output Section

The Output section provides output metering and controls for adjusting the level of the output signal. The
Output meters can also be toggled to show post-processing gain attenuation meters.
Output section with Meters and Output Level control
Output Level
The Output Level control sets the output level after processing, letting you boost or attenuate gain on the
channel where the Pro Expander plug-in is being used.

t Click in the Output Level field to type a value (–INF dB to +12 dB).
t Click the Output Volume control and drag up or down to adjust the Output Volume setting.
Output Meters
The Output meters show peak signal levels after processing.
Attenuation Meters
The Output meter can be switched to show Attenuation metering for the processed signal from 0 dB to –
36 dB.
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To toggle between the Attenuation and Output meters:
t Click the Output / Attenuation toggle in the top right-hand corner of the Output section.
Output / Attenuation toggle, Attenuation meters shown
Pro Expander Dynamics Graph

The Dynamics Graph display shows a curve that represents the level of the input signal (on the horizontal
x–axis) and the amount of processing applied (on the vertical y–axis). The display shows a vertical line
representing the Threshold setting for the Expander.
compression setting), the ball temporarily leaves the gain transfer curve.
audio signal.

You can drag in the Dynamics Graph display to adjust the corresponding Expander controls. The cursor
updates to show which control is being adjusted:
l Threshold
l Ratio
l Knee
l Depth
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To adjust the Threshold setting using the Dynamics graph:
t Position the cursor over the vertical Threshold line in the graph and drag left or right to make the
adjustment.
Cursor indicates Threshold adjustment
To adjust the Ratio setting using the Dynamics graph:

t Position the cursor over the ratio curve in the graph and drag up or down (or left or right) to make the
adjustment.
Cursor indicates Ratio adjustment
To adjust the Knee setting using the Dynamics graph:

t Position the cursor over the “knee” of the curve in the graph and drag up or down (or left or right) to
Cursor indicates Knee adjustment
To adjust the Depth setting using the Dynamics graph:

t Position the cursor over the horizontal Depth line in the graph and drag up or down (or left or right) to
Cursor indicates Depth adjustment
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Detection Modes
Pro Expander provides several different detection options for determining how the expander responds to the
input signal.
To change the detection mode for the expander:

t Click a detection mode from the options available above the Dynamics graph.
Detection mode options, Smart mode selected

l Smart
Select the Smart option for tracks with diverse input signals, or if you are simply not sure what
detector works best with the given audio material. The Smart option analyzes the incoming signal
and switches between the different detection modes as needed.
l RMS
Select the RMS option to apply processing according to the detected RMS (Root Mean Square)
amplitude of the input signal. The RMS option is similar to the Average option, but with a faster
release time.
l Average
Select the Average option to apply processing according to the detected average amplitude of the
input signal.
l Peak
Select the Peak option to apply processing according to the detected peak amplitude of the input
signal.
l Fast
Select the Fast option for tracks with high and short transient signals, such as a snare drum track.
The Fast option is similar to the Peak option, but with faster attack and release times. However, be
careful when using the Fast option as it distorts earlier than the other options. Be sure to configure
the other compressor settings with this in mind.
l Duck
Select the Duck option to use Pro Expander as a ducker. In order to use Pro Limiter as a ducker, set up
an external key input side-chain bussed from another track. When the Duck option is enabled, the
incoming signal is attenuated when the side-chain input crosses the Threshold setting. For the most
predictable results, set Ratio to GATE. Use the Depth control to adjust the amount of ducking
applied to the input signal. Dialog or voice-over is most commonly used as an external key input to
duck (attenuate) the program material (typically music) on the track where Pro Expander is inserted.
g As a general rule, when ducking program material with dialog or voice-over, set the Attack and
Release controls in a range of 300 to 500 milliseconds.
c When Duck is enabled, the Upward option is automatically disabled.

Attenuation Listen Mode
Attenuation Listen mode lets you isolate the processed part of the audio signal. This can help you hear what
parts of the input signal are triggering expansion or gating, which, in turn, can help you better understand
the characteristics of the current settings of the Expander / Gate.
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To enable (or disable) Attenuation Listen mode:
t Click the Attenuation Listen button (the speaker icon at the top right of the dynamics graph) so that
it is highlighted. The button flashes while Attenuation Listen mode is enabled. To disable it, click the
button again so that it is not highlighted.
Attenuation Listen mode enabled
Pro Expander Controls

Threshold
The Threshold control sets the level below which an input signal must fall to trigger expansion or gating.
Signals that fall below the threshold will be reduced in gain. Signals that are above it will be unaffected.
The Dynamics Graph display shows the threshold as a vertical line.
Ratio
the threshold by one half. At higher ratio levels, Pro Expander 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.
Upward
The Upward button enables Upward Expansion mode. When Upward Expansion mode is enabled,
Pro Expander amplifies signals above the Threshold. When it is disabled, the signal is attenuated when the
signal falls below threshold.
To enable (or disable) Upward Expansion mode:

t Click the Upward button so that it is highlighted. To disable it, click the button again so that it is not
highlighted.
Knee
The Knee control sets the rate at which the processor reaches full expansion or gating effect once the
threshold has been exceeded.
Attack
the threshold. Use this along with the Ratio setting to control how soft Pro Expander’s gain reduction curve
is.
Release
The Release control sets how long it takes for the expansion to close or the gate to open after the input
signal falls below the Threshold level and the Hold time has passed.
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Depth
The Depth control sets the depth of the processing (expansion or gating) when closed. Setting the
processing to higher Depth levels allows more audio that falls below the threshold to remain audible.
Hold
The Hold control specifies the duration (in seconds or milliseconds) during which Pro Expander stays in
effect after the initial attack occurs. This can keep processing 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.
Hysteresis
The Hysteresis control lets you adjust whether or not the gate rapidly opens and closes when the input
signal fluctuates near the Threshold. This can help prevent undesirably rapid gating of the signal. This
control is only available when Ratio is set to Gate.
Dry Mix
The Dry Mix control sets the balance between the processed signal (wet) and the original signal (dry). The
Dry Mix setting determines how much of the original signal is sent to the output rather than the processed
signal. For example, at 30%, the output will be 30% dry and 70% wet. Turn the Dry Mix knob
counterclockwise to 0% to pass only the processed signal (100% wet). Turn the Dry Mix knob clockwise to
100% to pass only the input signal (100% dry).
Lookahead
The Lookahead control lets you add a certain amount of delay in milliseconds for analyzing incoming
audio. All attack transients take a certain amount of time from the onset of the signal to the actual transient
peak (especially those with lower frequencies, like a kick drum). Adjust the Lookahead time to ensure that
processing with an Attack setting of 0 (or at least very short) can be accurate to the true attack time of
transients in the signal.
Note that as soon as Lookahead is engaged, the full amount of delay time is added to the Pro Expander
plug-in processing latency. When Lookahead is set to Off, no additional latency is incurred.
g You can compensate for plug-in processing delay using Automatic Delay Compensation in Pro Tools.
For more information, see the Pro Tools Reference Guide.
Pro Expander Side-Chain Processing

This is known as a side-chain signal. Pro Expander provides filters for side-chain processing and supports
external key side-chain capabilities.
(such as a separate reference track or audio source) instead of the input signal. This external source is
known as the key input.
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Source
The Source selector lets you set the source for side-chain processing: Int-Stereo Pairs, Ext-All
(w/LFE), Int All (no LFE), or Int-Front/Rear.
When Int-StereoPairs is selected, processing is triggered for both the Left and Right channel when the
input signal reaches the threshold on either the Left or the Right channel. With greater-than-stereo
multichannel processing, the input signal for each stereo pair affects only those same channels, and
likewise mono channels are affected 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.
External All (with LFE)
When Ext-All (w/LFE) is selected, the plug-in uses the amplitude of a separate reference track or
external audio source to trigger dynamics processing. Dynamics processing is applied equally to all
channels when the input signal reaches the threshold on any input channel. With greater-than-stereo
multichannel processing, the key signal triggers dynamics processing for all processed audio channels
equally.
The reference track used for side-chain processing is selected using the Plug-In Key Input selector in the
Plug-In window header.
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Configuring a key input for side-chain processing
Internal–All (No LFE)
When Int-All (no LFE) 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
Internal Front/Rear
For LCRS or greater channel formats, when Int-Front/Rear is selected, dynamics processing is applied
based on front channel inputs (LCR) and surround channel inputs (S) independently. For “.1” formats, the
LFE channel is processed independently based on its own input signal.
Side-Chain Listen Mode

Listen mode lets you hear the input signal for the side-chain to the compressor. This can be either the
external key input or the internal side-chain (including the applied filter).
To enable (or disable) Listen mode on the side-chain:

t Click the Listen button in the top-right corner of the Side-Chain section so that it is highlighted. To
disable it, click the button again so that it is not highlighted.
Side-Chain Listen mode enabled
g Attenuation Listen and Side-Chain Listen can be enabled simultaneously, in which case Attenuation
Listen is audible but Side-Chain Listen is not.
Side-Chain Filter On/Off

You can use the Side-Chain input with or without filtering by enabling or disabling the Side-Chain Filter
On/Off button.
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To enable (or disable) filtering on the side-chain:
t Click the Side-Chain Filter On/Off button on the right side of the Side-Chain section so that it is
highlighted. To disable it, click the button again so that it is not highlighted.
Side-Chain Filter enabled
Side-Chain Filter
Filter Frequency
The Freq control lets you set the center frequency for the selected Filter Type (from 20 Hz to 21.0 kHz).
Filter Q
Filter Type

l Low Pass
Select the Low Pass option to apply a low pass filter to the side-chain processing at the selected
frequency.
l High Pass
Select the High Pass option to apply a high pass filter to the side-chain processing at the selected
frequency.
l Notch
Select the Notch option to apply a notch filter to the side-chain processing at the selected frequency.
l 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
The Side-Chain Processing Graph display shows the frequency curve for the selected Filter Type at the
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Pro Limiter
Avid Pro Limiter is available in DSP, Native, and AudioSuite formats. Pro Limiter is a “brickwall” limiter that
provides true peak limiting. Pro Limiter limits incoming audio to the True Peak of the signal, to prevent inter-
sample peaks that could introduce distortion during encoding or analog conversion.
Pro Limiter complies with the ITU-R BS.1770-3 loudness metering standard, including True Peak, Integrated
Loudness, and Loudness Range measurements, and is suitable for both EBU R128 and ATSC A/85 (CALM
Act) broadcast workflows. Pro Limiter also provides a unique Character knob that lets you add soft
saturation for more loudness and greater gain reduction, without the unwanted digital artifacts of standard
brick wall limiters.
Use Pro Limiter to ensure that your mix output never exceeds digital 0 dB when hitting the digital-to-analog
converters on your audio interface.
Pro Limiter supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Pro Limiter supports mono, stereo, and greater-than-stereo multichannel formats up to 7.1.
Pro Limiter Metering

Pro Limiter uses sample peak meters for Input and Output signals.
signal.
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l Dark Blue
l Light Blue
l Yellow
Gain Reduction meters are orange for the entire dynamic range displayed.
Pro Limiter Input Section

The Input section provides input metering and controls for adjusting the level of the input signal and the
display of the Input meters.
Input section with Meters
Input Trim
The Input Trim control sets the input gain of the plug-in before processing, letting you boost or attenuate
To trim the level of the input signal, do one of the following:

t Click in the Input Level field to type a value (–30.0 dB to +30.0 dB).
t Click the Input Trim control and drag up or down to adjust the Input Trim setting.
Dim Input Meter Toggle
Pro Limiter provides a Dim Input Meter option that dims the input meters while highlighting the attenuation
meters. This lets you visually focus more on the gain reduction applied rather than on the incoming signal
levels.
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To toggle the Dim Input Meter option on or off:
t Click the Dim Input Meter toggle in the upper-left corner of the Input section.
Input and Gain Reduction Peak Hold Displays
Pro Limiter displays the Input Sample Peak Hold value (in dB) at the upper left corner of the Input meters
and the Gain Reduction Peak Hold value (in dB) at the upper right corner of the Input meters.
Input Sample Peak Hold Display
Shows the last greatest sample peak value from the input signal (on any channel).
Gain Reduction Peak Hold Display
Shows the last greatest attenuation value applied to the input signal.
To reset either the Input or Gain Reduction Peak Hold display value:
t Click the display value you want to reset.
Input Meters
The Input meters show peak signal levels before processing. The Input meter scaling is shown on the left side
of the Input meters (from –90 dB to +6 dB).
Gain Reduction Meters

The Gain Reduction meters are interleaved with the Input meters. Gain Reduction meters show the amount
of gain reduction applied to the Input signal. The gain reduction amount varies depending on the level of the
Input signal, as well as the Threshold and Character settings.
The Gain Reduction meter scaling is shown on the right side of the Input meters (from 0 dB to –36 dB).
To better see the Gain Reduction meters, you can dim the Input meters (see "Dim Input Meter Toggle" on the
previous page).
Gain Reduction meters shown with Input meters dimmed
Pro Limiter Output Section

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Output section with Meters
Output Meters
The Output meters show peak signal levels after processing. The Output meter scale is shown on the right
side of the Output meters (from –90 dB to +6 dB).
Output Sample Peak Hold Display

Pro Limiter provides a numerical display for the output sample peak hold value (in dB) above the output
meters. To reset the value, click it.
Pro Limiter Controls

Pro Limiter provides controls for setting the Threshold, Ceiling, Character (soft-saturation limiting),
and Release time.
Threshold
The Threshold control sets the level an input signal must exceed to trigger limiting. Signals that fall below
the Threshold setting are unaffected.
To adjust the input threshold, do one of the following:

t Click in the Threshold field to type a value (–30 dB to 0 dB).
t Click the Threshold control and drag up or down.
t Click the Threshold control on the Input meters and drag up or down.
Ceiling
The Ceiling control sets the maximum output level of the audio signal from -30 to 0 dBTP (dB relative to
full scale, measured as a true-peak value).
To adjust the output ceiling, do one of the following:

t Click in the Ceiling field (either below the Ceiling control or below the Output meters) to type a value
between –30 to 0 dBTP.
t Click the Ceiling control and drag up or down.
t Click the Output Ceiling control on the Output meters and drag up or down.
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g Shift-click and drag either the Ceiling control or the Threshold control to link both controls and adjust
them to match the same value.
Character
The Character control adds soft-saturation processing with no additional gain before applying limiting.
Release
The Release control sets the length of time it takes to cease limiting after the input signal crosses the
threshold.
reduction “recovers” smoothly. If the release time is too short, the gain can rapidly fluctuate as the limiter
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.
When the Auto Release option is enabled, the Release control is overridden and the Release value
display is grayed out.
Channel Linking
Pro Limiter provides four different options for determining how limiting processing is applied to greater-
than-stereo multi-channel formats.
Stereo Pairs
When selected, limiting is only applied to the Left and Right stereo pairs only when either the Left or Right
incoming signal exceeds the Threshold setting. Similarly, limiting is only applied to the Left Surround and
Right Surround stereo pairs only when either the Left Surround or Right Surround incoming signal exceeds
the Threshold setting. The processing of the Center channel (if present) is applied separately only when
the incoming signal on for the Center channel exceeds the Threshold setting.
All (w/LFE)
When selected, limiting is applied to all incoming channels whenever any channel exceeds the Threshold
setting.
All (no LFE)
When selected, limiting is applied to all incoming channels whenever any channel, except the LFE channel,
exceeds the Threshold setting. However, the LFE channel is still limited by it’s own signal, but not limiting
the signal on any other channel. This ensures that the LFE never clips on output.
Front/Back
When selected, limiting is only applied to the Left, Center (if present), and Right channels only when the
incoming signal on any front channel input exceeds the Threshold setting. Limiting is only applied to the
Surround channels only when the incoming signal on any surround (or back) channel) exceeds the
Threshold setting.
g Insert a multi-mono instance of Pro Limiter to ensure no linking between channels. Each channel will
trigger its own processing independently of the other channels.
Auto Release
When Auto Release is enabled, Pro Limiter overrides the Release setting and automatically adjusts the
limiter release time based on changes in the program material. When the Auto Release toggle is
disabled, you can set the limiter release time manually using the Release control.
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Listen
Enable Listen to isolate the processed part of the audio signal. This can help you hear what parts of the
input signal are triggering limiting, which, in turn, can help you better understand the characteristics of the
current Threshold, Character, and Release settings.
Pro Limiter Loudness Numeric Displays

Pro Limiter provides numeric displays that show the current loudness or peak level of the processed signal.
Note that Pro Limiter conforms to the ITU-R BS.1770-3 standard for loudness metering. The numeric
loudness values generated by Pro Limiter can be used for both EBU R128 and ATSC A/85 CALM Act loudness
compliance.
Integrated
Displays the current integrated level of the processed signal level in LUFS.
Range
Displays the range of the processed signal level over time in LU.
True Peak
Displays the true peak hold value of the Output signal in dB.
Short Term
Displays the short term output signal level in LUFS.
About LUFS, LU, and dB

LUFS (Loudness Unit, Referenced to Full Scale) is a unit of measurement that expresses loudness levels on an
absolute scale. LU (Loudness Unit) is a unit of measurement for differences between loudness levels
(loudness levels on a relative scale). For example, program material that has a loudness level of –23 LUFS
will be 2 LU quieter than program material that has a loudness level –21 LUFS. LU can also be used as the
units for loudness levels relative to the target level. Note that since K-weighting has been adopted as the
standard for loudness units, LUFS is equivalent with LKFS in Pro Limiter.
Decibels (dB) are an expression of the ratio of two levels: the level to be described (or measured) and a
reference level. Letters after dB (such as dBm) signify the reference level. For example, dBm is referenced to
1 milliwatt, whereas dBu is references to 0.775Vrms. LUFS is a measurement on a decibel scale and is
relative to the loudness level of a stereo (front left and front right) 1 kHz tone peaking at 0 dBFS (0 decibels
at full scale).
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Pro Limiter Histogram and Loudness Meters
Pro Limiter Histogram and Loudness meters
Histogram
Pro Limiter provides a histogram that shows a graphic representation of loudness over time within a window
of 60 seconds. The graph displays True Peak levels as a yellow line and the range of loudness over time as a
blue shadow around the peak level line.
Time Elapsed
Displays the amount of time elapsed since the current analysis pass started in hours, minutes, and seconds
(00:00:00).
Reset Analysis
Click the Reset button to reset the analysis.
Auto Analysis
When Auto is enabled, Pro Limiter automatically pauses the analysis pass when the Pro Tools transport is
stopped. This means that the drop in levels will not be reported with the Integrated and Range values. Note
that it may be useful to disable Auto if you want to include a live audio signal being monitored through
Pro Tools while the transport is stopped.
Run Analysis
Click the Run Analysis (Play) button to enable (lit) or disable (unlit) analysis reporting in the histogram while
the Pro Tools transport is stopped. Pro Limiter runs the analysis when the Pro Tools transport is running
regardless of whether or not the Run Analysis option is enabled. Note that when the Auto analysis option is
enabled, the Run Analysis option is overridden.
Loudness Meters
The Loudness meters to the right of the histogram show the level of the summed output of Pro Limiter. The
meters range from 0 LUFS down to –50 dB LUFS. –23 LUFS is a common standard loudness reference level.
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Momentary Loudness
Provides a display of the loudness range (as in the histogram). The current peak level is shown as a yellow
line using K-weighted metering.
Momentary Loudness Meter (M)
Graphically displays the current true peak level.
Short Term Loudness Meter (S)
Graphically displays the current short term output level.
Integrated Loudness (I)
Graphically displays the current integrated level of the processed signal using K-weighted metering.
g K-weighted metering implements a filter curve that models the human ear's perception of loudness. It
is an integral part of the ITU-R BS.1770 standard for loudness metering.
Adjusting the Loudness Meters View

The Loudness meters show a range of 27 LUFS. The default view shows from –14 to –41 LUFS. Drag up or
down in the histogram to adjust the viewed range for loudness metering from 0 LUFS on the upper end down
to –50 LUFS on the lower end.
AudioSuite Processing with Pro Limiter

When used as an AudioSuite plug-in, Pro Limiter does not provide real-time analysis data in the histogram
or the loudness meter. Once you have rendered the audio selection with Pro Limiter processing, Pro Limiter
loudness reporting updates to display the results of the processing.
Pro Limiter AudioSuite plug-in
AudioSuite Processing with Pro Limiter Loudness Analyzer

Use the Pro Limiter Loudness Analyzer to check loudness levels before rendering Pro Limiter processing if
you need to read the current loudness measurements of your program material.
To analyze audio using the Pro Limiter Loudness Analyzer:
1. Make an audio selection in the Edit window.

2. Choose AudioSuite > Other > Pro Limiter Loudness Analyzer.
3. Click Analyze.
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The Loudness numerical displays update to show the analyzed values (for information on the Loudness
numerical displays, see "Pro Limiter Loudness Numeric Displays" on page 148).
Pro Limiter Loudness Analyzer AudioSuite plug-in
g With the Pro Limiter Loudness Analyzer, the Preview and Render buttons do not do anything useful.
They are simply present as part of the AudioSuite plug-in framework.
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Pro Multiband Dynamics
Avid Pro Multiband Dynamics is an AAX plug-in (DSP, Native, and AudioSuite) that provides 4-band
dynamics processing for Pro Tools systems, along with monitoring across all frequencies in the FFT (Fast
Fourier Transform) display. You can control Pro Multiband Dynamics parameters using graphic controls laid
over the FFT display for immediate visual feedback. You can also edit parameters using knobs, sliders, and
numeric entry fields in each Frequency Band pane.
Pro Multiband Dynamics supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Pro Multiband Dynamics also supports mono, stereo, and greater-than-stereo multichannel formats up to
7.1.
g Greater-than-stereo formats are only available with Pro Tools Ultimate and Pro Tools Studio software.
FFT Display and Controls

Pro Multiband Dynamics splits the incoming audio signal into up to four separate Frequency Bands: Low,
Low Mid, High Mid, and High. Each Frequency Band can be processed independently. Additionally, each
Frequency Band can be routed to the channel inputs of other tracks (see "Pro Multiband Dynamics and
Multiband Splitter Plug-In Sends" on page 163). Pro Multiband Dynamics uses an 8th-order Linkwitz-Riley
crossover network in order to split the audio signal into up to four separate Frequency Bands, ensuring that
the output signal can be perfectly reconstructed.
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The FFT display plots the real-time magnitude of the audio signal (y-axis) versus frequency (x-axis). The
frequency range is marked from 20 Hz to 20 kHz along the bottom of the display. The magnitude range,
marked on the right of the display, shows the real-time magnitude of the signal in dB. Gain processing
(reduction or expansion) is shown on the display as a purple fill. The height of the fill corresponds to the
amount of dynamics processing that is being applied to that band's signal.
Use the dynamics processing scale, marked on the left of the display, to measure the adjustment of Gain,
Depth, and Threshold controls. (See "Band Pane Controls and Indicators" on page 157 for detailed
information on these controls.)
To adjust the Gain control for a Frequency Band, do one of the following:
t Click the Gain slider and drag it left or right.
t Click the Gain label, type a value, and press Enter.
t Click the red square Gain control icon on the FFT display and drag it up or down.
Adjusting the Gain control
To adjust the Depth control for a Frequency Band, do one of the following:
t Click the Depth knob and drag it left or right, or up or down.
t Click the Depth numeric entry field, type a value, and press Enter.
t Click the purple triangle Depth control icon on the FFT display and drag it up or down.
Adjusting the Depth control
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To adjust the Threshold control for a Frequency Band, do one of the following:
t Click the Threshold knob and drag it left or right, or up or down.
t Click the Threshold numeric entry field, type a value, and press Enter.
t Click the Threshold control to the left of the Band meter and drag it up or down.
t Click the blue circle Threshold control icon on the FFT display and drag it up or down.
Adjusting the Threshold control
Crossover Frequency Controls

Use the Crossover Frequency controls (up to three are available depending on how many of the four
possible frequency bands are enabled) to set the frequency split between bands.
To adjust the frequency split between adjacent bands, do one of the following:
t Drag the Crossover Frequency control to the left or right.
t Click the Crossover Frequency numeric entry field, type a value, and press Enter.
Adjusting Crossover Frequency control
FFT Display Options

Pro Multiband Dynamics lets you choose to show the Input signal (pre-processing) or the Output signal
(post-processing) in the FFT display. It also lets you show or hide the Gain and Threshold controls in the FFT
display, as well as hide the FFT amplitude graph. This can be useful if you want to reduce visual clutter in
the plug-in window.
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FFT Display options
To hide (or show) the Gain controls on the FFT display:

t Click the Show / Hide Gain icon so that it is not lit (or click the Show / Hide Gain icon so that it is lit).
To hide (or show) the Threshold controls on the FFT display:
t Click the Show / Hide Threshold icon so that it is not lit (or click the Show / Hide Threshold icon so that
it is lit).
To show the Input signal in the FFT display:
t Click the FFT Input toggle so that it is lit.
To show the Output signal in the FFT display:
t Click the FFT Output toggle so that it is lit.
To hide the FFT amplitude graph:
t Click either the lit FFT Input toggle or the lit FFT Output toggle (whichever one is lit) so that it
becomes unlit.
Input and Output Gain Controls

Pro Multiband Dynamics provides Input Gain (pre-processing) and Output Gain (post-processing) controls.
Pro Multiband Dynamics lets you independently adjust the Input Gain from –36.0 dB to +36.0 dB and the
Output Gain from –36.0 dB to +12.0 dB. These controls are located in the upper-left corner of the Plug-In
window.
To adjust the Input Gain to or the Output Gain from Pro Multiband Dynamics, do one of the following:
t Drag the fader icon, or drag in the numeric entry field.
t Click the numeric entry field, type a value, and press Enter.
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Source Linking
The Linking selector lets you set how individual channels of a multichannel source signal are combined in
each of the separate detectors (by band). The source signal can be the audio being processed by the Pro
Multiband Dynamics, or it can be an external side-chain signal (see "Side-Chain Processing" on page 161).
The available Linking modes are: Stereo Pairs, All (w/LFE), All (no LFE), or Front/Rear.
Selecting linked source channels for triggering dynamics processing
When Stereo Pairs is selected on a greater-than-stereo multichannel track, the input signal for each
stereo pair affects only those same channels, and likewise mono channels are affected 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.
All (with LFE)
When All (w/LFE) mode is selected, the plug-in combines every channel into the same detector and
applies dynamics processing on each channel identically. For example, when any signal channel crosses
the threshold in a 5.1 surround format, all of the channels will be processed equally.
All (No LFE)
When All (no LFE) 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.
Front/Rear
For LCRS or greater channel formats, when Front/Rear is selected, dynamics processing is applied
based on front channel inputs (LCR) and surround channel inputs (S) independently. For .1 formats, the LFE
Mini Multichannel Gain Reduction and Output Meters

Pro Multiband Dynamics provides mini Gain Reduction and Peak Meters for every possible channel output.
These meters are located in the upper-right corner of the plug-in window. The eight channels of Gain
Reduction Meters and the eight channels of Output Meters are always shown regardless of the channel
width of the track on which Pro Multiband Dynamics is inserted. However, only active channels will show
meter activity.
The channel ordering of the meters updates to match the selected channel width. For example, stereo
shows on the first two meters from left to right, but 5.1 shows L, C, R, Ls, and Rs from left to right with the
LFE always shown on the right-most meter.
The color coding under the Gain Reduction corresponds to the Channel Link mode: Channels with the same
color are combined in the same detector (see "Source Linking" above).
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Gain Reduction and Output meters
Band Pane Controls and Indicators

Each of the four bands of Pro Multiband Dynamics provide a control pane with the same controls and
indicators.
Adjusting Controls in a Band Pane

Click the button controls to toggle them on or off. When toggled on, the text in the button lights.
Turn the knob controls by dragging up and down, or left and right.
Adjust the slider controls by dragging from left to right.
Click in a numeric field and type a number. Press Enter to commit the entered value. The Gain, Slope,
Attack, and Release labels convert to numeric fields when you move the cursor over either the slider or the
label. The labels for Gain, Slope, Attack, and Release can be edited by clicking the numeric field.
Band Pane Buttons

At the top of each Band pane are a set of buttons that let you enable / disable the band, solo the audio
passing through the band, bypass dynamics processing on the band, link band controls, and toggle the
band dynamics processing between being a compressor or an expander.
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Enable / Disable Band
The Enable / Disable Band button displays the band number. Click it to toggle the band on (lit) or off (unlit).
Disabling a band results in the adjacent band taking over the frequency range occupied by the disabled
band, and adjusts the number of Crossover Frequency controls on the graph display.
Solo
The Solo button lets you solo or unsolo the band.
Link
When the Link button is enabled on more than one band, all controls are linked between these bands.
Linked bands retain offsets between continuous controls.
g To temporarily link the controls of all 4 bands, Shift-drag a control.

b To temporarily unlink and adjust a control independently of all other linked bands, Control-click (Mac)
or Start-click (Windows) and drag the control.
Auto
The Auto button lets you enable Automatic Gain Control. When Automatic Gain Control is enabled, the
Gain setting changes when you change the Depth setting. However, when you change Gain, the Depth
setting remains in a fixed relation to the Gain setting.
Flip
When Flip mode is enabled (lit), the operation of the band changes so that dynamics are adjusted when
the input signal is below the threshold. By default, each band adjusts the dynamics when the input signal
goes above the threshold. However, in Flip mode, the band functions as either a classic Downward
Compressor or an Upward Expander, depending on the Depth control setting.
For example, if you enable Flip on all the bands and set the Depth controls to negative values, Pro
Multiband Dynamics becomes a decent broadband noise reduction plug-in. If you have a signal source with
a lot of high-end hiss, enable Flip mode on just the high-band and noise-reduce that band with downward
expansion (while keeping the other bands as downward compressors).
To bring out the fine low-level details of a performance, enable Flip mode on the middle two bands and set
the Depth controls to positive values. This increases the volume of subtle nuances while ensuring that the
tone of anything above the threshold is unaffected.
Since Flip mode significantly changes the characteristics of the dynamics processing, the color of the
Threshold and Depth controls change from blue (Flip mode disabled) to orange (Flip mode enabled).
g For parallel processing workflows, you can use Flip mode to get results similar to parallel processing
while avoiding phase issues or having to set up parallel compression routing. Pro Multiband Dynamics
is not recommended for parallel dynamics processing as it can result in phase issues between the
processed signal and the unprocessed signal. Like many multiband compressors, Pro Multiband
Dynamics uses minimum-phase filters to ensure efficient low-latency processing. However, this results
in a signal that does not phase-align with a parallel unprocessed signal.
Bypass
When Bypass is enabled, the input signal passes through the band without any dynamics processing.
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Band Pane Meters
Each Band provides independent Level meters with Peak Hold indicators, as well as Gain Reduction meters
with Gain Reduction / Expansion Peak indicators. Additionally, there is a Threshold control that lets you set
the Threshold against the level meter.
Band Pane meters
Gain Reduction / Expansion Peak Indicator
Each band provides a numerical indicator for the peak gain reduction or expansion value in decibels.
Gain Reduction Meter
The Gain Reduction Meter displays the amount of gain reduction applied to the band.
Peak Indicators
The Peak Hold indicator appears as a thin line in the Level Meter. The Peak Gain Reduction indicator
appears as a thin line in the Gain Reduction Meter. These provide highly accurate visual metering
correlation with the frequency band’s audio signal.
Threshold Control
The Threshold control lets you adjust the Threshold setting against the Level Meter.
Level Meter
The Level Meter indicates the band output level based on the following color coding:
l Dark Blue
l Light Blue
l Yellow
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Dynamics Controls
Threshold
The Threshold control sets the level that an input signal must exceed to trigger dynamics processing (from
–60 dB to 0 dB). A band will be compressed if its level exceeds this setting. If the signal level falls below this
value, no processing will occur on this band. You can also adjust the Threshold setting using the blue circle
Threshold control on the FFT display or the Threshold control to the left of the band’s level meter.
Depth
The Depth control sets the maximum amount of gain reduction or expansion applied to the input signal of
the band (from –24 dB to +24 dB). For example, if you set Depth to –10 dB, no more than 10 dB of gain
reduction is applied to the incoming signal. You can also adjust the Depth setting using the purple triangle
Depth control on the FFT display.
Gain
The Gain control lets you boost or attenuate the output gain of the frequency band (from –24 dB to
+24 dB). You can also adjust the Gain using the Red Square gain control icon on the FFT display.
Slope
The Slope control lets you adjust the dynamic curve for compression and expansion (from 0% to 100%).
Slope is effectively a combined control for adjusting the Ratio and the Knee of the dynamics processor.
Lower settings offer a lower ratio and a softer knee. Higher settings increase the ratio and provide a sharper
knee.
When Slope is set to 0%, the effective Ratio is approximately 1:1.5. When Slope is set to 100%, the Ratio is
1:20.
At the same time, the knee ranges from about 1/2 of Depth in dB (min slope) to 0 dB (max slope).
Attack
The Attack control sets the attack time, or the rate at which gain is reduced after the input signal level
crosses the threshold (from 100.0 microseconds to 1.0 second).
The smaller the value, the faster the attack. The faster the attack, the more rapidly gain change is applied
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.
g The actual compression attack time is also dependent on the selected Detection mode. Each mode has
its own attack and release times that are calculated in advance of compression processing. If a slower
Detection mode is selected (such as AVG), the fastest possible actual attack time for compression can
only be about 20 ms. The selected Detection mode similarly affects the compressor release time.
Release
The Release control sets the length of time it takes for compression to be fully deactivated after the input
signal drops below the threshold (from 1.0 millisecond to 10.0 seconds).
without recovering.
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Detection Modes
Pro Multiband Dynamics provides several different detection options for determining how the compressor
responds to the input signal.
To change the detection mode for the compressor:

t Click a detection mode from the options available at the bottom of the Band Controls pane.
Detection mode options (Smart mode selected)
Smart
Select the Smart option for tracks with diverse input signals, or if you are simply not sure what detector
works best with the given audio material. The Smart option analyzes the incoming signal and interpolates
between the different detection modes as needed. This lets you apply a lot of compression without
distortion or pumping.
RMS
Select the RMS option to apply processing according to the detected RMS (Root Mean Square) amplitude of
the input signal. The RMS option is similar to the Average option, but with a faster release time.
Average
Select the Average option to apply processing according to the detected average amplitude of the input
signal.
Peak
Select the Peak option to apply processing according to the detected peak amplitude of the input signal.
This mode provides the fastest detector response.
Side-Chain Processing
Dynamics processors operate by changing the level of the main input signal according to what is detected
on a separate “key input” or “side chain.” Normally, the main input and the key input are fed from the same
source signal, but some dynamics plug-ins can have the side chain fed from a separate, external signal.
(such as a separate reference track or audio source) instead of the input signal.
On Pro Multiband Dynamics, the external side chain is processed through the same crossover network as
the main signal, so the side chain of each band will contain only that band’s frequency range.
To use a key input for external side-chain processing on any frequency band:
1. Use a send bus or output bus on the track with the audio that you want to use for external side-chain
processing
2. Click the Key Input selector and select the input or bus carrying the audio you want to use to trigger
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Bus 1 selected as the key input for side-chain processing
3. Enable side-chain processing on any of the frequency bands in Pro Multiband Dynamics.
4. Begin playback. The plug-in uses the input or bus that you chose as a key input to trigger its effect.
Side-Chain section
Side Chain Signal Level
The Side Chain Signal Level meter displays the level of the incoming side chain signal.
Side Chain Listen
Listen mode lets you hear the input signal of the side chain to the frequency band.
To enable (or disable) Listen mode on the side chain:

t Click the Listen button so that it is highlighted. To disable it, click the button again so that it is not
highlighted.
Side Chain Enable
Pro Multiband Dynamics lets you enable sidechain processing on a band-by-band basis.
To enable (or disable) side-chain processing for a given frequency band:

t Click the Side Chain Enable button so that it is highlighted. To disable it, click the button again so
that it is not highlighted.
Multiband Splitter Plug-In

Included with Pro Multiband Dynamics is a separate Multiband Splitter plug-in that lets you send audio by
frequency band to other tracks using plug-in sends. This lets you split the audio signal by frequency bands
for additional mixing and effects processing, but without any dynamics processing.
c The Multiband Splitter plug-in is not recommended for parallel effects processing as it can result in
phase issues between the processed signal and the unprocessed signal.
g VENUE and Media Composer do not support Multiband Splitter.
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Enable / Disable Band
The Enable / Disable Band button shows the band number. Click it to toggle the Band on (lit) or off (unlit).
Crossover Frequency Controls

Use the Crossover Frequency controls (up to three are available depending on how many of the four
possible frequency bands are enabled) to set the frequency split between bands.
To adjust the frequency split between adjacent bands:

t Drag the Crossover Frequency control to the left or right.
Pro Multiband Dynamics and Multiband Splitter Plug-In Sends
Both Pro Multiband Dynamics and Multiband Splitter plug-ins let you output signal on a band-by-band
basis for routing to Auxiliary Input tracks or audio tracks for additional mixing and processing.
g VENUE and Media Composer do not support the auxiliary output signal feature of Pro Multiband
Dynamics.
To route the audio signal output of individual frequency bands from Pro Multiband Dynamics or Multiband
Splitter:
1. Insert Pro Multiband Dynamics or Multiband Splitter on a track.

2. Adjust the Crossover Frequency controls.
3. Create an Auxiliary Input (or audio) track.
4. From the Auxiliary Input (or audio) track input selector, select a plug-in band.
Selecting the Low Band output from an instance of the Pro Multiband Dynamics plug-in as the input on an Auxiliary Input track
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Purple Audio MC77
Purple MC77 is a dynamics processing plug-in that is available in DSP, Native, and AudioSuite formats.
Purple MC77 supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Purple MC77 operates as a mono, multi-mono, or stereo plug-in.
The Purple Audio MC77 is a spot-on digital replica of Andrew Roberts’ 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.
Purple Audio MC77 Controls

Purple Audio MC77 has controls identical in name to those of the BF76, and which function similarly. For
more information, see "BF76" on page 64.
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Smack!
Smack! is a dynamics processing plug-in that is available in DSP, Native, and AudioSuite formats.
Smack! supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Smack! supports mono, stereo, and greater-than-stereo multichannel formats up to 7.1.
g Greater-than-stereo formats are only available with Pro Tools Ultimate and Studio. Note also that the
LFE channel is not processed in multichannel versions of Smack!
g 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.
The Smack! Compressor / Limiter plug-in has the following features.

l Three modes of compression:
l 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.
l Warm mode is based on Norm mode, but has release characteristics more like those of electro-optical
limiters.
l 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.
l “Key Input” side-chain processing, which lets you trigger compression using the dynamics of another
signal.
l Side-Chain EQ filter, which lets you tailor the compression to be frequency-sensitive.
l High Pass filter, which lets you remove “thumps” or “pops” from your audio.
l Distortion control, which lets you add different types of subtle harmonic distortion to the output
signal.
Smack! Controls and Meters

Smack! includes controls for multiple compression modes and a VU meter.
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Smack! Compression Mode Buttons
Smack! has three modes of compression: Norm (Normal), Warm, and Opto. Use the corresponding button to
select a mode.
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.
g 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.
g Setting the Input and Output controls to 5 is equal to unity gain at a compression ratio of 1:1.
Smack! Attack Control
In Norm and Warm modes, Attack controls the rate at which gain is reduced after the input signal crosses
the threshold.
g This control is not available 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 µs to 80
milliseconds.
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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 an input level
that is 2 dB above the threshold will be attenuated, resulting in an output level that is 1 dB over the
threshold.

g 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.
As you increase the Ratio control, Smack! goes from applying “soft-knee” compression to “hard-knee”
compression, as follows:
approaches the threshold. This creates smoother compression.
l In hard-knee compression, compression begins when the input signal exceeds the threshold. This can
sound abrupt, and is ideal for limiting or de-essing.
Smack! compression ratios range from subtle compression to hard limiting. At ratios of 10:1 and higher,
Smack! functions as a limiter. Selecting the Smack! setting lowers the threshold slightly and applies hard
limiting, which keeps the output level constant regardless of the input level. (This setting can also be used
for extreme compression effects.)
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.

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.
When you apply Smack! to stereo or multichannel tracks, the Output control determines master output
levels for all channels.
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.
g Setting the Input and Output controls to 5 is equal to unity gain at a compression ratio of 1:1.
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.
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g See "Using the Smack! Side-Chain Input " on page 170 for more information on using the Side-Chain
EQ on a Key Input.
The Side-Chain EQ filter has the following settings:
High Pass
Makes the compressor's detector less sensitive to low frequencies in the input signal or Key Input by rolling
off at a rate of 6 dB per octave. For example, you might use this setting on a mix to prevent a bass guitar or
bass drum from causing too much gain reduction.
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.
Band-Emphasis Side-Chain EQ
Combined
Enables the High Pass and peak settings simultaneously to make the compressor's detector more sensitive
to high frequencies and less sensitive to low frequencies.
Combined Side-Chain EQ
Off
Disables the Side-Chain EQ control.
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Smack! Distortion Control
When enabled, Distortion adds subtle second-order and third-order harmonic distortion to the output
signal.
l Odd harmonics produce waveforms that are more square-shaped and are often described as “harsh”
sounding.
l Even harmonics produce waveforms with more rounded edges and are often described as “smooth”
sounding.
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) harmonics to the distortion.
O+E
Applies an equal blend of odd and even harmonic distortion.
g 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.
Smack! VU Meter
The VU meter displays the amount of input level, output level, or gain reduction from compression,
depending on the current Meter Mode button setting. It is calibrated to a reference level of –14 dBFS = 0 VU.
Meter Mode Button and Clip Indicators
The Meter Mode button toggles between displaying three display modes, as follows:
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l In
Displays the input signal level, referenced to –14 dBFS = 0 VU.
l Out
Displays the output signal gain, referenced to –14 dBFS = 0 VU.
l GR
Displays the amount of gain reduction applied by the compressor.
The Input and Output meters indicate input and output signal levels in dBFS (dB relative to full scale or
maximum output).
The Internal Clipping indicator (labeled “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! 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.
g 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 167 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.
g 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.
g 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 167 for more information).
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4 Pitch and Time Shift Plug-Ins
Pitch and Time Shift plug-ins let you change the pitch (note) or time (length) of material. You can alter the
pitch with or without changing its length, and conversely adjust the length with or without altering the
pitch.

l "Pitch II" below
l "Pitch Shift Legacy" on page 174
l "Time Shift" on page 176
l "Vari-Fi" on page 184
l "X-Form" on page 185
Pitch II
Pitch II is a pitch-shifting plug-in that is available in DSP, Native, and AudioSuite formats. Pitch II is designed
for a variety of audio production applications, ranging from pitch correction of musical material to sound
design.
Pitch II supports 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz sample rates for all plug-in formats. Pitch II also
supports 176.4 kHz, and 192 kHz sample rates for Native and AudioSuite plug-in formats.
Pitch II is available in mono, mono-to-stereo, and stereo channel formats.
Pitch processing typically uses the technique of varying sample playback rate to achieve pitch
transposition. Changing audio sample playback rate results in the digital equivalent of vari-speeding with
tape. This is usually unsatisfactory since it changes the overall duration of the material.
Pitch transposition with Pitch II involves a much more complex technique. Pitch II digitally re-aligns portions
of the re-sampled 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
duration of the original, unprocessed signal.
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Pitch II Controls
Input and Transient Controls
Input
The Input control lets you attenuate the gain of the input signal from –6.0 dB to 0 dB to prevent clipping in
the pitch shift algorithm.
Input Polarity
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. The stereo
version of Pitch II provides adjacent left and right channel Input Polarity buttons.
Range
The Range selector lets you adjust the range of frequencies (Low, Mid, High, Wide) used for pitch
detection. For most program material, the Wide setting should work well for pitch detection and
transposition. If you encounter undesirable frequency artifacts with pitch transposition, experiment with
other settings. Set this parameter to match the expected frequency content of source material. For
example, when working with a bass part, set Range to Low. When pitch-shifting audio from material similar
to a soprano vocalist or a violin, set Range to High.
Clip Indicator
The Clip indicator shows 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.
Level Indicator
The Level indicator shows the presence of an input signal.
Threshold
Pitch II detects and responds to transients in the incoming audio signal to prevent “smearing” of sharp
attacks (such as drum hits or vocal plosives). The Threshold control (–40 dB to 0 dB, and Off) determines
how strong a transient needs to be in order to be recognized by the pitch detection algorithm. It you
encounter undesirable frequency artifacts in the middle of long held notes, try raising the Threshold
setting. If audio transients are obscured in the transposed signal, try lowering the Threshold setting.
Window
Pitch II changes the pitch by splitting the incoming audio up into small grains (6.0 ms to 42 ms), re-sampling
those grains, and adding those grains back together. The Window control determines the size of the grains.
If you are working with long pad-like material such as legato strings, then increasing the Window size may
improve audio quality; having too short of a Window in this situation may result in robotic or buzzy
sounding audio. If you are pitch-shifting material with sharp transients, such as a drum part, reducing the
Window size improves transient response; having too long of a Window in this situation may make the audio
sound smeared and choppy.
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Follow
Enable Follow to match the overall dynamic envelope of the source audio. If pitch-shifted audio does not
match the same decay sound as the original, turning Follow on may improve the sound. In most cases,
however, leaving Follow off should sound just fine.
Pitch Shift Controls
Keyboard (Relative Pitch Transposition)
Click any note on the keyboard to set a relative pitch transposition value that will be applied to the audio
signal. The “C” key in the middle of the keyboard represents the original pitch of the incoming signal: if the
C in the middle of the keyboard is selected, there is no pitch transposition. Click any other key to transpose
the pitch of the incoming signal by the interval difference between middle C and the selected key. For
example, if the E-flat key above middle C is selected, Pitch II transposes the incoming signal up a minor
third (or 3 semitones). The Coarse control and the Keyboard control are linked.
Coarse
This control adjusts the pitch of a signal in semitones over a two octave range. Pitch changes are indicated
in number of semitones.
Fine
This control controls the pitch of a signal in cents (hundredths of a semitone) over a 100 cent range. The
range of this control is –50 to +50 cents.
Ratio
The Ratio control lets you set the transposition between the pitch of the incoming signal and the selected
transposition value as a percentage. The Ratio setting is linked with the Coarse and Fine settings.
Link (Stereo Only)
Enable the Link option to link the controls for the left and right channels.
Show/Hide Panel
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Click the Show / Hide triangle in the upper left corner of the Pitch Shift panel to show or hide the panel.
Hiding the Pitch Shift panel can be useful for conserving screen space.
Effects Controls
Delay
The Delay control lets you set the delay time between the original signal and the pitch-shifted signal. It has
a maximum setting of 1000 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 rise in pitch.
Mix
The Mix control lets you adjust 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 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.
Feedback
The Feedback control lets you set the amount and type of feedback (positive or negative) applied from the
output of the Delay effect 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.
LPF
The LPF (Low Pass Filter) control lets you set the frequency under which audio signal is passed. The control
can be set between 10 Hz and 22.05 kHz.
Pitch Shift Legacy

Pitch Shift Legacy is a pitch-processing AudioSuite plug-in.
Pitch Shift Legacy 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.
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Pitch Shift Legacy 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).
Time Correction
Disabling this option has the effect of “permanently varispeeding” your audio file. The duration of the
processed audio 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.
Ratio
Adjusts the amount of transposition (pitch change). Moving the slider to the right raises the pitch of the
processed file, while moving the slider to the left decreases its pitch.
Crossfade
Use this to manually adjust crossfade length in milliseconds to optimize performance of Pitch Shift Legacy
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 crossfades. Long crossfade times may over-smooth a signal and
its transients. This is may not be desirable on drums and other material with sharp transients.
Use the Crossfade slider to adjust and optimize crossfade times. For audio material with sharper attack
transients, use smaller crossfade times. For audio material with softer attack transients, use longer
crossfade times.
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Min Pitch
Sets the lowest pitch used in the audio you want to process. The control has a range of 40 Hz to 1000 Hz.
Use it to focus the pitch shift process according to the audio’s spectral shape.
Use lower values when processing lower frequency audio material. Use higher values when processing
higher frequency audio material.
Accuracy
Sets the processing resources allocated to audio quality (Sound) or timing (Rhythm). Set the slider toward
Sound for better audio quality and fewer audio artifacts. Set the slider toward Rhythm for a more consistent
tempo.
Reference Pitch
Activates a sine wave-based pitch generator that you can use as an audible reference when previewing
pitch-shifted audio material.
Note
Adjusts the frequency of the reference tone in semitones (half steps).
Detune
Provides finer adjustment of the frequency of the reference tone in cents (100ths of a semitone).
Gain
Adjusts the volume of the reference tone in dB.
To use the 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 set the reference tone to the desired target pitch for pitch
shifting. Adjust the Gain setting to change the relative volume of the reference tone.
4. Select the audio material to be pitch shifted.
5. Adjust the Coarse and Fine controls to match the pitch of the audio playback to the reference
pitch.
6. Disable preview and click Process to pitch shift the selection.
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 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.
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Time Shift Controls
Time Shift controls are organized in the following four sections:
Audio
Use the controls in the Audio section to select the most appropriate time compression and expansion
algorithm (mode) for the type of material you want to process, and to attenuate the gain of the processed
audio to aid clipping.
Time
Use the controls in the Time section to specify the amount of time compression or expansion you want to
apply.
Formant or Transient
Use the controls in the Formant or Transient section to adjust either the amount of formant shift or the
transient detection, 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 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, Audio section
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Mode
The Audio Mode pop-up menu determines the following types of TCE and pitch shift algorithm for
processing audio:
l Monophonic
Select Monophonic for processing monophonic sounds (such as a vocal melody).
l Polyphonic
Select Polyphonic for processing complex sounds (such as a multi-part musical selection).
l Rhythmic
Select Rhythmic for processing percussive 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.
l Varispeed
Select Varispeed to link time and pitch change for tape-like pitch and speed change effects, and post
production workflows.
Range
The Audio Range pop-up menu determines the following frequency ranges for analysis:
l Low
For low-range material, such as a bass guitar, select Low.
l 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.
l High
For material with a high fundamental frequency such as female vocals, select high.
l 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.
g The range pop-up menu is unavailable in Rhythmic mode and Varispeed mode.
Gain
The Audio Gain control attenuates the input level to avoid clipping. Adjust the Gain control from 0.0 dB to –
6.0 dB to avoid clipping in the processed signal.
Clip Indicator
The Clip indicator indicates clipping in the processed signal. When using time compression or pitch shifting
above the original pitch, it is possible for clipping to occur. The Clip indicator lights when the processed
signal is clipping. If the processed signal clips, undo the AudioSuite process and attenuate the input gain
using the Gain control. Then, re-process the selection.
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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, 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 and type
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 and type values. The Processed Tempo value updates automatically
when adjusting the Time control.
Unit
Select a timebase for the Original and Processed time fields: Bars|Beats, Min:Sec, Timecode, Feet+Frames,
or Samples.
Speed
Displays the target time compression or expansion as a percentage of the original. Adjust the Time control
or click the Speed field and type a 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 184).
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
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characteristics of a female voice.
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 a value.
Time Shift Formant section
About Formants
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 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 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 Transient section with Rhythmic selected as the Audio Type
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Follow
The follow button enables an envelope follower that simulates the original acoustics of the audio being
stretched. Click the Follow button to enable or disable envelope following. Follow is only available when
Polyphonic is selected as the Audio Type.
Threshold
The Threshold controls sets the transient 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 a 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 a 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, Pitch section
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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 a value.
Shift
Displays the pitch shift amount as a percentage. 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 a value. The default value is 100% (no
pitch shift).
AudioSuite Input Modes and Time Shift

Time Shift supports the Pro Tools AudioSuite Input Mode selector for use on mono or multi-input processing.
Mono Mode
Processes each audio clip as a mono file with no phase coherency maintained with any other
simultaneously selected clips.
Multi-Input Mode
Processes up to 48 input channels and maintains phase coherency within those selected channels.
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.
Time Shift as AudioSuite TCE Plug-In Preference

The Time Shift 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
g See 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 a preset 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.
g Normalizing a selection before using Time Shift may produce better results.
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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.
6. If using Polyphonic or Rhythmic mode, adjust the Transient controls.
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 set the 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. Click Render.
Changing the Pitch Using Time Shift
To change the pitch of a selected audio clip:

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 transposing the pitch of the selection up, attenuate the Gain control as necessary.
5. If using Monophonic mode, adjust the Formant Shift control.
6. If using Polyphonic or Rhythmic mode, adjust the Transient controls.
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 set the 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:

2. Select Varispeed from the Audio Mode pop-up menu.
3. Adjust either the Time Shift or Pitch Shift control to set the amount of time and pitch change in terms
of a percentage of the original.
4. Click Render.
g Using the Monophonic, Polyphonic, or Rhythmic modes, you can adjust both the Time Shift and Pitch
Shift controls independently before processing.
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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.
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
Vari-Fi
Vari-Fi is an AudioSuite 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.
Features include:
l Speed up from a complete stop to normal speed
l Slow down to a complete stop from normal speed
Vari-Fi Controls
Change Controls
Slow Down
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.
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.
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Selection Controls
The Selection setting determines the duration of the rendered clip in relation to the processing.
Fit To
When the Fit To 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).
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.
Extend
When the Extend 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.
Fades 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 fade-out 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.
X-Form
X-Form is an 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.
g Normalizing a selection before using X-Form may produce better results.
185
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 for high quality time compression
or expansion.
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.
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X-Form, Audio section
Type
The Audio Type determines the type of TCE and pitch shift algorithm for processing audio: Polyphonic,
Monophonic, or Poly (Faster).
l Polyphonic
Use for processing complex sounds (such as a multipart musical selection).
g When previewing Polyphonic, Poly (Faster) is used for faster previewing. However, when you
process the audio selection, the high-quality Polyphonic setting is used.
l Monophonic
Use for processing monophonic sounds (such as a vocal melody).
l Poly (Faster)
Use for faster previewing and processing, 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.
X-Form Time Section Controls

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.
187
Original
The Original column 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
The Processed column 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 and type values. These values update automatically when adjusting the Time control.
Tempo
The Tempo row displays the Original Tempo and Processed Tempo in beats per minute (bpm). You can click
the Original Tempo and Processed Tempo fields and type values. The Processed Tempo value updates
automatically when adjusting the Time control.
Unit
Select a timebase for the Original and Processed time fields: Bars|Beats, Min:Sec, Timecode, Feet+Frames,
or Samples.
Shift
The Shift setting displays the target time compression or expansion as a percentage of the original. Adjust
the Time control or click the Shift field and type a 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 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 192).
2x, 4x, and 8x Range Buttons
The 2x, 4x, and 8x Range buttons set the possible range for the Time Shift, Pitch Shift, and Formant Shift
controls.
188
l 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).
l 4x
l 8x
g 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.
Sensitivity
The Sensitivity setting controls how X-Form determines and interprets transients from the original audio.
Part of X-Form’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.
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
The Window setting determines 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 a value. Window is only
available when Monophonic is selected as the Audio Type.
189
Try larger window sizes for low frequency sounds or sounds that do not have many transients. Try smaller
window sizes for tuned drums and percussion. However, the default of 25 milliseconds should work well for
most material.
X-Form Pitch Section Controls

The Pitch section provides controls for pitch shifting the selected audio. Use the Pitch control to transpose
the pitch from as much as –36.00 semitones (–3 octaves) to +36.00 semitones (+3 octaves), with fine
resolution in cents, depending on which Range button is enabled (2x, 4x, or 8x). X-Form also lets you
transpose the formant shape independently of the fundamental frequency.
Transpose
The Transpose setting displays the transposition amount in semitones. You can transpose pitch by 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. Adjust the Pitch control or click the Transpose field and type a value.
Shift
The Shift setting displays the pitch shift amount as a percentage. Pitch can be shifted 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 a value.
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.
190
To enable or disable formant shifting:
t 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 a value.
X-Form AudioSuite Input Modes

X-Form supports the Pro Tools AudioSuite Input Mode selector for use on mono or multi-input processing.
Mono Mode
Processes each audio clip as a mono file with no phase coherency maintained with any other
simultaneously selected clips.
Multi-Input Mode
Processes up to 48 input channels 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 TCE Plug-In Preference

The high quality time compression and expansion algorithms of X-Form can be used with the Pro Tools TCE
Trim tool.
TCE Plug-In option in Processing Preferences page
c 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%.
n 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 Avid X-Form.
4. Select a preset from the Default Settings pop-up menu.
5. Click OK.
Processing Audio Using X-Form

X-Form lets you change the time and pitch of selected audio independently or concurrently.
g You can adjust both the Time Shift and Pitch Shift controls independently before processing.
191
To change the time 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 compressing the duration of the selection, attenuate the Gain control as necessary.
4. Adjust the Transient controls.
5. Enable a Range button (2x, 4x, or 8x) to set the possible range for time change.
6. Adjust the Time Shift control to set the 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.
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.
5. Enable a Range button (2x, 4x, or 8x) to set the possible range for pitch change.
6. Adjust the Pitch Shift control to set the amount of pitch change. Pitch change is measured in
semitones (and cents) or as a percentage of the original pitch.
7. If you want to enable formant processing, 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.
g Use the X-Form Plug-In Settings for the corresponding post-production task.
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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5 Reverb Plug-Ins
Reverb provides a sense of room acoustics. Reverb can make one track, groups of tracks, or your whole mix
sound like it is in a big concert hall, an intimate room, a narrow hallway, or other acoustic spaces.

l "D-Verb" below
l "Eleven Effects Reverb Plug-ins" on page 195
l "Reverb One" on page 196
l "ReVibe II" on page 205
l "Space" on page 221
D-Verb
D-Verb is a studio-quality reverb plug-in that is available in DSP, Native, and AudioSuite formats.
D-Verb supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
D-Verb operates as a mono, multi-mono, mono-to-stereo, or stereo plug-in.
D-Verb Controls
D-Verb provides a variety of controls for adjusting plug-in parameters.
Input Level Meters

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.
Output Level Meter

Output meters indicate the output levels of the processed signal.
Gain and Input Level Controls

D-Verb provides a Gain control above the Input Level meter to let you adjust the input gain.
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%.
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Algorithm Control
This control selects one of seven reverb algorithms: Hall, Church, Plate, Room 1, Room 2, Ambience, or Non-
linear. Selecting an algorithm changes the preset provided for it. Switching the Size setting changes
characteristics of the algorithm that are not altered by adjusting the decay time and other user-adjustable
controls. Each of the seven algorithms has a distinctly different character:
Hall
A good general purpose concert hall with a natural character. It is useful over a large range of size and
decay times and with a wide range of program material. Setting Decay to its maximum value will produce
infinite reverberation.
Church
A dense, diffuse space simulating a church or cathedral with a long decay time, high diffusion, and some
pre-delay.
Plate
Simulates the acoustic character of a metal plate-based reverb. This type of reverb typically has high initial
diffusion and a relatively bright sound, making it particularly good for certain percussive signals and vocal
processing. Plate reverb has the general effect of thickening the initial sound itself.
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
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.
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Decay Control
Decay controls the rate at which the reverb decays after the original direct signal stops. The value of the
Decay setting is affected by the Size and Algorithm controls. This control can be set to infinity on most
algorithms for infinite reverb times.
Pre-Delay Control
Pre-Delay determines the amount of time that elapses between the original audio event and the onset of
reverberation. Under natural conditions, the amount of pre-delay depends on the size and construction of
the acoustic space, and the relative position of the sound source and the listener. Pre-Delay attempts to
duplicate this phenomenon and is used to create a sense of distance and volume within an acoustic space.
Long Pre-Delay settings place the reverberant field behind rather than on top of the original audio signal.
Hi Frequency Cut
Hi Frequency Cut controls the decay characteristic of the high frequency components of the reverb. It acts
in conjunction with the Low Pass Filter control to create the overall high frequency contour of the reverb.
When set relatively low, high frequencies decay more quickly than low frequencies, simulating the effect of
air absorption in a hall. The maximum value of this control is Off (which effectively means bypass).
Low Pass Filter

Low Pass Filter controls the overall high frequency content of the reverb by setting the frequency above
which a 6 dB per octave filter attenuates the processed signal. The maximum value of this control is Off
(which effectively means bypass).
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.
Eleven Effects Reverb Plug-ins

Black Spring
Black Spring was inspired by a classic outboard tube-driven spring reverb. It can add a “surfy” twang and
dark, warm ambiance to the audio signal.
Black Spring is available as a Mono or Stereo plug-in.
On/Bypass
The On/Bypass toggle lets you bypass the Spring Reverb effect.
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Mix
The Mix knob controls the blend between dry and reverbed signal.
Decay
The Decay knob controls the length of the reverb’s decay.
Tone
The Tone knob applies a high-cut EQ, making the reverb tone darker.
Studio Reverb
A smooth, clean digital reverb, Studio Reverb is based on the popular Reverb One Pro Tools plug-in from
Avid.
Studio Reverb is available as a Mono, Mono to Stereo, or Stereo plug-in.
On
The On button lets you toggle the Studio Reverb effect on or off (bypassed).
Type
The Type selector lets you choose from a variety of different reverb types.
Decay
The Decay control lets you set the decay time for the selected reverb (from 0.0 to 10.0 seconds).
Pre-Delay
The Pre-Delay control lets you set the amount of pre-delay, which changes the time relationship between
the dry signal and the reverbed signal. As this setting is turned higher, the apparent size of the synthesized
reverb “room” grows larger.
Tone
The Tone knob applies a high-cut EQ, making the reverb tone darker.
Mix
The Mix knob controls the blend between dry and reverbed signal.
Reverb One
Reverb One is a world-class reverb processing plug-in that 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 is available in DSP, Native, and AudioSuite formats.
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Reverb One supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Reverb One operates as a mono, multi-mono, mono-to-stereo, or stereo plug-in.
Reverb One features include:

l Editable Reverb EQ graph
l Editable Reverb Color graph
l Reverb Contour graph
l Dynamic control of reverb decay
l Chorusing
l Early reflection presets
l Extensive library of reverb presets
About Reverb
Digital reverberation processing can simulate the complex natural reflections and echoes that occur after a
sound has been produced, imparting a sense of space and depth—the signature of an acoustic
environment. When you use a reverberation plug-in such as Reverb One, you are artificially creating a
sound space with a specific acoustic character.
This character can be melded with audio material, with the end result being an adjustable mix of the
original dry source and the reverberant wet signal. Reverberation can take relatively lifeless mono source
material and create a stereo acoustic environment that gives the source a perceived weight and depth in a
mix.
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Creating Unique Sounds
In addition, digital signal processing can be used creatively to produce reverberation characteristics that
do not exist in nature. There are no rules that need to be followed to produce interesting treatments.
Experimentation can often produce striking new sounds.
Acoustic Environments
When you hear live sound in an acoustic environment, you generally hear much more than just the direct
sound from the source. In fact, sound in an anechoic chamber, devoid of an acoustic space’s character,
can sound harsh and unnatural.
Each real-world acoustical environment, from a closet to a cathedral, has its own unique acoustical
character or sonic signature. When the reflections and reverberation produced by a space combine with the
source sound, we say that the space is excited by the source. Depending on the acoustic environment, this
could produce the warm sonic characteristics we associate with reverberation, or it could produce echoes
or other unusual sonic characteristics.
Reverb Character
The character of a reverberation depends on a number of things. These include proximity to the sound
source, the shape of the space, the absorptivity of the construction material, and the position of the
listener.
Reflected Sound
In a typical concert hall, sound reaches the listener shortly after it is produced. The original direct sound is
followed by reflections from the ceiling or walls. Reflections that arrive within 50 to 80 milliseconds of the
direct sound are called early reflections. Subsequent reflections are called late reverberation. Early
reflections provide a sense of depth and strengthen the perception of loudness and clarity. The delay time
between the arrival of the direct sound and the beginning of early reflections is called the pre-delay.
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 you can create extremely natural-
sounding reverb effects.
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 202.
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.
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100% Wet
Toggles the Wet/Dry control between 100% wet and the current setting.
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.
g To hear examples of decay dynamics, load one of the Dynamics presets using the Plug-In Librarian
menu.
Decay Ratio
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.
For example, if Decay Ratio is set to 4, the reverb time is increased by a factor of 4 when the signal is above
the threshold level. If the ratio is 0.25, reverb time is increased by a factor of 4 when the signal is below the
Threshold level.
Threshold
Sets the input level above or below which reverb decay time will be modified.
Chorus Controls
The Chorus section has controls for setting the depth and rate of chorusing applied to a reverb tail.
Chorusing thickens and animates sounds by adding a delayed, pitch-modulated copy of an audio signal to
itself.
Chorusing produces a more ethereal or spacey reverb character. It is often used for creative effect rather
than to simulate a realistic acoustic environment.
g 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.
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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.
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.
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.
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.
l Size
Determines the rate of diffusion buildup and acts as a master control for Time and Spread within the
reverberant space.
Size values are given in meters and can be used to approximate the size of the acoustic space you
want to simulate. When considering size, keep in mind that the size of a reverberant space in meters
is roughly equal to its longest dimension.
l 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.
l Pre-Delay
Determines the amount of time that elapses between the original audio event and the onset of
reverberation. Under natural conditions, the amount of Pre-delay depends on the size and
construction of the acoustic space, and the relative position of the sound source and the listener. Pre-
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delay attempts to duplicate this phenomenon and is used to create a sense of distance and volume
within an acoustic space. Long Pre-Delay settings place the reverberant field behind rather than on
top of the original audio signal.
g For an interesting musical effect, set the Pre-Delay time to a beat interval such as 1/8, 1/16, or 1/32
notes.
Reverb One Early Reflection Controls

The Early Reflections section has controls for the various early reflection elements, including ER setting,
level, spread, and delay.
Calculating Early Reflections
A particular reflection within a reverberant field is usually categorized as an early reflection. Early
reflections are usually calculated by measuring the reflection paths from source to 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.
ER Settings
Selects an early reflection preset. These range from realistic rooms to unusual reflective effects. The last five
presets (Plate, Build, Spread, Slapback and Echo) feature a nonlinear response.
Early reflection presets include:

l Room: Simulates the center of a small room without many reflections.
l Club: Simulates a small, clear, natural-sounding club ambience.
l Stage: Simulates a stage in a medium-sized hall.
l Theater: Simulates a bright, medium-sized hall.
l Garage: Simulates an underground parking garage.
l Studio: Simulates a large, live, empty room.
l Hall: Places the sound in the middle of a hall with reflective, hard, bright walls.
l Soft: Simulates the space and ambience of a large concert hall.
l Church: Simulates a medium-sized space with natural, clear-sounding reflections.
l Cathedral: Simulates a large space with long, smooth reflections.
l Arena: Simulates a big, natural-sounding empty space.
l Plate: Simulates a hard, bright reflection. Use the Spread control to adjust plate size.
l Build: A nonlinear series of reflections.
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l Spread: Simulates a wide indoor space with highly reflective walls.
l Slapback: Simulates a large space with a long-delayed reflection.
l Echo: Simulates a large space with hard, unnatural echoes. Good for dense reverb.
Level
Controls the output level of the early reflections. Turning the Early Reflections Level slider completely off
produces a reverb made entirely of reverb tail.
Spread
Globally adjusts the delay characteristics of the early reflections, moving them closer together or farther
apart. Use Spread to vary the size and character of an early reflection preset. Setting the Plate preset to a
Spread value of 50%, for example, will change the reverb from a large, smooth plate to a small, tight plate.
Delay Master
Determines the amount of time that elapses between the original audio event and the onset of early
reflections.
Early Reflect On
Toggles early reflections on or off. When early reflections are off, the reverb consists entirely of reverb tail.
Reverb One Graphs

The reverb graphs display information about the tonal spectrum and envelope contour of the reverb. The
Reverb EQ and Reverb Color graphs provide graphic editing tools for shaping the harmonic spectrum of the
reverb.
Reverb One EQ graph controls
Editing Graph Values

In addition to the standard slider controls, the Reverb EQ and Reverb Color graph settings can be adjusted
by dragging elements of the graph display.
To select the EQ or Color graph for editing:

t Select the EQ icon or the Color icon.
Reverb One (AAX version), EQ icon selected
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To cut or boost a particular band:
t Drag a Band Cut / Boost breakpoint up or down.
To adjust frequency or crossover:
t Drag a Frequency / Crossover slider right or left.
To adjust high-frequency cut or damp:
t Drag the HF Cut / HF Damp control point right or left.
Adjusting graph controls
Reverb EQ Graph
You can use this 3-band equalizer to shape the tonal spectrum of the reverb. The EQ is post-reverb and
affects both the reverb tail and the early reflections.
l 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.
l 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.
l HF Cut Breakpoint
Sets the frequency above which a 6 dB/octave low pass filter attenuates the processed signal. It
removes both early reflections and reverb tails, affecting the overall high-frequency content of the
reverb. Use the HF Cut control to roll off high frequencies and create more natural-sounding
reverberation. The adjustable range is from 120.0 Hz to 24.0 kHz.
Reverb Color Graph
You can use the Reverb Color graph to shape the tonal spectrum of the reverb by controlling the decay
times of the different frequency bands. Low and high crossover points define the cut and boost points of
three frequency ranges.
For best results, set crossover points at least two octaves higher than the frequency you want to boost or
cut. For example, to boost a signal at 100 Hz, set the crossover to 400 Hz.
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Set the crossover to 500 Hz to boost low frequencies most effectively. Set it to 1.5 kHz to cut low frequencies
most effectively.
l Crossover Sliders
Sets the frequency boundaries between the low, mid, and high frequency ranges of the reverberation
filter.
The low-frequency slider sets the crossover frequency between low and mid frequencies in the
reverberation filter. The adjustable range is from 60.0 Hz to 22.5 kHz.
The high-frequency slider sets the crossover frequency between mid and high frequencies in the
reverberation filter. The adjustable range is from 64.0 Hz to 24.0 kHz.
l 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.
l 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.
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
To use tool tips, move the cursor over the name of any control, and an explanation appears as a tool tip.
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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.
ReVibe II
ReVibe II is a studio-quality reverb and acoustic environment modeling plug-in available in DSP, Native, and
AudioSuite formats.
Using ReVibe II
ReVibe II makes it possible to model extremely realistic acoustic spaces and place audio elements within a
mix.
ReVibe II supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
ReVibe II 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 II.
ReVibe II supports the following combinations of track types and plug-in insert formats:
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Track Type Plug–in Insert Format
Mono Stereo LCR LCRS Quad 5.0 5.1

Mono • • • • • • •
Stereo • • • • • •
LCR • • • • •
LCRS •
Quad •
5.0 •
5.1 •
Dragging in the Graphic Display to Adjust Controls

In addition to dragging controls and typing control values, you can adjust settings on the Decay Color & EQ
graphic displays by dragging control points on the graph.
To cut or boost a particular EQ band:

t Drag a control point up or down.
Cutting or boosting an EQ frequency band
To adjust EQ frequency crossover:

t Drag the control point right or left.
Setting the EQ crossover frequency
To adjust high frequency rear cut:

t Drag the control point right or left.
Setting the rear cut frequency
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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.
Input and Output meters (5.1)
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.
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 settings remain unchanged. To create a preset that includes all parameters, use the Plug-In
Settings menu.
n For more information on saving and importing plug-in presets, see the Pro Tools Reference Guide.
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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.
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.
g See "ReVibe II Room Types" on page 215 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:
l Automatic selects the reverb tail type stored with the room type.
l Natural is an average reverb tail type with no extreme characteristics.
l Smooth is optimized for large rooms.
l Fast Attack can be useful for plate reverbs.
l Dense is similar to smooth, and can also be good for a plate reverb.
l Tight is good for small to medium rooms.
l Sparse 1 produces sparse early reflections with a high diffusion buildup.
l Sparse 2 can be useful for a spring reverb.
l Wide is a generic large reverb.
l Small is optimized for small 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.
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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).
Higher Size settings increase both the Time and Spread values.
g 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
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.
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.
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.
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 ambiance 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%.
Pre-Delay 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.
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.
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.
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Attack Shape Control
Attack Shape determines the contour of the attack portion of the reverberation envelope. At 0%, there is
no buildup contour, and the reverb tail begins at its peak level. At a high Attack Shape setting the reverb tail
begins at a relatively low initial level and ramps up to the peak reverb level. The range of this control is from
0% to 100%.
Rear Shape Control
Rear Shape adjusts the envelope of the reverb in the rear channels to control the length of the attack time.
This gives more reverb presence and a longer reverb bloom in the rear channels. The range of this control is
from 0% to 100%.
ReVibe II 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 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.
Level Control
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.
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%.
g 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.
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.
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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.
High Frequency Color 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%.
Low Frequency Color 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%.
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 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.
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Center Control
Center controls the output level of the center channel outputs of multichannel formats that have a center
channel (such as LCR or 5.1).
When ReVibe 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 reflections in the rear outputs. The range of this control is from –
INF (minus infinity) to 0.0 dB.
g 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 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%.
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.
Chorus On Button
This button toggles the chorus effect on or off.
ReVibe II Mix Section Controls

The Mix section has controls for adjusting the relative levels of the source signal and the reverb effect.
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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 control. The range of this control is from 0% to 100%.
g 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.
The range of this control is from 0% to 150%. The default setting is 100%.
g 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 setting (Levels section) to –
INF dB.
100% Wet and Dry Mix Buttons
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.
ReVibe II Decay EQ Graph

The EQ display lets you adjust the Decay EQ settings for ReVibe II. Click the EQ button to toggle the display
to show the Decay EQ settings. To adjust a setting on the graph, drag the corresponding control point.
EQ display
Each control point on the graph has corresponding text fields above and below the display that show the
current values.
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.
Low Gain Control

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.
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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

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 adjust the Decay Color settings for ReVibe II. Click the Color button to toggle the
display to show Decay Color settings. To adjust a setting on the graph, drag the corresponding control
point.
Color display
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.
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.
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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.
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.
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
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 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
l Large Natural Studio 1
l Large Natural Studio 2
l Large Live Room 1
l Large Live Room 2
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l Large Dense Studio 1
l Large Dense Studio 2
l Medium Natural Studio 1
l Medium Live Room 1
l Medium Live Room 2
l Medium Dense Studio 1
l Medium Dense Studio 2
l Small Natural Studio 1
l Small Dense Studio 1
l Small Dense Studio 2
l Vocal Booth 1
l Vocal Booth 2
l Vocal Booth 3
l Vocal Booth 4
Rooms
l Large Bright Room 1
l Large Bright Room 2
l Large Neutral Room 1
l Large Neutral Room 2
l Large Dark Room 1
l Large Dark Room 2
l Large Boomy Room
l Medium Bright Room 1
l Medium Neutral Room 1
l Medium Dark Room 1
l Small Bright Room 1
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l Small Neutral Room 1
l Small Dark Room 1
l Small Dark Room 2
l Small Boomy Room
Halls
l Large Natural Hall 1
l Large Dense Hall
l Large Sparse Hall
l Medium Natural Hall 1
l Medium Dense Hall
l Small Natural Hall 1
l Small Natural Hall 2
Theaters
l Large Theater 1
l Large Theater 2
l Medium Theater 1
l Medium Theater 2
l Small Theater 1
l Small Theater 2
Churches
l Large Natural Church 1
l Large Natural Church 2
l Large Dense Church
l Large Slap Church
l Medium Natural Church 1
l Medium Natural Church 2
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l Medium Dense Church
l Small Natural Church 1
l Small Natural Church 2
Cathedrals
l Natural Cathedral 1
l Dense Cathedral 1
l Dense Cathedral 2
l Slap Cathedral
Plates
l Large Natural Plate
l Large Bright Plate
l Large Synthetic Plate
l Medium Natural Plate
l Medium Bright Plate
l Small Natural Plate
l Small Bright Plate
Springs
l Guitar Amp Spring 1
l Studio Spring 1
l Studio Spring 2
l Studio Spring 3
l Studio Spring 4
l Dense Spring 1
l Dense Spring 2
l Resonant Spring
l Funky Spring 1
l Funky Spring 2
l Funky Spring 3
l Funky Spring 4
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Chambers
l Large Chamber 1
l Large Chamber 2
l Large Chamber 3
l Large Chamber 4
l Large Chamber 5
l Large Chamber 6
l Medium Chamber 1
l Medium Chamber 2
l Medium Chamber 3
l Medium Chamber 4
l Medium Chamber 5
l Small Chamber 1
l Small Chamber 2
l Small Chamber 3
l Small Chamber 4
Ambience
l Large Ambience 1
l Large Ambience 2
l Large Ambience 3
l Large Ambience 4
l Medium Ambience 1
l Medium Ambience 2
l Medium Ambience 3
l Medium Ambience 4
l Medium Ambience 5
l Small Ambience 1
l Small Ambience 2
l Small Ambience 3
l Very Small Ambience
Film and Post
l Medium Kitchen
l Small Kitchen
l Bathroom 1
l Bathroom 2
l Bathroom 3
l Bathroom 4
l Bathroom 5
l Shower Stall
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l Hallway
l Closet
l Classroom 1
l Classroom 2
l Large Concrete Room
l Medium Concrete Room
l Locker Room
l Muffled Room
l Very Small Room 1
l Very Small Room 2
l Very Small Room 3
l Car 1
l Car 2
l Car 3
l Car 4
l Car 5
l Phone Booth
l Metal Garbage Can
l Drain Pipe
l Tin Can
Large Spaces
l Parking Garage 1
l Parking Garage 2
l Parking Garage 3
l Warehouse 1
l Warehouse 2
l Stairwell 1
l Stairwell 2
l Stairwell 3
l Stairwell 4
l Stairwell 5
l Gymnasium
l Auditorium
l Indoor Arena
l Stadium 1
l Stadium 2
l Tunnel
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Vintage Digital
l Large Hall Digital
l Medium Hall Digital
l Large Room Digital
l Medium Room Digital
l Small Room Digital
Effects
l Mono Slapback 1
l Mono Slapback 2
l Mono Slapback 3
l Wide Slapback 1
l Wide Slapback 2
l Wide Slapback 3
l Multi Slapback 1
l Multi Slapback 2
l Multi Slapback 3
l Multi Slapback 4
l Spread Slapback 1
l Spread Slapback 2
l Mono Echo 1
l Mono Echo 2
l Mono Echo 3
l Wide Echo 1
l Wide Echo 2
l Multi Echo 1
l Multi Echo 2
l Prism
l Prism Reverse
l Inverse Long
l Inverse Medium
l Inverse Short
l Stereo Enhance 1
l Stereo Enhance 2
l Stereo Enhance 3
Space
Space is an AAX format convolution reverb plug-in that is available in DSP, Native, and AudioSuite formats.
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, Space offers stunning realism
with full control of reverb parameters in mono, stereo, and surround formats.
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Space supports 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz sample rates.
Space works with mono, stereo, and mono-to-stereo formats. With Pro Tools Ultimate and Studio, Space
also supports Quad, 5.0, mono-to-Quad, stereo-to-Quad, mono-to-5.0, and stereo-to-5.0 multichannel
formats.
Space Feature Highlights

Space features let you create the best reverb effect in the shortest possible time.
Reverb Features
l Mono, Stereo, Quad, and 5.0–channel output support
l Multiband EQ
l Independent wet/dry and decay levels
l Separate reverb early and late levels and length
l Control of early size, low-cut, and balance
l Pre delay and late delay controls
l Precise control of low, mid, and high decay crossover
l Adjustable waveform reverse, displayed in beats per minute
l Waveform processing bypass
Interface Features
l Full waveform view, zoom, and channel highlight functions
l On-screen input and output metering with clip indicators
l Impulse response information display
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Impulse Response (IR) Loading and Organization Features
l Scrollable IR browser makes finding impulse responses easy
l Browser supports user-defined IR groups on any local drives
l Browser keyboard shortcuts
l IR favorites function
l Automatically recognizes common IR formats for one click loading
l Quick browser buttons allow rapid IR loading and preview
Automation and Ease of Use Features
l Snapshot mode supports rapid changes between ten predefined reverb scenes
l Picture preview mode allows you to view image files stored with impulse responses
l Impulse responses stored directly in Pro Tools presets and sessions for easy session sharing
l New impulse responses can be copied to system and loaded without closing Space
l iLok support for quick and easy relocation to other Pro Tools systems
Surround and Post-Production Features
l Full input and output surround metering on screen at all times
l Separate front, center, and rear levels
l Independent front and rear decay
l Snapshot mode ideal for post automation requirements
l Seamless snapshot switching
l Automatic phantom channel creation
IR Library
l A wide variety of both real and synthetic reverb spaces and effects
l Mono, stereo, and surround formats
l All reverb impulse responses stored in WAV file format
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
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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 amplifiers beginning in the 1960s. Similar to the plate
reverb in operation, the spring reverb uses a transducer to feed the signal into a coiled steel spring and
create vibrations. These are then captured via a pickup and fed back into an amplifier.
Since the advent of digital audio technology in the 1980s, artificial reverberation has been created primarily
by digital algorithms that crudely mimic the physics of natural reverb spaces by using multiple delay lines
with feedback. Digital “synthetic” reverb units offer a new level of realism and control unavailable with
older analog reverb systems, but still fall short of the actual reverb created by a real space.
Components of Reverb
Reverberation sound in a normal space usually has several components. For example, the sound of a single
hand clap in a large cathedral will have the following distinct parts. 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 typically occurs first, followed 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. Space offers various controls over early reflection parameters.
The time delay between the direct sound and the first reflection is usually known as pre-delay. Space lets
you adjust pre-delay. Increasing the pre-delay often changes 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. 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, Space lets you adjust the decay as desired.
For surround processing, decay can be adjusted for individual channel groups.
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 Space, clearly shows the early reflections and the long decay of the
reverb tail.
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Impulse Response sample
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, 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 has been the heavy CPU processing
requirement, which can result in 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.
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.
Space System Design

Space uses advanced DSP algorithms to deliver convolution reverb processing.
The following figure shows the internal system design of Space and demonstrates how Space processes the
audio signal.
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Space internal system design
The impulse computer is an internal module of Space that provides extensive control over the currently
loaded impulse response waveform. When you adjust 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.
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Space internal functions of the impulse computer
Impulse Response (IR) and Space

This section covers aspects of impulse response (IR) and Space.
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IR Processing Control Lag
Adjusting some controls in 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.
Similarly, an IR of a hardware effects unit can be captured 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.
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.
Multiple IRs may be taken of a physical space where the sound source has been moved to different 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.
Space IR Library Installation

You can download IR Libraries from Avid’s Space Online IR Library. For more information on downloading
and installing IR Libraries from the Space Online IR Library, see "Installing Space IR Packages " on
page 238.
Using Third-Party IRs in Space

Space reads a wide range of IR formats automatically, including WAV and AIFF file formats, allowing you to
import a variety of IRs. Space supports IR sample rates from 22 kHz up to 96 kHz in bit depths from 16 to 32
bits. In addition, Space supports the display of JPEG format picture files stored with IRs.
To use third-party IR libraries with Space:
1. In the IR Browser, select Edit > Import Other IR Folder.

2. Locate and select the library on your hard drive.
3. Click Choose.
Space Multichannel IR Formats
Space supports IRs in multichannel or multiple mono audio files. IRs with a single input are used for mono or
summed stereo processing and can be stored as a single interleaved multichannel file, or as multi-mono
files. IRs with stereo inputs used for true stereo processing must be stored as multi-mono files.
The following table shows Space IR channel formats.
Input Output Channel Order File Format
Mono Mono — Mono file
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Input Output Channel Order File Format
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, Space understands the following filename conventions, based on those used by
Pro Tools. The filename format (Impulsename.inputchannel.outputchannel.type) is based on the impulse
name plus two suffixes which indicate input and output channels as follows:
l Impulsename is the name of the impulse. Mixing multiple IR files with the same Impulsename in the
same folder is not supported.
l 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.
l 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.
l Type is optionally .WAV or AIFF. For best performance, filenames should always be suffixed with type
to avoid Space having to open the file to determine audio format.
The following examples show how various multi-mono IR files could be named.
Stereo to Stereo IR
l Cathedral.1.L.wav
l Cathedral.1.R.wav
l Cathedral.2.L.wav
l Cathedral.2.R.wav
Stereo to 5.0 IR
l Cathedral.1.L.wav
l Cathedral.1.C.wav
l Cathedral.1.R.wav
l Cathedral.1.Ls.wav
l Cathedral.1.Rs.wav
l Cathedral.2.L.wav
l Cathedral.2.C.wav
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l Cathedral.2.R.wav
Mono to Quad IR
l Cathedral.L.wav
l Cathedral.R.wav
l Cathedral.Ls.wav
l Cathedral.Rs.wav
Stereo to Quad IR
l Cathedral.1.L.wav
l Cathedral.1.R.wav
l Cathedral.2.L.wav
l Cathedral.2.R.wav
Channel Compatibility and Space
Space works best with IRs that match your current channel configuration. For example, if Space is
instantiated in a mono to stereo configuration, stereo IRs will be highlighted in the IR browser. The IR
information 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, 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 Space, 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, Space will automatically create a phantom channel for
the IR if needed. For example, if a stereo IR is loaded into a quad instantiation, 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, 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 Space
will automatically have a phantom center channel created.
Space Presets
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 Space:
l 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.
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l A Space preset only includes the currently selected snapshot.
c IR files are audio files only and do not contain information about 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 Space.
Space Snapshots
In addition to presets, 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
Space controls.
IRs in a snapshot have been pre-processed by the impulse computer and can be loaded instantly into the
convolution processor. 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 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 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 Space’s Preferences. If IR embedding is disabled, Space stores only a
reference to the name of the IR file. When the session is transferred to a different system, Space attempts to
load the matching IR file from the 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.
g By default, Pro Tools presets or session files created using Space automatically include copies of all
relevant IR waveforms. This provides maximum compatibility of session files between different
Pro Tools systems.
c It is your responsibility to ensure that you observe the copyright on any IR transferred to a third party
in this fashion.
Space Controls and Displays

The Space interface is divided into the following sections:
1. Display area ("Space Display Area" on the next page)

2. IR Browser ("Space IR Browser" on page 236)
3. Primary controls ("Space Primary Controls" on page 239)
4. Group Selectors and Controls ("Space Group Selectors and Controls" on page 239)
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Space Display Area
The display area of Space operates in the following four modes, indicated by the Display Mode selectors at
the top right hand corner of the Space window:
l Waveform mode
l Picture Preview mode
l Snapshot mode
l Preferences mode
Display Mode selectors
The Display area changes based on the selected mode.
Info Bar
At all times, the Info bar at the bottom of the display area window shows the following controls and
information.
Snapshot Menu
A pop-up menu allowing quick selection or automation of a snapshot.
IR Name
Displays the folder and file name of the currently loaded IR.
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Quick Browser Controls
The Quick browser controls allow the IR to be quickly changed even when the IR browser is closed,
automatically loading each IR sequentially. The Waveform icons step backwards and forwards through IRs
and automatically load the IR file. The Folder icons step backwards and forwards through folders. The
Quick browser requires an IR to be currently loaded from the IR browser. If no such 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.
Space Waveform Mode

Waveform mode is selected using the Waveform icon at the top of the Space window.
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.
Display area, Waveform mode
The controls in Waveform mode function as follows:
Original
Bypasses all waveform processing, allowing the original IR to be auditioned. This control effectively
bypasses the processing in the IR computer as shown in the system diagram.
Channel Selectors
Displays from one to five channels (in the order Left, Center, Right, Left Surround, Right Surround). Click the
desired channel to display the IR waveform for that channel. In Mono mode, no channel selector is
displayed.
Zoom
Zooms in and out on the time axis for the waveform display.
Space Picture Preview Mode

Picture Preview mode is selected using the Picture Preview icon at the top of the Space window. When
selected, Picture Preview mode shows pictures associated with the IR. For an IR provided with 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.
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Display area, Picture Preview mode
Space Snapshot Mode

Snapshot mode is selected using the Snapshot icon at the top of the Space window. Space provides up to
ten snapshots that are 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.
The name of the currently selected snapshot is always displayed in the Info bar at the bottom of the display
area, and can be automated. This lets you switch reverb settings during playback and is useful for post
production sessions where the reverb setting may change as the scene changes.
Display area, Snapshot mode
The active snapshot can be selected in one of two ways. At any time, a snapshot can be selected by using
the snapshot menu in the Info bar. Alternatively, when the display area is in Snapshot mode, a snapshot
can be selected by clicking the selection area next to the snapshot name.
Select
Lets you select which snapshot is currently loaded.
Name
Displays the name of each snapshot. By default, snapshots are named “Snapshot 1” through “Snapshot
10.” Snapshots can be renamed by clicking on the snapshot name and entering a new name followed by the
Enter key (Windows) or the Return key (Macintosh).
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Sample Path
Displays the name of the IR selected for each snapshot.
Copy
Copies the currently selected snapshot settings into a clipboard.
Paste
Pastes the clipboard into the currently selected 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.
Space Preferences Mode

Preferences mode is selected using the Preferences icon at the top of the Space window. This displays a
number of preferences settings for Space.
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.
Installed IR Packages
Displays a list of installed Space IR packages and their versions.
Space Meters
The Meters display the amplitude of the incoming and outgoing audio signals by channel. The number of
meters shown will depend on the number of input and output channels. Input meters may be mono or
stereo, and output meters may be mono, stereo, quad, or 5.0 channels. Each meter is marked as either
mono, left, right, center, left surround, or right surround. A logarithmic scale marked in decibels and
momentary peaks are also displayed on the meter.
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Meters, stereo input to 5.0 output shown
The red Clip indicators at the top of the meters indicate clipping on the corresponding channel. 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 Track > Clear All Clip Indicators in Pro Tools, or pressing Option+C (Mac) or Alt+C
(Windows).
Space IR Browser
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.
Space automatically highlights each IR that matches the current channel configuration. For example, when
using a Space Stereo to Quad inset, each IR with that configuration is highlighted. Impulses that are not
highlighted can still be loaded, and Space tries to adapt the IR to the current channel format (see "Channel
Compatibility and Space" on page 230).
IR Browser
An IR can be loaded by double clicking with the mouse, or using the Load button displayed at the top of the
IR browser drawer. The currently loaded IR is highlighted with a small dot next to the file name in the
browser.
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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.
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. In the IR browser, select the desired IR file or folder.

2. From the IR browser’s Edit menu, select Add to Favorites.
Space IR Browser Edit Menu
The IR browser’s Edit menu contains the following commands:
Download Space IR Package
Opens a Web browser to the Space online IR library.
Install Space IR Package
Installs a new IR package downloaded from the Space online library (see "Installing Space IR Packages " on
the next page).
Import Other IR Folder
Lets you import a new IR folder in common file formats. By default, the new IR is given the same name as the
selected folder.
Remove Imported IR Folder
Lets you remove the currently selected IR folder.
Rename Imported IR Folder
Lets you rename the currently selected IR folder.
Add to Favorites
Adds the currently selected IR to the Favorites group at the top of the browser window.
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New Folder in Favorites
Creates a folder in the Favorites group. Favorite IRs can be dragged and dropped into the folder.
Rename Favorites Folder
Lets you rename the 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 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 Space online IR library.
Rescan for Files
Forces 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 Space without needing to close
Space or the Pro Tools session.
c 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.
Installing Space IR Packages

Additional IR packages for Space are available for registered users to download from the Space Online IR
Library at: www.avid.com/plugins/space-impulse-response-library
These package files are supplied in a lossless compressed format.
To install a Space IR package:
1. In the Space IR browser, select Download IR Package from the Edit menu. Your default Web
browser launches and loads the Avid Space Online IR Library website.
2. Click Download.
3. Log in using your email address and password. You may need to create a new account if you have
not yet registered Space.
c To download IR packages from the 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 Space, select Install Space IR Package from the Edit menu.
7. In the resulting dialog, locate and select the file you downloaded.
8. Click Choose.
9. Click Install to install the IR package. A window is displayed with the results of the installation.
The IR browser in Space updates to include the new IR.
If a problem occurs with the IR installation, Space displays an error message. Review the log file stored in
the Space IR library for further details. Each IR package has a version number, and Space warns you if an IR
package has already been installed.
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The details of all installed IR packages can be reviewed using the Show Packages option in Preferences
mode.
Space Primary Controls

The primary control group is visible at all times and allows control of key reverb parameters. This includes
the wet and dry levels of the audio passing through Space.
Reset
Resets all 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 waveform and is displayed as a percentage of the original. When Decay
is adjusted, the waveform is recalculated in real time.
Space Group Selectors and Controls

Space presents reverb controls in five different groups. Each group is activated by selecting the
corresponding selector.
Group Selectors
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.
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Level controls (5.0 shown)
Input
Cuts or boosts the input signal level from –inf dB to +12 dB.
Output
Cuts or boosts the output signal level from –inf dB to +12 dB.
Early
Cuts or boosts the levels of the early reflections from –inf dB to +12 dB.
Late
Cuts or boosts the levels of the late reflections from –inf dB to +12 dB.
Front/Rear/Center
In quad and 5.0 channel output modes, Space provides additional controls to attenuate or boost the Front
(left and right), Rear (left and right), and Center (5.0 only) 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.
Space Delay Controls

The Delays group provides controls for the delay timings of the reverb. When changes are made to any
control in the Delays group, the IR waveform is recalculated and displayed in the Waveform display.
Delay controls (5.0 shown)
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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.
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, 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.
Early controls
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.
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Early controls indicated in IR waveform
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.
For the most realistic reverb results, Early Length should be adjusted while viewing the waveform display.
The early portion of a reverb IR is typically seen as a series of discrete spikes at the beginning of the
waveform. Early Length can however be adjusted to any value to explore other creative possibilities.
Size
Changes the size of the Early reflections, from 50% to 200%. Early Size expands or contracts the reflections
in the early portion of the IR (as specified by the Early Length control). Reduce the Early Size to give the
space a smaller, tighter sound. Increase the Early Size to give the space a larger, roomier sound.
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.
c When loading an IR from an audio file, 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.
Space Reverb Section Controls

The Reverb group offers a low and high shelf EQ in addition to width and balance controls. The EQ operates
prior to convolution processing.
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Reverb controls
Lo Freq
Adjusts the frequency of a low frequency filter from 20 to 500 Hz.
Lo Gain
Cuts or boosts the frequency set in Lo Freq from –15 dB to +15 dB.
Hi Freq
Adjusts the frequency of a high frequency filter from 500 Hz to 20 kHz.
Hi Gain
Cuts or boosts the frequency set in Hi Freq from –15 dB to +15 dB.
Width
Increase or reduces the stereo 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.
c 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.
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.
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Decay controls
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.
Using Space
This section addresses some common scenarios in which Space can be used during a Pro Tools session.
Using Space Presets

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.
Using Space on an Effect Send

When Space is used on an Aux Input track as an effects send, the Dry control should be set to –inf dB.
Automating Space Snapshots

Snapshot automation is a powerful method of changing the reverb parameters without having to
individually automate each parameter.
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To automate Space Snapshots:
1. Insert Space on a track.

2. Select Snapshot mode.
3. Load an IR into each Snapshot and make any desired changes to specific Space controls.
4. Name each Snapshot as desired.
5. Click Auto.
6. Add Snapshot to the list of automated controls.
7. Select Space > Snapshot from the automation menu for the track.
8. Draw the desired automation on the track with the Pencil tool. The names displayed in the
automation track will match the names entered for each Snapshot.
Space IR Library Categories

Space includes an extensive impulse response library, divided into the following categories.
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
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6 Delay Plug-Ins
Delay plug-ins let you add echo effects to material. Synchronizing delay to a song's tempo can add
interesting rhythmic qualities to a track. Other uses for delay include stylistic effects such as slap-back on
vocals or subtle support underneath the main signal of a guitar solo, or special effects as heard in dub and
other styles of music. Delay is also useful in post-production to add realism to scenes that take place in a
canyon, parking garage, or other echo-heavy environment, and in all production environments for time-
aligning signals for corrective or creative applications.

l "Mod Delay III" below
l "Moogerfooger Analog Delay" on page 249
l "Reel Tape Delay" on page 251
l "Tel-Ray Variable Delay " on page 255
l "TimeAdjuster" on page 257
Mod Delay III

Mod Delay III provides mono, multi-mono, mono-to-stereo, and stereo modulating delay effects.
Mod Delay III is available in DSP, Native, and AudioSuite plug-in formats.
Mod Delay III supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Mod Delay III Controls

Mod Delay III provides separate sections in the plug-in window for Input and Output metering, Delay and
Modulation controls, and for the Wet/Dry Mix control. Stereo and mono-to-stereo versions provide meters
and controls for each channel. Delay, Modulation, and Mix controls for stereo and mono-to-stereo
instances of Mod Delay III can be linked, or can be operated independently.
Input
Input Meters
The Input meters show peak signal levels before processing:

l Dark Blue
l Light Blue
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l 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:

t Click the Phase Invert button so that it is highlighted. Click it again so that it is not highlighted to
disable it.
Delay
Link
For stereo and mono-to-stereo tracks, enable the Link button to link the Delay, Modulation, and Mix
controls between the Left and Right channels. This option is highlighted when it is enabled.
For mono tracks, this option reads Mono and is display only.
Delay Time
The Delay Time control sets the delay time between the original signal and the delayed signal (from 0.0 ms
to 5,000.0 ms).
Feedback (FBK)
The Feedback setting controls the amount of feedback applied from the output of the delay back into its
input (from –100% to 100%). It also controls the number of repetitions of the delayed signal. Negative
feedback settings give a more intense “tunnel-like” sound to flanging effects.
Low Pass Filter (LPF)
The Low Pass Filter setting controls the cutoff frequency of the Low Pass Filter (from 10 Hz to 22 kHz). Use
the LPF setting to attenuate the high frequency content of the feedback signal. The lower the setting, the
more high frequencies are attenuated. The maximum value for LPF is Off. This lets the signal pass through
without limiting the bandwidth of the plug-in.
Sync
When Sync is enabled, and a Duration (a rhythmic note value) is selected, the Delay Time setting is affected
by the session tempo. When Sync is disabled, and a Duration is selected, the Delay Time setting is affected
by changes to the Tempo setting.
When Tempo Sync is enabled, the Tempo and Meter controls are uneditable and follow the session tempo
and meter changes in the Pro Tools timeline. The Duration and Groove controls apply regardless of whether
Sync is enabled.
Meter
The Meter setting lets you enter either simple or compound time signatures. The Meter control defaults to a
4/4 time signature.
When Sync is enabled, the Meter control is unavailable.
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Tempo
The Tempo control sets the tempo in beats per minute (from 5.00 to 500.00 bpm). This setting is
independent of the Pro Tools session tempo. When a specific Duration is selected, moving this control
affects the Delay Time setting.
When Sync is enabled, the Tempo control is unavailable.
Duration
The Duration setting lets you set the Delay Time based on a rhythmic value. Select a 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
Rate
The Rate control sets the rate of modulation of the delayed signal (from 0.00 Hz to 20.0 Hz).
Depth
The Depth control sets the depth of the modulation applied to the delayed signal (from 0% to 100%).
Mix
The Mix control sets the balance between the delayed signal (wet) and the original signal (dry). If you are
using a delay for flanging or chorusing, you can control the depth of the effect somewhat with the Mix
setting. Click the Dry button to set the Mix to 100% dry. Click the Wet button to set the Mix to 100% wet.
Output
Output Meters
The Output meters show peak signal levels after processing:

l Dark Blue
l Light Blue
l Red
Indicates full scale levels (clipping)
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Output Gain
The Output Gain control sets the output level after processing. For mono instances of Mod Delay III, there is
a single Gain control. For stereo and mono-to-stereo instances of Mod Delay III, there are independent Gain
controls for each channel (left and right).
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 will be cut off.
Moogerfooger Analog Delay

Moogerfooger Analog Delay is a delay plug-in that is available in DSP, Native, and AudioSuite formats.
Moogerfooger Analog Delay supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample
rates.
Moogerfooger Analog Delay operates as a mono, multi-mono, or stereo plug-in.
Moogerfooger Analog Delay provides a warm sounding delay in the digital domain. 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. 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—a classic musical effect.
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.
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.
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Avid’s digital replica re-creates all the warm, natural sounds of its analog counterpart. 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 frequencies from the feedback loop. It removes undesirable low frequency
“mud” common when mixing with delays and also allows the creation of amazing echo swarms that won’t
clip the output. Dial in a highpass frequency from 50 Hz to 500 Hz. Frequencies below the setting are
filtered from the feedback loop.
HPF On/Off
The HPF Off/HPF On enables or disables the highpass filter (HPF).
Drive
The Drive control sets the input gain.
Mix
The Mix control blends the original input signal with the delayed signal.
LED Indicators
Three LEDs down the center of the unit provide visual feedback.
Input Level
The Input Level LED glows green when signal is present.
HPF
The HPF LED turns green when the highpass filter is enabled.
Bypass
The Bypass LED glows either red (bypassed) or green (not bypassed) to show whether or not the effect is in
the signal path.
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Moogerfooger Analog Delay Tips and Tricks
Infidelity
Because analog delay chips offer only a fixed number of cells, the extended delay times store a lower-
fidelity version of the input signal. Try the Long delay setting when going for cool “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.
Reel Tape Delay

Reel Tape Delay is part of the Reel Tape suite of tape-simulation effects plug-ins. Reel Tape Delay simulates
an analog tape echo effect, modeling the frequency response, noise, wow and flutter, and distortion
characteristics of analog tape. It also reproduces the varispeed effect you get when the tape speed control
is adjusted.
Reel Tape Delay is available in DSP, Native, and AudioSuite formats.
Reel Tape Delay supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Reel Tape Delay operates as a mono, multi-mono, or stereo plug-in.
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:
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l Speed: 15 ips
l Bias: 0.0 dB
l Cal Adjust: +9 dB
You can use the BPM Sync feature to synchronize the Reel Tape Delay effect to the current tempo of the
Pro Tools session.
Reel Tape Delay can be placed on mono, stereo, or multichannel tracks.
Reel Tape Common Controls

All Reel Tape plug-ins share the following controls:
Drive
Drive controls the amount of saturation effect by increasing the input signal to the modeled tape machine
while automatically compensating by reducing the overall output. Drive is adjustable from –12 dB to +12 dB,
with a default value of 0 dB.
Output
Output controls the output signal level of the plug-in after processing. Output is adjustable from –12 dB to
+12 dB, with a default value of 0 dB.
Tape Machine
The Tape Machine control lets you select one of three tape machine types emulated by the 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.
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:
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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. "Synchronizing Reel Tape
Delay to Session Tempo" on the next page
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
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.
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.
g Settings for this parameter are saved with plug-in presets. If you use a preset for the Native, DSP, 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.
g Note that this control does not affect the first delayed signal, only the repeated delays caused by the
Feedback control.
Treble
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.
g 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.
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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.
g Settings for this parameter are saved with plug-in presets. If you use a preset for the Native, DSP, or
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.
BPM Sync controls
2. To set a rhythmic delay, click the Note Value to choose from the available note values (whole, half,
quarter, eighth, sixteenth, or thirty-second note)
3. To adjust the rhythm further, do any of the following:
t To enable triplet rhythm delay timing, click the Triplet (“3”) button so that it is lit.
t To set a dotted rhythm delay value, click the Dot (“.”) button so that it is lit.
g You can override the settings derived from BPM Sync at any time by manually adjusting the plug-in
Speed control.
g 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.
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.
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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.
30 ips Flutter
Adds Wow/Flutter to the highest Speed Control setting.
Rockabilly
A common tape slap effect, useful on vocals or electric guitar. Sets the delay time to 130 ms, which
corresponds to the delay time resulting from the distance between the record and play heads on an Ampex
300-series or Ampex 350-series tape transport.
Rockabilly Plus
Includes the Rockabilly setting plus Feedback, Wow/Flutter, Bass and Treble adjustments on feedback.
Tel-Ray Variable Delay

Tel-Ray Variable Delay is a delay/echo plug-in that is available in DSP, Native, and AudioSuite formats.
Tel-Ray Variable Delay supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Tel-Ray Variable Delay operates as a mono, multi-mono, or stereo plug-in.
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.
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).
One invention involved a tuna can, a motor, and a few tablespoons of 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 technology
that gives Tel-Ray its unique sound.
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Space-age technology in a can
Tel-Ray Controls
Input / Output Section
Input
Input sets the signal level to the tuna can echo unit.
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 control.
Echo/Delay Section
Variable Delay
Variable Delay selects the delay time. Delay times vary from 0.06 to 0.3 seconds. Full clockwise is slowest.
Variation
Variation adjusts how much variation occurs in the delay. The more variation you use, the more warbled
and wobbly the sound becomes.
Sustain
Sustain determines how long the delay takes to die out. It is actually a feedback control similar to the one
found on the Moogerfooger Analog Delay.
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Echo/Doubler
Echo/Doubler determines whether or not a second record head is engaged, resulting in a double echo.
Tel-Ray Tips and Tricks

Each and every Tel-Ray varies drastically in motor and flywheel stability, resulting in different pitch and
variation effects. Even the same unit may sound different day to day, depending on temperature, warm-up
time, and other factors. Since the original units are basically thirty year-old tuna cans bolted to plywood
with springs and motors flopping around inside, the virtual Tel Ray Delay provides a Variation knob so
that you can dial in a Tel-Ray in whatever state of disrepair you desire.
TimeAdjuster
TimeAdjuster is a time-processing plug-in that is available in DSP and Native formats.
TimeAdjuster supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
TimeAdjuster operates as a mono, multi-mono, or stereo plug-in.
Use the TimeAdjuster plug-in for any of the following:

l Delay compensation
l Gain compensation (+/– 24 dB)
l Phase inversion for correcting out-of-phase signals
n 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:
Phase Invert
This controls inverts the phase (polarity) of the input signal. While most Avid plug-ins supply a phase invert
button of their own, some third-party plug-ins may not. Phase inversion is also useful for performing delay
compensation by tuning unknown delay factors by ear (see "TimeAdjuster" above).
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 as an insert effect 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.
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Delay
Provides up to 8192 samples of delay compensation adjustment, or general adjustment of phase

relationships of audio recorded with multiple microphones (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.
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.
g Some plug-ins (such as Avid’s Maxim) have different delays at different sample rates.
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:
t Phase inversion
t Comb-filter effect cancellation
Phase Inversion
If you are working with phase-coherent track pairs, or tracks recorded with multiple microphones, you can
invert the phase to negate the delay. If you don’t hear any audio when you invert a signal’s phase, you
have precisely adjusted and compensated for the delay. This is because when you monitor duplicate signals
and invert the polarity (phase) of one of them, the signals will be of opposite polarity and cancel each other
out. This technique is convenient for finding the exact delay setting for any plug-in.
To determine the delay of a plug-in by inverting its signal phase:
1. Place duplicate audio clips on two different audio tracks and pan them to the center (mono).
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.
Comb-Filter Effect Cancellation
Adjust the delay with the signal in phase until any comb-filter effects cancel out.
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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:

t 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.
t Select the Delay Compensation view from the Mix Window View selector. This view will appear
below the track name. Delay, User Offset, Compensation Amount values are shown in samples.
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 Control-click
(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.
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.
n For more information about delays and mixing with Pro Tools, see the Pro Tools Reference Guide.
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7 Modulation Plug-Ins
Modulation effects include wah, chorus, vibrato, flanging, phasing, and more. In traditional applications,
modulation can add color, character, and texture. At extreme levels, modulation can mutate a sound
beyond recognition to create something entirely new.

l "Eleven Effects Modulation Plug-Ins" below
l "Moogerfooger Lowpass Filter" on page 265
l "Moogerfooger 12-Stage Phaser" on page 268
l "Moogerfooger Ring Modulator" on page 271
l "Reel Tape Flanger" on page 273
l " Sci-Fi" on page 278
l "Voce Plug-Ins" on page 280
Eleven Effects Modulation Plug-Ins

Eleven Effects Modulation plug-ins include the following:
l "Black Shiny Wah" below
l "C1 Chorus/Vibrato" on the next page
l "Flanger" on page 262
l "Orange Phaser" on page 263
l "Roto Speaker" on page 264
l "Vibe Phaser" on page 264
Black Shiny Wah

Black/Shiny Wah is a standard wah pedal effect, offering a sweepable resonant filter that lets you add
animation and special timbral effects to the guitar signal. It also offers a Shine option with a vintage-style
modeled filter circuit and sweep response.
Black/Shiny Wah is available as a Mono (or Multi-Mono) plug-in only.
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Black/Shine
The Black/Shine switch lets you toggle between the Black Wah effect and the Shine Wah effect.
Pedal Control
Click and drag up or down on the virtual pedal control to engage the Wah effect. The control is often best
used with plug-in automation.
C1 Chorus/Vibrato
Based on a heavyweight late-70s analog chorus/vibrato pedal, C1 Chorus/Vibrato offers warm, liquid
modulation effects. In Chorus mode, the signal is routed through a modulated short delay, which is mixed
with the dry signal, creating a washy, doubled sound.
In Vibrato mode, the dry signal is absent and there is more control over the depth of pitch modulation,
allowing for everything from an understated “wobble” to wacky, synth-like pitch modulation.
C1 Chorus/Vibrato is available as a Mono, Mono to Stereo, or Stereo plug-in.
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Chorus
When the Chorus/Vibrato switch is set to Chorus, the Chorus knob sets the intensity and speed of the
Chorus effect.
Sync
The Sync control lets you synchronize the speed of the Chorus effect to the Pro Tools session tempo. Simply
select the rhythmic value from the Sync selector. Set it to Sync Off to control the Chorus setting
manually.
Depth
When the Chorus/Vibrato switch is set to Vibrato., the Depth knob setting determines the amount of the
Vibrato effect.
Rate
When the Chorus/Vibrato switch is set to On, the Rate knob controls the Vibrato rate.
Chorus/Vibrato
The Chorus/Vibrato button toggles the effect between Chorus and Vibrato. The Chorus LED or the
Vibrato LED lights to show which effect is selected.
g When set to Chorus, the Vibrato’s Rate and Depth controls are inactive. Conversely, when set to
Vibrato, the Chorus control will not function.
Flanger
Flanger effects originated with the act of pressing on the flanges of tape reels. Flangers became ubiquitous
with the advent of effects pedals. The Flanger effect can be coaxed into bell-like resonant sweeps, or add a
silky, shimmering sheen to your sound. This effect works well when positioned before or after an amplifier
modeler in the signal chain.
Flanger is available as a Mono or Stereo plug-in.
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Pre-Delay
The Pre-Delay knob sets the amount of pre-delay, which changes the phase relationship between the dry
signal and the delayed signal, with timbral results.
Depth
The Depth knob sets the amount of delay. The higher the setting, the more “jet-engine” artifacts will be
introduced.
Rate
The Rate knob lets you manually set the modulation rate for the flanger effect.
Sync
The Sync control lets you synchronize the modulation rate to the Pro Tools session tempo. Simply select the
rhythmic value from the Sync selector. Set it to Sync Off to control the Chorus setting manually.
Feedback
The Feedback knob sets the amount of signal fed back into the modulated delay. Higher settings introduce
more ringing, whistling artifacts.
Orange Phaser
Don’t let that single knob fool you! Inspired by a ubiquitous 70s analog phaser pedal, Orange Phaser offers
a deep, warm phasing effect that ranges from a slow harmonic sweep to out-of-control wobbles.
Orange Phaser is available as a Mono or Stereo plug-in.
Rate
The Rate knob controls the rate of modulation for the phaser effect.
Sync
The Sync selector lets you synchronize the modulation rate to the session tempo by the specified rhythmic
subdivision.
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Roto Speaker
Inspired by the rotating speaker cabinets that made classic tonewheel organs roar, Roto Speaker offers
added motion and vintage grit.
Roto Speaker is available as a Mono, Mono to Stereo, or Stereo plug-in.
Speed
The Speed lever sets the speed of the rotating speaker effect in three increments: Slow, Fast, and Brake.
Balance
The Balance knob sets the blend between the upper and lower rotors of the rotating speaker.
Type
The Type selector lets you choose between various types of rotary speakers.
Vibe Phaser
The psychedelic-era phaser that inspired our Vibe Phaser effect was traditionally paired with an expression
pedal that could be used to vary the rate of pitch modulation over time. Like the C1, you can choose to run
it as a Chorus, with the modulated and dry signals mixed together, or in Vibrato mode, which leaves the
pitch-modulated signal alone, with rippling, disorienting effects.
Vibe Phaser is available as a Mono or Stereo plug-in.
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On/Bypass
The On/Bypass toggle lets you bypass the Vibe Phaser effect.
Rate
The Rate slider lets you set the modulation rate for the phaser effect.
Sync
The Sync selector lets you synchronize the modulation rate to the session tempo by the specified rhythmic
subdivision.
Volume
The Volume knob lets you set the signal volume passing through the effect.
Chorus/Vibrato
The Chorus/Vibrato switch toggles the dry signal on (Chorus) and off (Vibrato).
Depth
The Depth knob setting determines the amount of the effect.
Power
The Power button for Vibe Phaser functions exactly the same as the On/Bypass toggle.
Moogerfooger Lowpass Filter

Moogerfooger Lowpass Filter features a 2-pole/4-pole variable resonance filter with envelope follower. Use
it to achieve classic 60s and 70s sounds on bass and electric guitar, or just dial in some warm, fat virtual
analog resonance when you need it.
Moogerfooger Lowpass Filter is available in DSP, Native, and AudioSuite formats.
Moogerfooger Lowpass Filter supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample
rates.
Moogerfooger Lowpass Filter operates as a mono, multi-mono, or stereo plug-in.
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With the invention of the MOOG® synthesizer in the 1960s, Bob Moog started the electronic music
revolution. A direct descendant 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.
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
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Moogerfooger Lowpass Filter Controls
Envelope Section
Amount
The Amount knob determines how much the envelope varies the filter. When the knob is counterclockwise,
the envelope signal has no effect on the filter. When the knob is fully clockwise, the envelope signal opens
and closes the filter over a range of five octaves.
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.
The 2/4 pole switch selects the filter slope
Resonance
Resonance changes the way the filter sounds. At low resonance, low frequencies come through evenly. At
high resonance, frequencies near the cutoff frequency are boosted, creating a whistling or vowel-type
quality. When resonance is maxed out, the filter oscillates and produces its own tone at the cutoff
frequency. This oscillation interacts with other tones as they go through the filter, producing the signature
Moog sound.
2-Pole/4-Pole
The 2-Pole/4-Pole switch selects whether the signal goes through half the filter (2-pole) or the entire filter
(4-pole). 2-pole is brighter, while 4-pole has a deeper, mellow quality.
Drive
The Drive control sets the input gain. Use it to adjust the input to the filter and envelope follower.
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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

Try inserting an LFO ahead of the Moogerfooger Lowpass Filter to produce a cool “auto wah” effect. Use
this in conjunction with Voce Spin’s rotating speaker for a swirling auto-wah effect.
Moogerfooger 12-Stage Phaser

Moogerfooger 12-Stage Phaser combines a 6- or 12-stage phaser with a wide-ranging variable LFO. Start
with subtle tremolo or radical modulation effects, then crank the distortion and resonant filters for
unbelievable new tones—all featuring classic MOOG sound.
Moogerfooger 12-Stage Phaser is available in DSP, Native, and AudioSuite formats.
Moogerfooger 12-Stage Phaser supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample
rates.
Moogerfooger 12-Stage Phaser operates as a mono, multi-mono, or stereo plug-in.
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.
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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.
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.
Moogerfooger 12-Stage Phaser Controls

LFO Section
Control the LFO using the Amount and Rate knobs and the Lo/Hi selector switch.
Amount
Amount varies the depth of phaser modulation, from barely perceptible at the full counterclockwise
position, to the full sweep range of the phaser (full clockwise or “Kill” setting).
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.
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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.
Responses of a phaser with high resonance
Sweep
Sweep adjusts the center frequency point of the filters. Use it in conjunction with Amount to control the
frequencies affected by the phaser.
Sweep adjusts the center frequency point
Drive
LED Indicators
Three LEDs provide visual feedback.
Level
Level glows green when signal is present.
LFO
LFO blinks to show the LFO rate.
Bypass
signal path.
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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 and Extreme

Dr. Moog apparently took these mantras of early 21st Century recording science to heart when he designed
the Rate knob on his phaser. Flick the Rate switch to Hi and let the party begin. Try muting a track and
mixing in bits of extremely phase-swept material.
It’s an Effect—Play with It!

All the controls on the Moogerfooger 12-Stage Phaser are fully independent of one another. This means you
can set them in any combination that you wish. There is no such thing as a “wrong” combination of
settings, so you can experiment all you like to find new, exciting effects for your music.
Moogerfooger Ring Modulator

Moogerfooger Ring Modulator provides a wide-range carrier oscillator and dual sine/square waveform LFO.
Add motion to rhythm tracks and achieve radical lo-fidelity textures—you set the limits!
Moogerfooger Ring Modulator is available in DSP, Native, and AudioSuite formats.
Moogerfooger Ring Modulator supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample
rates.
The Moogerfooger Ring Modulator operates as a mono, multi-mono, or stereo plug-in.
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 wide-range, dual-waveform (sine/square) oscillator.
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.
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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 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.
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 frequency, the carrier oscillator provides one input to the ring modulator, with the
other coming from the input signal.
Lo
In the Lo position, the Frequency knob ranges from 0.5 Hz to 80 Hz.
Hi
In the High position, the Frequency knob ranges from 30 Hz to 4 kHz.
Mix
The Mix control blends the input signal and the Ring Modulator output. You hear only the input signal when
the knob is counterclockwise, and only the ring modulated signal with the knob fully clockwise.
Drive
LED Indicators
Three LEDs provide visual feedback.
Level
Level glows green when signal is present.
LFO
LFO blinks to show the LFO rate.
Bypass
signal path.
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Moogerfooger Ring Modulator Tips and Tricks
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.
Reel Tape Flanger

Reel Tape Flanger is part of the Reel Tape suite of tape-simulation effects plug-ins. 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 is available in DSP, Native, and AudioSuite formats.
Reel Tape Flanger supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Reel Tape Flanger operates as a mono, multi-mono, or stereo plug-in.
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:
l Speed: 15 ips
l Bias: 0.0 dB
l Cal Adjust: +9 dB
Use the BPM Sync feature to synchronize the Reel Tape Flanger effect to the current tempo of the Pro Tools
session.

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Drive
Output
Tape Machine
US
Swiss
Lo-Fi
effect.
Tape Formula
Classic
Hi Output
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.
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Operation with “Flange” Range setting (no offset)
ADT (Artificial Double Tracking)
Range settings outside the narrow center band simulate artificial double tracking, in which the variable
delay does not cross the “zero” point. This varying delay creates a unique doubling effect, essentially an
analog precursor to chorusing. (You can hear ADT-type effects on many classic analog recordings, such as
those of the Beatles or Led Zeppelin.)
Operation with “ADT” Range setting (positive offset)
g When the LFO Depth control is set to zero, you can still achieve a “manual” flanging or ADT effect by
varying the Range control.
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.
Wow/Flutter
The Wow/Flutter control adjusts the amplitude of the variable delay tape machine’s wow and flutter, or the
amount of fluctuation in tape speed. A higher setting results in wider fluctuations in speed. A lower setting
results in narrower fluctuations in speed. Wow/Flutter is adjustable from 0 to 1 percent, with a default value
of 0.03 percent.
Rate
The LFO Rate control adjusts the rate of change in the variable delay. A higher setting results in faster
fluctuations in speed. A lower setting results in slower fluctuations in speed. LFO Rate is adjustable from
0.05 Hz to 5 Hz, with a default setting of 0.14 Hz.
You can set the LFO Rate control to synchronize to the current tempo of the Pro Tools session.
"Synchronizing Reel Tape Flanger to Session Tempo" on the next page
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Depth
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.
Mix
The Mix control adjusts the amount of fixed delay signal mixed with the variable delay signal in the final
output of the plug-in. The default Mix value is adjustable from –100 (all fixed delay signal) to +100 (all
variable delay signal) percent, with a default value of 0 (50% fixed delay, 50% variable delay signals).
Invert
(Plug-In Automation Playlist or Control Surface Access Only) The Invert parameter inverts the polarity of the
signal coming from the variable delay tape machine, so that complete audio cancellation occurs when the
flanger effect crosses the zero point. The default setting for the Invert parameter is Off.
g Settings for this parameter are saved with plug-in presets. If you use a preset for the DSP, Native, or
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.
g Settings for this parameter are saved with plug-in presets. If you use a preset for the DSP, Native, or
Synchronizing Reel Tape Flanger to Session Tempo

You can set the LFO Rate in Reel Tape Flanger to synchronize to the session tempo (in beats per minute).
To synchronize the LFO Rate control setting to the session tempo:
1. In the BPM Sync section, click the On button. The Tempo/Rate display changes to synchronize with
the current session tempo.
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)
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3. To adjust the rhythm further, do any of the following:
t To enable triplet rhythm delay timing, click the Triplet (“3”) button so that it is lit.
t To set a dotted rhythm delay value, click the Dot (“.”) button so that it is lit.
Reel Tape Flanger Tips
l 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.)
l 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).
l 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.
l To add complexity to flanging or ADT effects, turn up the Wow/Flutter control to introduce more
fluctuation in the variable delay.
l 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.
l 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.
Reel Tape Flanger Presets

12-String
Moderate-depth ADT setting that works well with acoustic guitar sounds
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.
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Sci-Fi
Sci-Fi features analog synthesizer-type effects that include:
l Ring modulation
l Frequency modulation
l Variable-frequency, positive and negative resonator
l Modulation control by LFO, envelope follower, sample-and-hold, or trigger-and-hold
Sci-Fi is available in DSP, Native, and AudioSuite formats.
Sci-Fi supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Sci-Fi operates as a mono, multi-mono, or stereo plug-in.
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 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.
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 wah-wah-type effect. When you select the Envelope Follower, the Mod Amount slider
changes to a Mod Slewing control. Slewing provides you with the ability to smooth out extreme dynamic
changes in your modulation source. This provides a smoother, more continuous modulation effect. The more
slewing you add, the more gradual the changes in modulation will be.
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.
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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.
Voce Plug-Ins
The Voce plug-ins provide a pair of vintage modulation effect plug-ins: Voce Chorus Vibrato and Voce Spin.
The Voce plug-ins are available in DSP, Native, and AudioSuite formats.
The Voce plug-ins support 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
The Voce plug-ins support mono, mono-to-stereo (Voce Spin only), multi-mono, and stereo operation.
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.
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 B3 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 B3 sound more like a pipe
organ, it imparts frequency variation to the sound.
Although well received by churches, the signature B-3 Chorus/Vibrato graced jazz and rock recordings ever
since. Now you can use this beautiful effect on any instrument.
Voce Chorus/Vibrato Controls

Simply click the Big Knob to rotate between settings of Vibrato and Chorus. V1 provides the least amount of
vibrato, V2 slightly more, and V3 the most. Likewise the amount of Chorus increases from C1 to C3.
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Option-click (Mac) or Alt-click (Windows) the knob to rotate it in the opposite direction, or click the lettering
to select a specific setting.
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.
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 or Eleven.
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!
Don Leslie invented the rotating speaker in 1937. His design is simple and elegant: an internal 40-watt tube
amplifier feeds a speaker crossover, which splits the signal.
All frequencies below 800 Hz go to a 15” bass speaker and all frequencies above 800 Hz go to a
compression horn driver.
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Lower speaker assembly
The large bass speaker is bolted to the cabinet and a foam drum directly below the speaker reflects the
bass outward.
For the high frequencies, a treble horn with two bells reflects the sound from the compression horn driver
located below.
Only one bell actually produces sound; the other is merely a counterbalance.
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.
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.
n See also "Voce Spin Additional Controls" on the next page.

Spin Presets
l 122
Model 122 speaker, medium pulleys.
l 122 (Small Pulley)
Small pulleys (fastest rotation).
l 122 (Large Pulley)
Large pulleys (slowest rotation).
l 122 (Wide Stereo)
Middle pulleys, wide stereo microphone placement.
l 122 (Mono)
Middle pulleys, one mic each top and bottom.
l 21H
Model 21H speaker.
l Foam Drum
Middle pulleys, microphones close.
l Memphis
Lower drum slow motor unplugged, microphones close.
l Steppenwolf
Lower drum only, loose belts, microphones close.
l Rover (Slow to Fast)
Guitar rotating speaker, maximum speed differential.
l Rover (Slow to Medium)
Guitar rotating speaker, slower variation.
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l Rover (Medium to Fast)
Guitar rotating speaker, faster variation.
l Phaser
Medium rotation rate, microphones very close.
l Watery Guitar
Fast rotation rate, microphones close.
l Speed Options
l Chorale
Slow rotation.
l Tremolo
Fast rotation.
l Off
No rotation, but still through the crossover and speakers (wherever the speakers comes to rest
relative to the microphones!).
Voce Spin Additional Controls
Though the Voce Spin plug-in window contains only the Chorale/Off/Tremolo control, the following controls
are also available:
l Input Trim
l Speed Switch
l Rotor Balance
l Upper Slow Speed
l Upper Accel Rate
l Upper Decel Rate
l Upper Mic Angle
l Lower Fast Speed
l Lower Slow Speed
l Lower Accel Rate
l Lower Decel Rate
l Lower Mic Angle
You can adjust and automate controls 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.
You can access these additional controls through Pro Tools plug-in automation, and / or from a compatible
control surface.
Accessing Voce Spin Controls

Accessing Voce Spin Controls On-Screen
All Voce Spin controls can be adjusted on-screen by editing Pro Tools breakpoint automation data.
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To access additional Voce Spin controls on-screen:
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 a 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:
t Click the Track View selector and select the automation control you just enabled from the Voce
Spin sub-menu.
t 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. Edit the breakpoint automation for the enabled control.
Accessing Voce Spin Controls on a Control Surface
When using a control surface, all Voce Spin controls are available whenever the plug-in is focused. You only
need to enable plug-in automation if you want to record your adjustments as breakpoint automation.
To access additional Voce controls from a control surface:
1. Focus the Voce Spin plug-in on your control surface. All available controls are mapped to encoders,
faders, and switches.
2. Adjust the corresponding control.
3. Use the previous / next Page controls to access additional controls.
g To automate your adjustments, enable automation for that control. For more information on plug-in
automation, see the Pro Tools Reference Guide.
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.
Spin into Moogerfooger Lowpass Filter
Try using the amplitude modulation effects of Spin as an LFO driving the Moogerfooger Lowpass Filter!
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.
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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 was hurt. 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.
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 B3 sound leaks out.
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8 Harmonic Plug-Ins
Harmonic plug-ins affect the harmonic spectrum of a sound, and include amp simulation, tape emulation,
sub-harmonics, and more.

l "Aphex Aural Exciter Type III " below
l "Aphex Big Bottom Pro" on page 293
l "Eleven" on page 296
l "Eleven MK II" on page 323
l "Eleven Effects Harmonic Plug-ins" on page 349
l "Lo-Fi" on page 352
l "Pro Subharmonic" on page 354
l "Recti-Fi" on page 365
l "Reel Tape Saturation" on page 367
l "SansAmp PSA-1" on page 370
Aphex Aural Exciter Type III

Aural Exciter Type III is an AAX plug-in that retains the look and sound of Aphex Systems’ renowned
hardware units. Aural Exciter makes it possible to recreate and restore missing harmonics.
Aphex Systems, Inc. first introduced Aural Exciter in 1975. Since then, several refinements and improvements
have been incorporated into its original design. The Aural Exciter plug-in is modeled after the TYPE III Aural
Exciter. Aural Exciter has become a standard in the professional audio industry, and has been used on
many albums, CDs, movies, broadcast productions, commercials, and concerts. The Aural Exciter plug-in
for Pro Tools continues this tradition of success, and is ready for use with the latest cutting edge music
productions.
Harmonics are musically and dynamically related to the original sound, and reveal the fine differences
between voices and various instruments. Reproduced sound is audibly different from the original live sound
because of the loss in harmonic detail, often sounding dull and lifeless.
Aural Exciter is an audio process that recreates and restores missing harmonics. It actually adds harmonics,
restoring the sound’s natural brightness, clarity and presence, effectively improving detail and intelligibility.
Use Aural Exciter on specific instruments or in the final mix to bring life back to recordings.
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Unlike EQs and other brightness enhancers which only boost the high frequencies that often alter the
overall tonal balance, Aural Exciter extends the high frequencies. The stereo image is enhanced with Aural
Exciter. This results in a greater perceived loudness without an introduction of noise into the audio path,
commonly caused by increased gain.
Aural Exciter is a single-ended process which can be inserted at any point within the audio chain. The input
signal is split into two paths. One path goes to the output unmodified, while the other path, known as a
side-chain, goes through the Aural Exciter, which includes a tunable high-pass filter and a harmonics
generator. Aural Exciter applies frequency-dependent phase shift and amplitude-dependent harmonics.
The output of the Aural Exciter's harmonic generator is mixed back with the unmodified signal, lower in
level.
When used at nominal settings, Aural Exciter does not add significant average level to the original signal.
Even though the added information is low level, the perception is a dramatic increase in mid and high
frequencies.
The Aural Exciter is patented in the United States, Japan and most of Europe. Others may claim they are
doing the same thing, but they can only resort to some form of EQ (amplitude correction or expansion),
phase scrambling and/or filtering. They can only increase peak levels causing clipping, feedback, tape
distortion and listener fatigue.
Aural Exciter is available in DSP, Native, and AudioSuite formats.
Aural Exciter supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Aural Exciter operates as a mono, multi-mono, or stereo plug-in.
Meters
Drive Meter
The Drive meter monitors the peak level to the harmonic generator. It works in conjunction with the Drive
switch.
For optimal performance keep the peak hold meter of the Drive meter inside the yellow area. The harder you
drive the Exciter, the more Exciter enhancement you generate. If you cannot get the Drive meter to register
in the yellow area, try setting the Drive switch to High (Drive switch enabled).
Out Meter
The Out meter lets you monitor the output level after Aural Exciter processing.
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Rotary Controls
Level Control
The Level control sets the attenuation of the input signal. For normal operation set the Level control on Max
(no attenuation).
Aural Exciter has an internal gain structure that boosts +6 dB of the output from the high-pass filter into the
side-chain. The Drive switch further boosts the signal level fed into the harmonics generator. When Drive is
set to Normal, you obtain a boost of +6 dB; in the High position you can get an additional 6 dB of gain, for a
total maximum boost of +18 dB. You can also generate a boost in the high-pass filter section by setting
Peaking to Max.
If you run into headroom problems when adjusting the Mix control, adjust the Level control to generate the
necessary headroom.
Tune Control
The Tune control sets the bandwidth (corner frequency) of the second order high-pass filter in the side-
chain prior to the harmonics generator. The range of the control extends from 700 Hz to 7 kHz. The following
figure shows the range of the Tune control from 700 Hz to 7 kHz with Null Fill set near Min and Peaking set at
the mid-point position.
Notice the interaction that the Peaking and Null Fill controls have on Tune, as well as on each other.
Tune control
Peaking Control
The Peaking control provides a damping effect on the leading frequency edge of the high-pass filter
controlled by Tune. As you vary this control from Min to Max, the Tune frequency becomes more
accentuated, as shown in the following figure. However, at the same time, a dip is created just before the
accentuated Tune frequency. This dip or null becomes larger as Peaking is increased.
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Peaking control
The amount of Peaking is displayed as a percentage.
Null Fill Control

The Null Fill control adjusts the curve of the high-pass filter to “fill in” the null caused by the summing of the
side-chain return signal and the input signal.
The amount of Null Fill is displayed as a percentage.
This control compensates for “phase pulling,” which occurs as a side effect of the time delay present in the
side-chain signal, an important part of the Aural Exciter operating theory. As the time delay “stretches”
transient waveforms to create a perception of louder sound, a “dip” or “null” also occurs in the output
equalization curve at the Tune frequency. As a result, the “null” frequencies are de-emphasized, thus giving
even more emphasis to the higher frequencies. Although this often is a desirable effect, the Null Fill control
was created to allow the user to fill-in the null by a selectable amount for any applications requiring less
emphasis.
The following figure shows three different Null Fill settings with Tune set at the mid-point position. With the
Null Fill control set at Min, there is a noticeable drop in the frequency response, just before the start of the
high-pass shelf boost. At this setting, program material under enhancement would lose some presence.
When the Null Fill control is set at Max, the frequency dip is filled, but the frequencies associated with the
shelf top become accentuated. Also notice the shift in the Tune frequency (0 dB axis) for the range of Null
Fill settings.
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Null Fill control
Harmonics Control
The Harmonics control adjusts the amount of harmonics being generated, which is displayed as a
percentage underneath the controls.
The harmonic generator produces harmonic components according to a complex set of laws, including
considerations for transient and steady-state qualities, as well as relative amplitude of the original audio
signal.
As you move the control up, harmonic content increases proportionally as it works in conjunction with the
Timbre control. Moreover, the amount of harmonics generated is dependent on the input level. The gain of
the harmonics automatically increases as the input level increases.
The generated harmonics are not products of harmonic distortion, since they are intelligently produced and
formed into a power envelope that enhances rather than distorts the final audio signal.
Timbre Control
The Timbre control sets the order or type of harmonic signal being generated by way of the Harmonics
control. The control can be varied from all Even harmonics in the Min position, to all Odd harmonics at the
Max position. Odd order harmonics will sound softer to your ear, while even harmonics will sound harsher.
Varying the Timbre control between the two extremes will provide you with a mix of both Even and Odd
harmonics in proportion to the control position. To emphasize the effect of the Timbre control, set the
Density switch to High.
The Timbre control display ranges from +100% (all Odd) to –100% (all Even).
Mix Control
The Mix control determines the amount of Aural Exciter enhancement mixed into the original signal. The
control ranges from Min (no enhancement), up to Max, representing approximately a 6 dB boost when the
Drive switch is set to Normal, and approximately an 18 dB boost when it is set to High.
The amount of enhancement mixed into the original signal is displayed as a percentage.
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Using Aural Exciter III
In the recording studio, post production suite, or similar environment, post-processing of previously
recorded audio tracks with Aural Exciter can restore lost vibrancy and realism, even to the extent of saving
dialogue or sound effects which were thought to be unusable. Instruments and vocals can be made to stand
out in the mix without substantially increasing the mix levels or using equalization.
The Pro Tools mixing environment is very flexible, offering several ways to route and use Aural Exciter in a
session. This section provides some suggestions for efficient use of Aural Exciter in your Pro Tools sessions.
The exact steps you take to use Aural Exciter’s capabilities will differ depending on the nature of your
session and your specific Pro Tools mixer configuration.
Setting the Gain Structure

If the input material has a very high peak-to-peak level and no additional headroom for Exciter effects, use
the Level fader to adjust the signal level to avoid over-coloring the signal.
When using digital audio as a sound source, such as a CD Player with S/PDIF outputs, there is a very high
peak-to-peak level because the material on the CD is optimized for the best signal-to-noise performance. In
this situation the Level fader can be used to adjust the signal level to gain additional headroom.
In an analog based system you will have the same headroom problem when using a very high peak-to-peak
level signal.
Using the Level fader to adjust for more headroom is also useful when restoring older recordings.
For optimal performance keep the peak hold meter of the Drive meter inside the yellow area. The harder you
drive the Exciter, the more Exciter enhancement you generate. If you cannot get the Drive meter to register
in the yellow area, try setting the Drive switch to High.
Optimizing Aural Exciter Effects

When using Aural Exciter, the output signal level has to be equal to the input signal level plus the enhanced
Exciter effect. The dynamic characteristics of the harmonic generator used in the Aural Exciter plug-in are
based on a complex algorithm that includes the signal peak level, the averaged steady state level, and the
dynamic characteristics.
Unlike an EQ, which adds a constant boost in the high end, Aural Exciter enhancement is added into the
input signal is such a way that the average signal level will be virtually unchanged.
The Level, Tune, Peaking, Null Fill, Harmonics, Timbre and Mix faders provide separate left and right faders
when in stereo. For stereo, a separate set of switches for independent control of the left and right channels
is provided for Ax, Solo, SPR, Bypass, Drive and Density.
The Tune fader adjusts the corner frequency of the high pass filter and the Mix fader varies the amount of
Aural Exciter enhancement that is mixed with the unmodified signal.
Experiment with the Aural Exciter controls to hear how each one enhances the original audio signal.
To experiment with Aural Exciter:
1. Set the Level fader on Max.

2. Set the Drive switch to reflect the current nominal level.
3. Make sure the Bypass switch is deactivated (Bypass light off).
4. Make sure the Ax switch is activated (Ax light on).
5. Set Density to Normal. (Density light off). As you make the following adjustments, alternate the
Density switch between Normal and High to hear the change in the Aural Exciter effect.
6. Put the Aural Exciter Mix fader on Max, making it easier to hear the effect as it changes.
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7. Vary the Tune fader and listen for the frequency range that is being enhanced. The Tune fader can be
used to enhance a particular instrument so it stands out in the mix.
8. Adjust the Harmonics fader and listen for the change in harmonics being added to the original audio
signal.
9. When finished experimenting, set the Mix control to taste. Keep in mind that a little Aural Exciter goes
a long way.
Using the Tune Fader
After a while you’ll get a sense of where you like your Tune setting when using Aural Exciter on individual
tracks. It’s best not to process the same range of frequencies with the Tune fader during the final mix. If you
already processed individual tracks with Aural Exciter, try starting the final mix with the Tune fader in the
maximum position which is approximately 7 kHz. You should get a spacious, three-dimensional mix with an
open “airy” quality.
Using the SPR Switch

The SPR function can produce a useful effect with solo voices (human and instrumental) or mixed programs
(such as drama and music). There is no specific time when SPR should or shouldn’t be used. Experiment with
it on various types of material until you get used to the effect. Listen carefully as you operate the SPR
switch. The effect may be noticed only at certain times (such as specific modulations of a voice or during a
particular instrumental playing style or passage). Don't expect to hear the sound change radically. The SPR
is usually subtle, adding a certain beauty and good feeling to the sound. In time you will find that the SPR
does indeed produce demonstrable results.
For instance, the SPR effect can:

l Drop pitch of ultra low bass
l Increase apparent bass power
l Unmask instruments or sonic details hidden in the mix
l Improve definition of high frequency sounds (such as on cymbals and bells)
l Improve speech articulation and presence
l Increase depth and clarity of male voices
Successful use of the SPR depends on the character of the original sound. It is hard to predict in advance
what will be the effect of the SPR. Typically, you may find it useful about 50% of the time. At other times,
there will be little or no discernible effect. Seldom does the effect damage good audio, so it could be left on
as a matter of course.
A few examples of audio material likely to be helped by the SPR are:

l Highly overdubbed tape tracks
l Live acoustic recordings
l Electronic keyboards
l Productions composed from tape cartridge and cassette sources
l Material recorded with transformer coupled mic preamps
l Vocals recorded with dynamic microphones
l Reverberant live sound or recordings
l Highly equalized material
l Delayed, flanged, or digitally processed material
l Material from broadcast audio reception (such as store casting and muzak)
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Aphex Big Bottom Pro
Aphex Big Bottom Pro is an AAX plug-in that retains the look and sound of Aphex Systems’ renowned
hardware units. Aphex Systems, Inc. first introduced Big Bottom Pro in 1992 as part of the Model 104. Since
then, Big Bottom Pro has become a standard in the professional audio industry, and has been used on
numerous albums, CDs, movies, broadcast productions, commercials, and concerts. The Big Bottom Pro
plug-in for Pro Tools continues this tradition of success, and is ready for use with the latest cutting edge
music productions.
Big Bottom Pro provides more energy to the bass (increasing its sustain and density). It dynamically
contours the bass response of a complex range of shapes in the 40 to 400 Hz range, isolating and
enhancing the lowest frequencies to provide a deeper, more resonant bass. Big Bottom increases the
perception of low frequencies without significantly increasing the maximum peak output.
Big Bottom Pro is a single-ended process which can be inserted at any point within the audio chain. The
input signal is split into two parts. One part goes to the output unmodified, while the other part, known as a
side-chain, goes through Big Bottom Pro. The side-chain consists of a tunable low pass filter followed by a
dynamic processor.
Big Bottom is available in DSP, Native, and AudioSuite formats.
Big Bottom supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Big Bottom Pro operates as a mono, multi-mono, or stereo plug-in.
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Meters
Drive Meter
The input Drive peak meter indicates the actual peak level to the Big Bottom Pro side-chain.
Compression Meter
The Compression (Comp) meter indicates the actual amount of compression taking place in the Big Bottom
Pro side-chain. If the Comp meter is not showing any activity the input level is too low. Adjust the Level and
Drive controls accordingly.
Out Meter
The Output peak meter indicates the actual peak level after mixing the Big Bottom Pro side-chain with the
original input signal.
g Audition the loudest or peak sections of your audio material to avoid Big Bottom Pro output clipping:
Use the Out Meter to check for clipping.
Rotary Controls
Level Control
The Level control sets the attenuation of the input signal. For normal operation set the Level control on Max
(0 dB).
In the event you are not generating enough bass enhancement (even when the Mix control is also set on
Max), lower the Level control. This will give the plug-in more headroom by generating less compression in
the dynamic processor, resulting in a more powerful side-chain signal. If you need more headroom when
adjusting the Mix control, lower the input Level and re-tune the Mix control.
Drive Control
The Drive control sets the sensitivity to the bass generating side-chain. The corresponding Drive meter
shows the actual peak level of the side-chain input. There is a boost in the side-chain signal of +12 dB, when
the Drive control is set to Max.
The Drive control needs to be set at a point where the dynamic processor receives the optimum level
required for Big Bottom Pro to work effectively. To find the optimum level, adjust the Drive control until the
Comp meter displays in the yellow area. Make sure the Drive meter does not indicate clipping.
If the Comp meter is not showing any activity, the input level is too low. Adjust the Level control
accordingly. When the AutoTrace switch is set to the On position, the setting of the Drive control is less
sensitive, and the Big Bottom Pro side-chain affects a wider input range.
In general, higher Drive settings to the side-chain provide better control over peaks, while lower Drive
settings tend to produce a more open sound.
By adjusting both the Drive and Mix controls, you can experiment with the different “colors” or timbral
modifications Big Bottom Pro is able to generate.
Tune Control
The Tune control sets the bandwidth (corner frequency) of the low pass filter in the side-chain prior to the
dynamics processor. The range of the Tune control is from 40–400 Hz.
Aside from the Mix control, this is the most important control on the Big Bottom Pro plug-in.
The Tune control is used to isolate the range of frequencies being enhanced by Big Bottom Pro.
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Mix Control
The Mix control adjusts the amount of the Big Bottom Pro enhancement signal being added to the original
signal. The lower the setting the subtler the effect. The higher the setting, the more dramatic the effect. It’s
important to note that higher settings may increase the peak output.
Switches
In/Out Switch
The In/Out switch gives you the choice of turning the Big Bottom Pro process On or Off. When the switch is
set to the On position, Big Bottom Pro enhancement is sent to the outputs.
The switch illuminates when the Big Bottom Pro effect is activated.
Unlike system bypass, the audio from the input travels through the DSP algorithm on the way to the output
whenever the In/Out switch is set to Off.
Switching back and forth from On to Off provides a quick A/B comparison, allowing you to hear the
enhancements from the Big Bottom Pro effect in your program content.
Solo Switch
When engaged, the Solo switch allows you to audition the Big Bottom Pro side-chain effect without the
main audio signal. The switch illuminates when the Solo switch is activated.
Phase Switch
The Phase switch allows you to alter the phase of the side-chain signal, which contains the Big Bottom Pro
effect, before it is mixed with the original input signal. This function is used as a optional way to change the
“quality” of the Big Bottom Pro effect.
The switch illuminates when the Phase switch is activated.
Altering the side-chain signal’s phase dramatically effects the sound of the Big Bottom Pro enhancement.
With the Phase switch turned Off, you will recognize the Big Bottom Pro effect found in the Aphex Model
104.
As an exclusive feature for this DSP plug-in, we have added the Phase switch. When activated, the Phase
switch alters the Big Bottom Pro effect by setting the side-chain in-phase with the main signal. This
increases the output peak level. Use the Mix or Level controls to restore the output peak level if the Drive
meter indicates clipping.
AutoTrace Switch
Activating the AutoTrace switch enables an automatic threshold function for the compressor within the Big
Bottom Pro side-chain. The AutoTrace function enables the dynamic processor to self-optimize during
normal operation. The switch illuminates when the AutoTrace switch is activated.
This control is particularly useful when you want a subtle Big Bottom Pro effect, or when the peak level of
the input material varies over time. The AutoTrace feature is also ideal for changing the sound
characteristics of the Big Bottom Pro effect. Drive control adjustments will be reduced when the AutoTrace
switch is activated.
Link Switch
The Link switch is for stereo operation only. It links the left and right controls so they work as one. Grab a
control on one page with the cursor and move it to the desired position. The control on the other page
automatically updates. In this way both controls can be set to the exact same position. Stereo controls may
be linked temporarily by holding down the Shift key while adjusting the control.
The switch illuminates when Link is activated.
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LR (Left/Right) Switches
The LR switch is for stereo operation only. It allows you to view or change parameters on one channel at a
time.
The switch for the currently displayed channel illuminates. Clicking the unlit switch changes the display to
the other channel.
To edit both channels simultaneously, click on the Link switch.
Using Big Bottom Pro

By putting Big Bottom Pro to use in a Pro Tools session you will find many creative uses for its powerful
processing capabilities.
The remaining sections provide instructions on how to get the most out of Big Bottom Pro.
Setting the Gain Structure

If the amount of Big Bottom Pro effect is limited by a lack of headroom in the input material, use the Level
control to adjust the signal level to avoid clipping. When using Big Bottom Pro with the Phase switch in the
Off position, it is possible to achieve a substantial increase in bass energy without significantly increasing
the peak level output.
For optimal performance keep the peak hold meter of the Drive meter inside the yellow area.
Optimizing Big Bottom Pro Effects

When using Big Bottom Pro, the output signal level is equal to the input signal levels plus the bass enhanced
Big Bottom Pro effect. The dynamic characteristics of Big Bottom Pro are based on a complex algorithm
that includes the signal peak level, the average steady staid level as well as the dynamic characteristics.
Unlike a bass EQ, which adds a constant boost in the low end, Big Bottom Pro enhancement is added into
the input signal dynamically.
Starting with the factory settings, experiment with the controls on Big Bottom Pro to hear how this plug-in
effects the low-end frequencies of your source material, as follows:
l If the Drive meter is clipping (in the yellow area), adjust the Drive control for optimal operation.
l Activate the Solo switch to listen to only the Big Bottom Pro side-chain effect.
l Vary the Tune control to hear the low-pass filter isolate the low-end bandwidths of the original input
signal.
l De-activate the Solo switch and continue to vary the Tune control until you find the optimal setting.
l Adjust the Mix control to set the amount of Big Bottom Pro effect.
l Use the In/Out switch for an A/B comparison with the output signal and the original input signal.
l Activate the Phase switch and observe the change in the sound characteristics of the Big Bottom Pro
effect. For most applications, leave the Phase switch in the Off position.
l Activate the AutoTrace switch and observe the change in the sound characteristics. Also notice that
the compression level in the dynamic processor, shown by the Comp meter, is affected as well.
l Readjust the Mix control as desired, to experience the benefits of the Big Bottom Pro plug-in.
Remember that a little Big Bottom Pro effect goes a long way.
Eleven
Eleven is a guitar amplifier plug-in that is available in DSP, Native, and AudioSuite formats. Eleven gives you
stunning guitar amplifier, cabinet, and microphone models of the “best of the best” vintage and
contemporary gear.
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Eleven Lite is a free version of Eleven that comes with every Pro Tools system, with a reduced feature set.
Eleven Lite comes in Native and AudioSuite formats only.
Eleven Lite Plug-In Features

l Two custom amp models from Avid
l Two speaker cabinet models
l Amps and cabinets can be mixed and matched
l Noise Gate to control any unwanted noise
l Settings files (presets) to store and recall factory and custom tones
l Support of any compatible work surface or MIDI controller. MIDI Learn provides effortless mapping to
any continuous controller (CC)–capable MIDI device
l Support for sample rates of 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz
l Support for mono or multi-mono operation, in up to 8 channel (7.1) format
g Eleven can share preset data with the Eleven Rack guitar processor/audio interface from Avid. For
more information, see the Eleven Rack User Guide.
Eleven Plug-In Features

l Classic amp models that faithfully recreate the sound and dynamic response of the original amps
l Highly accurate speaker cabinet models with variable speaker breakup (cone distortion)
l Selectable mics, with on- and off-axis options
l Amps, cabs, and mics can be mixed and matched into nearly limitless combinations
l Amps and cabs can be bypassed separately
l All controls can be automated
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l Support of any compatible Ethernet or MIDI controller. MIDI Learn provides effortless mapping to any
continuous controller (CC)–capable MIDI device
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.
l If you are working with pre-recorded guitar tracks, see "Using Eleven with Pre-Recorded Tracks" on
page 302.
l 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.
Basic gain stages to calibrate live guitar input for Eleven
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.)
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To connect your guitar to a Pro Tools host-based system:
1. Do one of the following, depending on your hardware configuration:

t If you are using an interface that has a DI input (such as a Pro Tools | Carbon), plug your guitar
into an available DI input.
t If you are using your computer’s built-in inputs, plug your guitar into an available input.
g 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.
Mbox Pro DI Inputs 1 and 2
c Mbox Pro back-panel 1/4” inputs are line-level only and should not be used with a guitar.
To connect your guitar to a Pro Tools HD system:
1. Make sure you have a pre-amp (such as an Pro Tools | PRE) or similar unit connected to a
Pro Tools HD audio interface (such as a Pro Tools | HD I/O). (Note that Pro Tools | 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.
g 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 you are using a PRE, you can plug your guitar directly into the front panel Line/Inst 1 input,
then set its source to Inst.
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.
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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).
Input 1 Gain on Mbox Pro
To prepare your guitar and 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.
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).
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.
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One audio track for input calibration on Pro Tools
4. Record enable the audio track, or enable its TrackInput monitoring button (Pro Tools Ultimate and
Studio software only).
Set Up Eleven
Use Eleven’s Input LED to make final gain adjustments and complete the input calibration process.
To calibrate your input signal to the Eleven plug-in:
1. Open the Eleven plug-in window by clicking its insert slot. Leave it at its default settings.
Eleven’s Input LED (top) and Clip LED (bottom)
2. Strum as hard as you can a few more times and watch Eleven’s Input LED to see where your level
registers. The Input LED lights green, yellow, orange, or red to indicate the following level ranges:
– 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.
– 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.
c 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 305.
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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.
n See "Processing Pre-Recorded Tracks Through Eleven" on page 315 for more information.
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 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.
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 310.)
4. Refer to "Using Eleven" below for details on Eleven’s main controls, and for suggested track setups for
recording and mixing.
g Use the Settings menu to save, copy, paste, and manage plug-in settings files. To save a setting, see
"Eleven Settings (Presets)" on page 304.
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:

t Click an Insert selector on the track and choose Eleven or Eleven LE.
Channel Formats
Eleven is available as a mono or multi-mono plug-in only. For use in stereo or greater formats choose the
multi-mono version.
Sample Rates
Eleven supports 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz sample rates.
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Category and Manufacturer
When Pro Tools plug-ins are organized by Category or Manufacturer, Eleven is listed as follows:
l Category (Harmonic)
l Manufacturer (Avid)
Adjusting Eleven Controls
This section tells you how to adjust controls using your mouse, a Pro Tools controller, or with a MIDI device.
Navigating the Amp, Cab, and Mic Type Selectors
You can click on the name of the current Amp Type, Cab Type, or Mic Type to display their pop-up menus
and select an item.
You can also click the Previous / Next arrows to step through Amp, Cabinet, and Mic choices one at a time.
Previous arrow (top) and Next arrow (bottom) (Amp Type shown)
g You can control the Amp, Cab, and Mic Type selectors with MIDI. See "Using MIDI and MIDI Learn with
Eleven" below.
Groups and Linked Plug-In Controls
Eleven’s parameters can follow Pro Tools Groups (Mix, Edit, or Mix/Edit) for linked control of multiple inserts.
For more information, see the Pro Tools Reference Guide.
Using Automation
All of Eleven’s parameters can be automated. When a parameter has been enabled for automation, an LED
appears lit near that control.
n See the Pro Tools Reference Guide for more information on plug-in automation.
Using a Controller with Eleven
Eleven can be controlled directly from any compatible Pro Tools controller. Eleven appears along with other
plug-ins and can be assigned, edited, bypassed and automated using the Insert section as available on the
particular controller being used.
About Unused Controls and Controllers

Some amps that have relatively few controls (such as the Tweed Lux) will display controls on a controller
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 controller.
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.
Using MIDI and MIDI Learn with Eleven

Eleven supports MIDI Control Change (CC) messages, meaning that the Master section, amp, cabinet and
mic parameters can be controlled remotely by any CC-capable MIDI device. This includes MIDI controllers,
mixers, and instruments, as well as the 003® (in MIDI Mode).
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Settings files (presets) for Eleven do not store or recall MIDI Learn assignments.
To map a MIDI controller to a parameter:
4. Set the output of the MIDI track to Eleven.
5. Right-click on any control in Eleven, and do one of the following:
menu.
6. Right-click on any control in Eleven.
Right-clicking for MIDI Learn
g On Mac, you can Control-click an Eleven parameter to show the MIDI Learn menu. Note that you

Eleven Settings (Presets)
You can pick a preset from the plug-in Librarian menu.
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To load a preset:
t Click the Librarian menu and select an available Settings file.
Plug-In controls for Eleven Settings files
You can save, import, copy, paste, and manage settings using the Settings menu.
To save your settings as an Eleven preset:

t Click the Settings menu and choose Save Settings. Name the preset, choose a location, and click
Save.
You can scroll through and select pre-configured Eleven Settings files (presets) using the plug-in Librarian
menu, and the +/– buttons.
n For more information on Settings files and folders, see the Pro Tools Reference Guide.
Master Section
The Master section includes plug-in I/O (Input / Output) and noise gate controls, the Amp Type selector and
the Cab Type selector.
The Master section doesn’t change when you switch amps. Master section settings are stored and recalled
with plug-in presets.
Master section
Input LED
The Input LED shows green, yellow, orange, or red to indicate whether you are under- or over-driving 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 322.
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).
g To learn more about the Input control, see "Eleven Signal Flow Notes" on page 322
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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.
g 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" below).
Cab Type
This selector lets you select which speaker cabinet model to use (see "Eleven Cabinet Types" on page 309).
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.
g For suggested gate applications, see "Using the Noise Gate" below. For details on where it derives its
key (trigger) and applies its gate, see "Eleven Signal Flow Notes" on page 322.
Using the Noise Gate
You can use the Noise Gate to silence unwanted signal noise or hum, or just for an effect.
To use the Noise Gate to clean up unwanted, low level noise:
1. Connect and calibrate your guitar as explained in "Connect your Guitar and Configure Source Input"
on page 298.
2. 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.
3. To make it easier to hear the effect, begin by setting the Release to its middle (12 o’clock) position.
4. 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).
5. Slowly lower the Threshold control until the Gate opens again to find the cutoff (or, threshold) of the
noise.
6. 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.
7. If you lowered the Release setting as suggested in step 3, make sure to return it to its maximum
setting (fully clockwise) before continuing.
Amp Types
The Amp Type selector lets you choose an amp.
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Choosing an amp from the Amp Type selector
Available Amp Types in Eleven include the following:

l ’59 Tweed Lux *
l ’59 Tweed Bass *
l ’64 Black Panel Lux Vibrato *
l ’64 Black Panel Lux Normal *
l ’66 AC Hi Boost *
l ’67 Black Panel Duo *
l ’69 Plexiglas – 100W *
l ’82 Lead 800 – 100W *
l ’85 M-2 Lead *
l ’89 SL-100 Drive *
l ’89 SL-100 Crunch *
l ’89 SL-100 Clean *
l ’92 Treadplate Modern *
l ’92 Treadplate Vintage *
l DC Modern Overdrive
l DC Vintage Crunch
* These models only appear in the full version of Eleven.
g Eleven is not affiliated with, or sponsored or endorsed by, the makers of the amplifiers emulated in the
product.
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.
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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.
Bright
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 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 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.
g All Eleven controls provide identical ranges as the original amps, but some numbers have been
adjusted for consistency.
Gain 2
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.
l 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.
l Tone
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.
l Bass
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.
l Middle
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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.
l 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.
Presence
The Presence control provides a small amount of boost at frequencies above the treble control. Presence is
applied at the end of each amp model pre-amp stage, acting as a global brightness control that is
independent of other tone controls. The default setting is 3.0. The Presence range is from 0 to 10.
Master
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.
g Some might assume a Master volume knob iwould be capable of silencing the amp completely. Not so.
Use the Output knob (in the Master section) to silence the output of the plug-in. Use Master volume for
tone and distortion.
Tremolo
Tremolo is achieved through the use of amplitude modulation, multiplying the amplitude of the pre-amp
output by a waveform of lower frequency. Tremolo is not available on all amps.
l 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.
c Eleven does not support Tempo Sync.

l 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.
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:
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l 1x12 Black Panel Lux *
l 1x12 Tweed Lux *
l 2x12 AC Blue *
l 2x12 Black Panel Duo *
l 4x10 Tweed Bass *
l 4x12 Classic 30
l 4x12 Green 25W
* These models only appear in the full version of Eleven.
Cabinets are listed by their number and diameter of their speakers. For example, “1x12” means a cabinet
has a single 12-inch speaker.
g Eleven is not affiliated with, or sponsored or endorsed by, the makers of the loudspeakers and
cabinets that are emulated in the product.
n Visit the Avid website (www.avid.com) to learn about each of the cabinets used to create Eleven.
Pairing Amps and Cabinets
Eleven lets you combine amps and cabinets in traditional pairings (such as the combo amps through their
default cabinets) and non-traditional match ups.
Some of the amps modeled in Eleven are “combo” amps. Combo amps have both their amp and speaker
housed in the same physical box, meaning there is one and only one cabinet associated with the signature
sound of a combo amp. The Tweed Lux and AC Hi Boost are both examples of combo amps.
Other amps are amps-only (heads), and were designed to be run through a speaker cabinet. Many
amp/cab pairings have become standards.
Using Settings for Realistic and Classic Pairings
You can use Eleven’s factory Settings files (presets) for combo amps and classic combinations. Settings files
store and recall all controls, (including Amp and Cabinet Type).
For combo amps and default combinations:

t 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):

t Click and choose from the Amp Type and Cabinet Type selectors to create new pairings.
g Use the Settings menu to save new combinations and build your own custom library (see "Eleven
Settings (Presets)" on page 304).
Eleven Cabinet Controls

All cabinets provide Bypass, Speaker Breakup, Mic Type, and Position controls.
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Cabinet Bypass
The Bypass switch in the Cabinet section lets you bypass cabinet and microphone processing. When in the
Bypass position, no cabinet or microphone processing is applied to the signal. When in the On position,
cabinet and microphone settings are applied.
Speaker Breakup
(Full version, HDX Only) The Speaker Breakup slider lets you specify how much distortion is produced by the
current speaker model. Increasing the Speaker Breakup setting adds distortion that is a combination of
cone breakup and other types of speaker distortion (emulated by the speaker cabinet model). Range is from
1 to 10.
Below certain frequencies, the speaker cone vibrates as one piece. Above those frequencies (typically
between 1 kHz and 4 kHz), the cone vibrates in sections. By the time a wave travels from the apex at the
voice coil out to the edge of the speaker cone, a new wave has started at the voice coil. The result is comb
filtering and other anomalies that contribute to the texture of the overall sound.
g When enabled, Speaker Breakup draws additional CPU resources.

Mic Type
The Mic Type selector lets you choose the microphone to use with the selected cabinet.
Mic Type selector in the Cabinet section
Available Mic Types include the following:

l Dynamic 7
l Dynamic 57
l Dynamic 409
l Dynamic 421
l Condenser 67
l Condenser 87
l Condenser 414
l Ribbon 121
g Eleven is not affiliated with, or sponsored or endorsed by, the makers of the microphones that are
emulated in the product.
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Mic Axis
When capturing the sound of a speaker cabinet in a studio, sound engineers select different microphones
and arrange them in different placements to get a particular sound. For example, a mic can be pointed
straight at a speaker or angled slightly off-center, in order to emphasize (or de-emphasize) certain
frequencies that the mic picks up.
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.
Mic Axis switch in the Cabinet section
About Mic Placement
All Eleven cabinets and mics were close mic’d (whether on- or off-axis). This provides the purest tones
possible, of any room tone or ambiance specific to the Eleven recording environment.
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:
l "Recording Dry: Monitor Through Eleven" below.
l "Recording Wet: Record Eleven-Processed Track to Disk" on the next page.
l "Recording Dry and Eleven Simultaneously" on page 314.
l "Processing Pre-Recorded Tracks Through Eleven" on page 315
l "Blending Eleven Cabinets and Amps" on page 316.
Recording Dry: Monitor Through Eleven
This workflow lets you record dry (clean) while the recorded signal is processed through Eleven, letting you
hear it but without committing the track to that tone forever.
The flexibility to audition and compare different settings and combinations of amps, cabinets and
microphones is a very creative and powerful tool for mixing and arranging.
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To record dry and monitor through Eleven:
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 302).
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 304).
5. Record enable the track, or enable TrackInput monitoring (Pro Tools Ultimate and Studio only) and
check your levels.
6. 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.
Recording Wet: Record Eleven-Processed Track to Disk
In this workflow, the audio output of Eleven is recorded to disk while tracking. Usually, no additional dry
track is recorded.
This method commits your track to the original Eleven tone used while tracking. It requires two tracks (an
Auxiliary Input and an audio track), but after tracking, the plug-in can be deactivated or removed to up
processing resources.
To record guitar with Eleven while playing:
1. Choose Track > New.

2. Configure a new track by doing the following:
a. Create one mono Auxiliary Input track.
b. Click the Add Row button (+).
c. Create one mono audio track.
d. Click Create.
3. In the Mix (or Edit) window, configure the Aux Input by doing the following:
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a. Click the Input selector and choose your guitar input (the audio interface input your guitar is
plugged in to).
b. Click the Output selector and choose Bus 1.
c. Click the Insert selector and select Eleven.
4. Configure the audio track by doing the following:
a. Click the Input selector and choose Bus 1.
b. Record enable the audio track.
Recording Eleven (printing its output)
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 output from Eleven is recorded to disk. If you need to conserve DSP or Native processing resources, you
can remove or deactivate Eleven after recording.
Recording Dry and Eleven Simultaneously
You can record a dry, unprocessed track and an Eleven-processed track simultaneously.
This method gets the best of both worlds by tracking dry (to experiment with tones later) and at the same
time recording the tone used on the original tracking session. It requires two audio track, as follows:
To record guitar dry and with Eleven live:

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:
plugged in to).
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d. Record enable the audio track.
4. Configure the second audio track by doing the following:
5. Make sure you are not overloading your input signal by checking levels in all tracks and Eleven's Input
LED.
The dry guitar is recorded to the first audio track, processed through Eleven, then bussed to the second
audio track and recorded to disk.
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.

2. Assign the audio track Output to Bus 1 (or Bus 1-2 if working with stereo material).
3. Create an Aux Input track, and configure it by doing the following:
a. Click its track Input selector and choose Bus 1 (or Bus 1-2).
b. 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:
t Try different Settings files (presets) to get your basic amp/cab/mic combination.
t Adjust amp controls.
t Try different cabinets and varying amounts of Speaker Breakup.
t 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.
To process and re-record tracks through Eleven:

2. Configure the source audio track by doing the following:
a. Assign the audio track Output a bus (such as Bus 1 if mono, or Bus 1-2 if stereo).
3. Choose Track > New and create one mono audio track.
4. Configure the new audio track by doing the following:
a. Click its track Input selector and choose the Bus 1 (or Bus 1-2).
5. Record enable the new audio track (or enable TrackInput monitoring if using Pro Tools Ultimate and
Studio).
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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.
Blending Eleven Cabinets and Amps
You can use Eleven for multi-cabinet and multi-amp 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.
To blend multiple cabinets:
1. Choose Tracks > New.

a. Create one mono Audio Track.
b. Click the Add Row button.
c. Create three mono Aux Inputs.
d. Click Create.
3. In the Mix or Edit window, configure the audio track by doing the following:
a. Click the audio track Input selector and choose your guitar input (the audio interface input
your guitar is plugged in to).
4. Select all three Aux Input tracks by Shift-clicking their Track Name displays (make sure your audio
track isn’t still selected). This lets you work with the three Aux tracks “as one” in the next few steps.
Three tracks, selected
5. Hold Option+Shift (Mac) or Alt+Shift (Windows) while doing each of the following:
a. Choose Bus 1 from the Input selector of any of the three selected Aux Inputs.
b. Click the Insert selector of any of the three and select Eleven.
c. 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.
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Setup for blending cabinets
9. Click to open the Eleven plug-in window on the first Aux Input, and do any of the following:
t Choose a cabinet.
t Choose a mic and its position.
t Adjust Speaker Breakup.
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.
13. When you have set your cabinet tones, make sure to unsolo 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:
a. Balance the tracks using the volume faders on the Aux Input tracks.
b. Try different pan positions for each Aux Input track.
c. Evaluate the phase relationships of the combined signals and adjust accordingly (see "Phase
Considerations with Blending in Eleven" on page 320).
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.
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.
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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.
g Different amps can also have a different number of stages, which can affect polarity. See "Phase
Considerations with Blending in Eleven" on page 320 for more information.
How Do I Use This?
Here are a few suggested ways you can pair Eleven’s amps and cabinets:
l To accurately capture the sound of one amp split to and driving multiple cabinets, make sure the
same Amp Type is selected in all the Eleven plug-ins (all the cabinets as well as the active amp).
l For maximum variety, mix and match bypassed amps with active cabinets.
l For realism with the combo amps (such as the Tweed Lux and AC Hi Boost), make sure to use their
default cabinets.
Blending Eleven Amps
You can easily set up tracks and Eleven for multi-amp setups.
To blend multiple amps:
1. Set up tracks and signal routing as explained in the previous workflow (see "To blend multiple
cabinets: " on page 316).
2. Remove (or simply bypass) the Eleven plug-in on the source input/track.
g 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.
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Setup for blending amps

4. Click to open the Eleven plug-in window on the soloed Aux Input, and do any of the following:
t Make sure the amp and cabinet are active (not bypassed).
t Choose a preset (Settings file).
t Pair any amp with any cabinet.
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 un-solo all the Aux Inputs.
8. Continue playing so you can hear the combined tone of all the amps.
b. Try different pan positions for each Auxiliary Input track.
10. Evaluate the phase relationships of the combined signals and adjust accordingly (see "Phase
Considerations with Blending in Eleven" on the next page).
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Phase Considerations with Blending in Eleven
When multi-tracking guitar, experienced engineers know how to identify and take advantage of the phase
relationships that exist between different signals. Adjusting phase is not just a corrective technique either,
it’s also a powerful creative technique for tone, as well as for special effects.
You can use the TimeAdjuster plug-in to flip phase and to adjust timing in single-sample increments, as
described in the next sections.
Flipping Phase (Polarity)
Electric guitar is often recorded to more than one track, such as one dry or DI track, plus one or more tracks
of a mic’d amp. The different signal paths of direct tracks versus mic tracks affect the timing relationships
of the audio. Depending on the signal chain of each track, the signals can get so out of alignment that they
nearly cancel each other out.
Sending a single source track through multiple, unique amps can pose an additional challenge in that each
tube stage in an amp usually inverts the signal. So, depending on whether the number of tube stages in an
amp is odd or even, that amp will either be inverting or non-inverting, respectively. If you send an identical
signal to two amps and one is inverting while the other is non-inverting, signal cancellation will result. All
amps in Eleven accurately model the number of amp stages found in all the original hardware.
If you want to keep it simple and be able to experiment with phase flip, do the following.
To use the TimeAdjuster plug-in to flip phase when blending amps or cabinets:
1. Configure your audio track and Aux Inputs as instructed in "Blending Eleven Cabinets and Amps" on
page 316. 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.
Tweaking Phase
If each of the mics used on a single cabinet are not positioned carefully, comb filtering and other frequency
anomalies can occur. With real amps, the engineer moves one or more mics to find their optimal positions
relative to the source, and to each other.
To hear the effect of small adjustments to the phase relationships of signals, do the following.
To use the Time Adjuster plug-in to control phase:
1. Configure your audio track and Aux Inputs as instructed in "Blending Eleven Cabinets and Amps" on
page 316. Make sure each Aux Input has an Eleven plug-in followed by a TimeAdjuster (short) plug-in.
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.
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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:
l Soft (or clean) tone for the verse, kick in the distortion for the chorus.
l Using tremolo during the intro and the bridge.
l Doubling the rhythm track halfway through the verse to build momentum.
Pro Tools automaton is the key to these and other techniques:
l 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.
l 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.
n See the Pro Tools Reference Guide to learn about Snapshot automation, Glide, and other
automation features.
l 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 318 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 313.
Or, use the AudioSuite version to print Eleven tracks to disk. AudioSuite is especially useful when you’re
processing loops or other shorter-form guitar material.
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.
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:
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l 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).
l 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.
Eleven Signal Flow Notes
Signal flow through Eleven
Plug-Ins are Pre-Fader

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.
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.)
Input Knob and Amp Gain

Eleven actually gives you two separate input gain stages to the plug in:
l The Input knob in the Master section, which affects the signal level before entering the amplifier
model.
l 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.
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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, it silences amp output
to the speaker cabinet.
Eleven MK II
Avid Eleven MK II is a guitar amplifier plug-in that is available in DSP, Native, and AudioSuite formats. Eleven
MK II gives you stunning guitar amplifier, cabinet, and microphone models of the “best of the best” vintage
and contemporary gear. Additionally, Eleven MK II Cab is a speaker cabinet modeling plug-in that makes it
easy to blend cabinets with Eleven MK II amp modeling.
Eleven MK II Plug-In Features

l Classic amp models that faithfully recreate the sound and dynamic response of the original amps
l Amps, cabs, and mics can be mixed and matched into nearly limitless combinations
l Amps and cabs can be bypassed separately
l All controls (except Link) can be automated
l For general information on using plug-ins with Pro Tools, see the Pro Tools Reference Guide.
l Eleven MK II Cab Plug-In Features
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l All controls can be automated
Eleven MK II Cab Plug-In

The Eleven MK II Cab plug-in provides all of the features of the Cabinet section of the Eleven MK II plug-in.
Use the Eleven MK II Cab plug-in on mono Auxiliary Input (or audio) tracks fed by the Auxiliary Output Send
of an instance of the Eleven MK II plug-in on a mono audio (or Auxiliary Input) track for blending different
speaker cabinets from a single amp source.
Eleven MK II Input Calibration and QuickStart

This section shows you how to get connected, calibrated, and cranking through Eleven MK II as quickly as
possible.
Before You Begin with Eleven MK II

Eleven MK II 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.
l If you are working with pre-recorded guitar tracks, see "Using Eleven MK II with Pre-Recorded Tracks"
on page 328.
l 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 MK II, its amps, and its factory presets.
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Basic gain stages to calibrate live guitar input for Eleven MK II
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:

t If you are using an interface that has a DI input (such as a Pro Tools | Carbon), plug your guitar
into an available DI input.
t If you are using your computer’s built-in inputs, plug your guitar into an available input.
g 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.
Mbox Pro DI Inputs 1 and 2
c Mbox Pro back-panel 1/4” inputs are line-level only and should not be used with a guitar.
To connect your guitar to a Pro Tools | HDX or HD Native system:
1. Make sure you have a pre-amp (such as an Pro Tools | PRE) or similar unit connected to a
Pro Tools HD audio interface (such as a Pro Tools | HD I/O). (Note that Pro Tools | 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.
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g 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 you are using a PRE, you can plug your guitar directly into the front panel Line/Inst 1 input,
then set its source to Inst.
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 MK II.
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).
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).
Input 1 Gain on Mbox Pro
To prepare your guitar and 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.
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).
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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 MK II.
One audio track for input calibration on Pro Tools
4. Record enable or enable TrackInput monitoring the audio track (Pro Tools Ultimate and Studio
software only).
Set Up Eleven MK II
Use Eleven MK II’s Input LED to make final gain adjustments and complete the input calibration process.
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To calibrate your input signal to the Eleven MK II plug-in:
1. Open the Eleven MK II plug-in window by clicking its insert slot. Leave it at its default settings.
Eleven MK II’s Input LED (top) and Clip LED (bottom)
2. Strum as hard as you can a few more times and watch Eleven MK II’s Input LED to see where your
level registers. The Input LED lights green, yellow, orange, or red to indicate the following level
ranges:
– 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.
– 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 MK II’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 MK II’s Master section to raise or lower the plug-in output signal.
c Proper input calibration of live guitar does not require any adjustment of Eleven MK II’s Input control.
To learn how this control was designed to work with the amp models, see "Input " on page 332.
Using Eleven MK II with Pre-Recorded Tracks

If the pre-recorded tracks were not calibrated with the Eleven MK II plug-in using the method previously
described, you can use the Input control in Eleven MK II 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.
n See "Processing Pre-Recorded Tracks Through Eleven MK II" on page 342 for more information.
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Getting Started Playing Music with Eleven MK II
To get started playing music with Eleven MK II:
1. Make sure you already calibrated your input signal as explained in the previous sections of this
chapter.
2. Click the plug-in Librarian menu and choose a factory preset, then play guitar. Take your time to
explore — the Presets let you hear all of Eleven MK II’s different amps and combos.
Settings menu (left) and Librarian menu (right)
3. Pick any amp and cabinet from the available types (see "Pairing Amps and Cabinets" on page 337.)
4. Refer to "Using Eleven MK II" below for details on Eleven MK II’s main controls, and for suggested
track setups for recording and mixing.
g Use the Settings menu to save, copy, paste, and manage plug-in settings files. To save a setting, see
"Eleven MK II Settings (Presets)" on page 331.
Using Eleven MK II
The following sections introduce you to the main sections and controls in Eleven MK II 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 MK II.
Inserting Eleven MK II on Tracks

Eleven MK II can be inserted on Pro Tools audio, Auxiliary Input, Master Fader, or Instrument tracks.
To insert Eleven MK II on a track:

t Click an Insert selector on the track and choose Eleven MK II.
Channel Formats
Eleven MK II is available as a mono or multi-mono plug-in only. For use in stereo or greater formats choose
the multi-mono version.
Sample Rates
Eleven MK II supports 44.1 kHz, 48 kHz, 88.2 kHz and 96 kHz sample rates.
Category and Manufacturer
When Pro Tools plug-ins are organized by Category or Manufacturer, Eleven MK II is listed as follows:
l Category (Harmonic)
l Manufacturer (Avid)
Adjusting Eleven MK II Controls
This section tells you how to adjust controls using your mouse, a Pro Tools controller, or with a MIDI device.
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Navigating the Amp, Cab, and Mic Type Selectors
You can click on the name of the current Amp Type, Cab Type, or Mic Type to display their pop-up menus
and select an item.
You can also click the Previous / Next arrows to step through Amp, Cabinet, and Mic choices one at a time.
Previous arrow (top) and Next arrow (bottom) (Amp Type shown)
g You can control the Amp, Cab, and Mic Type selectors with MIDI. See "Using MIDI and MIDI Learn with
Eleven MK II" below.
Groups and Linked Plug-In Controls
Eleven MK II’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 MK II’s parameters can be automated. When a parameter has been enabled for automation, an
LED appears lit near that control.
n See the Pro Tools Reference Guide for more information on plug-in automation.
Using a Controller with Eleven MK II
Eleven MK II can be controlled directly from any compatible Pro Tools controller. Eleven MK II appears along
with other plug-ins and can be assigned, edited, bypassed and automated using the Insert section as
available on the particular controller being used.
About Unused Controls and Controllers
Some amps that have relatively few controls (such as the Tweed Lux) will display controls on a controller
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 controller.
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.
Using MIDI and MIDI Learn with Eleven MK II

Eleven MK II 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).
Settings files (presets) for Eleven MK II do not store or recall MIDI Learn assignments.
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To map a MIDI controller to a parameter:
4. Set the output of the MIDI track to Eleven MK II.
5. Right-click on any control in Eleven MK II, and do one of the following:
menu.
6. Right-click on any control in Eleven MK II.
Right-clicking for MIDI Learn
g On Mac, you can Control-click an Eleven MK II parameter to show the MIDI Learn menu. Note that you

Eleven MK II Settings (Presets)
You can pick a preset from the plug-in Librarian menu.
To load a preset:
t Click the Librarian menu and select an available Settings file.
Plug-In controls for Eleven MK II Settings files
You can save, import, copy, paste, and manage settings using the Settings menu.
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To save your settings as an Eleven MK II preset:
t Click the Settings menu and choose Save Settings. Name the preset, choose a location, and click
Save.
You can scroll through and select pre-configured Eleven MK II Settings files (presets) using the plug-in
Librarian menu, and the +/– buttons.
n For more information on Settings files and folders, see the Pro Tools Reference Guide.
Master Section
The Master section includes plug-in I/O (Input / Output) and noise gate controls, the Amp Type selector and
the Cab Type selector.
The Master section doesn’t change when you switch amps. Master section settings are stored and recalled
with plug-in presets.
Input LED
The Input LED shows green, yellow, orange, or red to indicate whether you are under- or over-driving 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 MK II signal chain, see "Eleven MK II Signal Flow Notes" on page 348.
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 MK II.
The setting of the Input knob is saved and restored with Settings files (presets).
g To learn more about the Input control, see "Eleven MK II Signal Flow Notes" on page 348
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.
g When you want to adjust Eleven MK II’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 the next page).
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Link
The Link button links the Cab type to follow the selected Amp type. When enabled, the Link button turns
orange and the Cab type follows changes to the Amp type. If the Cab type is changed, the Amp type does
not change, but if the Cab type doesn’t match the Amp type, the Link button turns blue.
Cab Type
This selector lets you select which speaker cabinet model to use (see "Eleven MK II Cabinet Types" on
page 336).
Gate
l 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.
l 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.
g For suggested gate applications, see "Using the Noise Gate" below. For details on where it derives its
key (trigger) and applies its gate, see "Eleven MK II Signal Flow Notes" on page 348.
Using the Noise Gate
You can use the Noise Gate to silence unwanted signal noise or hum, or just for an effect.
To use the Noise Gate to clean up unwanted, low level noise:
1. Connect and calibrate your guitar as explained in "Connect your Guitar and Configure Source Input"
on page 325.
2. 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.
3. To make it easier to hear the effect, begin by setting the Release to its middle (12 o’clock) position.
4. 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 MK II).
5. Slowly lower the Threshold control until the Gate opens again to find the cutoff (or, threshold) of the
noise.
6. Raise the Threshold control again slightly, increasing it only enough to once again silence the noise
(hold Command (Mac) or Control (Win) while adjusting to be able to fine-tune the setting in tenths of
a dB). Now you’re in the ballpark.
7. If you lowered the Release setting as suggested in step 3, make sure to return it to its maximum
setting (fully clockwise) before continuing.
Amp Types
The Amp Type selector lets you choose an amp.
Available Amp Types in Eleven MK II include the following:

l ‘59 Tweed Lux
l ‘59 Tweed Bass
l ‘64 Black Panel Lux Vibrato
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l ‘64 Black Panel Lux Normal
l ‘64 Black Vib
l ‘64 Black SR (based on the 1965 Fender Super Reverb combo amp)
l ‘64 Black Mini
l ‘64 J45 (based on the 1965 Marshall JTM45 head)
l ‘66 AC Hi Boost
l ‘67 Black Panel Duo
l ‘67 Plexiglas Vari (based on the 1967 Marshall Super Lead “Plexi” head with Variac modification)
l ‘68 Plexiglas - 50W (based on the 1968 Marshall Super Lead 50W head)
l ‘69 Plexiglas - 100W
l ‘69 Blue Line Bass (based on the 1969 Ampeg SVT head)
l ‘82 Lead 800 - 100W
l ‘85 M-2 Lead
l ‘89 SL-100 Drive
l ‘89 SL-100 Crunch
l ‘89 SL-100 Clean
l ‘92 Treadplate Modern
l ‘92 Treadplate Vintage
l ‘93 MS-30 (based on the 1993 Matchless D/C-30 combo amp)
l ‘97 RB-01b Red (based on the Bogner Ecstasy 101B head)
l ‘97 RB-01b Blue (based on the Bogner Ecstasy 101B head)
l ‘97 RB-01b Green (based on the Bogner Ecstasy 101B head)
l DC Modern Overdrive
l DC Modern SOD
l DC Modern 800
l DC Modern Clean
l DC Vintage Crunch
l DC Vintage OD
l DC Vintage Clean
l DC Bass
g Eleven MK II is not affiliated with, or sponsored or endorsed by, the makers of the amplifiers emulated
in the product.
Eleven MK II Amp Controls

Each Eleven MK II 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.
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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.
Bright
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 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 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.
g All Eleven MK II controls provide identical ranges as the original amps, but some numbers have been
adjusted for consistency.
Gain 2
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.
l 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.
l Tone
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.
l Bass
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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.
l 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.
l 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.
Presence
The Presence control provides a small amount of boost at frequencies above the treble control. Presence is
applied at the end of each amp model pre-amp stage, acting as a global brightness control that is
independent of other tone controls. The default setting is 3.0. The Presence range is from 0 to 10.
Master
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.
g Some might assume a Master volume knob capable of silencing the amp completely. Not so. Use the
Output knob (in the Master section) to silence the output of the plug-in. Use Master volume for tone
and distortion.
Tremolo
Tremolo is achieved through the use of amplitude modulation, multiplying the amplitude of the pre-amp
output by a waveform of lower frequency. Tremolo is not available on all amps.
l 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.
g Eleven MK II does not support Tempo Sync.

l 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.
Eleven MK II 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.
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Cabinet Type selector in the Master section
Available cabinets include the following:

l 1x8 Custom
l 1x12 Black Panel Lux
l 1x12 Tweed Lux
l 1x15 Open Back (based on an Ampeg Reverberocket 1x15 with Jensen C15Nspeaker)
l 2x12 AC Blue
l 2x12 Black Panel Duo
l 2x12 B30 (based on a Bogner 2x12 with Celestion Vintage 30 speakers)
l 2x12 Silver Cone (based on a Roland JC-120 2x12)
l 4x10 Tweed Bass
l 4x10 Black SR (based on a Fender Super Reverb 4x10 with CTS Alnico speakers)
l 4x12 Classic 30
l 4x12 65W (based on a Marshall 4x12 with original issue Celestion G12-65 speakers)
l 4x12 Green 20W (based on a Marshall 4x12 with Celestion Heritage G12M speakers)
l 4x12 Green 25W
l 8x10 Blue Line (based on an Ampeg SVT 8x10 with “towel bar”)
Cabinets are listed by their number and diameter of their speakers. For example, “1x12” means a cabinet
has a single 12-inch speaker.
g Eleven MK II is not affiliated with, or sponsored or endorsed by, the makers of the loudspeakers and
cabinets that are emulated in the product.
n Visit the Avid website (www.avid.com) to learn about each of the cabinets used to create Eleven MK II.
Pairing Amps and Cabinets
Eleven MK II 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 MK II 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 MK II’s factory Settings files (presets) for combo amps and classic combinations. Enable
the Link button to ensure that classic cabs follow classic amp settings. Settings files store and recall all
controls, (including Amp and Cabinet Type).
For combo amps and default combinations:

t Choose a factory Settings file for that amp from Eleven MK II’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.
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If you want to combine amps and cabs (unlinked):
t Click and choose from the Amp Type and Cabinet Type selectors to create new pairings.
g Use the Settings menu to save new combinations and build your own custom library (see "Eleven MK II
Eleven MK II Cabinet Controls

All cabinets in Eleven MK II and Eleven MK II Cab provide Bypass, Speaker Breakup, Mic Type, and Position
controls.
Cabinet controls
Cabinet Bypass
The Bypass switch in the Cabinet section lets you bypass cabinet and microphone processing. When in the
Bypass position, no cabinet or microphone processing is applied to the signal. When in the On position,
cabinet and microphone settings are applied.
Speaker Breakup
The Speaker Breakup slider lets you specify how much distortion is produced by the current speaker model.
Increasing the Speaker Breakup setting adds distortion that is a combination of cone breakup and other
types of speaker distortion (emulated by the speaker cabinet model). Range is from 1 to 10.
Below certain frequencies, the speaker cone vibrates as one piece. Above those frequencies (typically
between 1 kHz and 4 kHz), the cone vibrates in sections. By the time a wave travels from the apex at the
voice coil out to the edge of the speaker cone, a new wave has started at the voice coil. The result is comb
filtering and other anomalies that contribute to the texture of the overall sound.
Mic Type
The Mic Type selector lets you choose the microphone to use with the selected cabinet.
Mic Type selector in the Cabinet section
Available Mic Types include the following:

l Dynamic 7
l Dynamic 12 (only with 8x10 Bass Cab replacing Condenser 87)
l Dynamic 20 (only available with 8x10 Bass Cab replacing Dynamic 409)
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l Dynamic 57
l Dynamic 409 (not available with 8x10 Bass Cab)
l Dynamic 421
l Condenser 67
l Condenser 87 (not available with 8x10 Bass Cab)
l Condenser 414
l Ribbon 121
g Eleven MK II is not affiliated with, or sponsored or endorsed by, the makers of the microphones that
are emulated in the product.
Mic Axis
When capturing the sound of a speaker cabinet in a studio, sound engineers select different microphones
and arrange them in different placements to get a particular sound. For example, a mic can be pointed
straight at a speaker or angled slightly off-center, in order to emphasize (or de-emphasize) certain
frequencies that the mic picks up.
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 MK II, 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.
Mic Axis switch in the Cabinet section
About Mic Placement
All Eleven MK II 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 MK II recording environment.
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 MK II. This section gives you a few specific examples of some of the many
different ways you can choose to work:
l "Recording Dry: Monitor Through Eleven MK II" on the next page.
l "Recording Wet: Record Eleven MK II–Processed Track to Disk" on the next page.
l "Recording Dry and Eleven MK II Simultaneously" on page 341.
l "Processing Pre-Recorded Tracks Through Eleven MK II" on page 342
l "Blending Eleven MK II Cabinets and Amps" on page 343.
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Recording Dry: Monitor Through Eleven MK II
This workflow lets you record dry (clean) while the recorded signal is processed through Eleven MK II, letting
you hear it but without committing the track to that tone forever.
The flexibility to audition and compare different settings and combinations of amps, cabinets and
microphones is a very creative and powerful tool for mixing and arranging.
To record dry and monitor through Eleven MK II:
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 MK II on the track (see "Inserting Eleven MK II on Tracks" on page 329).
Audio track for recording dry while hearing Eleven MK II
4. Choose a Settings file (preset), or adjust Eleven MK II’s parameters to get your tone (see "Eleven MK II
5. Record enable the track, or enable TrackInput monitoring (Pro Tools Ultimate and Studio only) and
check your levels.
6. 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 MK II and any other plug-ins inserted on the track.
Recording Wet: Record Eleven MK II–Processed Track to Disk
In this workflow, the audio output of Eleven MK II is recorded to disk while tracking. Usually, no additional
dry track is recorded.
This method commits your track to the original Eleven MK II 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.
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To record guitar with Eleven MK II while playing:

a. Create one mono Auxiliary Input track.
b. Click the Add Row button (+).
c. Create one mono audio track.
d. Click Create.
3. In the Mix (or Edit) window, configure the Aux Input by doing the following:
plugged in to).
c. Click the Insert selector and select Eleven MK II.
4. Configure the audio track by doing the following:
Recording Eleven MK II (printing its output)
5. Make sure you are not overloading your input signal by checking levels in all tracks and Eleven MK II's
Input LED.
The output from Eleven MK II is recorded to disk. If you need to conserve DSP or Native processing resources,
you can remove or deactivate Eleven MK II after recording.
Recording Dry and Eleven MK II Simultaneously
You can record a dry, unprocessed track and an Eleven MK II-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:
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To record guitar dry and with Eleven MK II live:

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:
plugged in to).
c. Click the Insert selector and select Eleven MK II.
d. Record enable the audio track.
4. Configure the second audio track by doing the following:
5. Make sure you are not overloading your input signal by checking levels in all tracks and Eleven MK II's
Input LED.
The dry guitar is recorded to the first audio track, processed through Eleven MK II, then bussed to the
second audio track and recorded to disk.
Processing Pre-Recorded Tracks Through Eleven MK II
You can process pre-recorded guitar tracks, or almost any material, through Eleven MK II.
To listen to pre-recorded tracks through Eleven MK II (without re-recording):

2. Assign the audio track Output to Bus 1 (or Bus 1-2 if working with stereo material).
3. Create an Aux Input track, and configure it by doing the following:
a. Click its track Input selector and choose Bus 1 (or Bus 1-2).
b. Click the Insert selector and select Eleven MK II.
4. Begin playback and watch Eleven MK II’s Input LED to check your level. Make sure you’re not clipping
Eleven MK II’s input.
5. While listening, adjust Eleven MK II’s Input knob to increase or decrease input level.
6. After setting your gain structure, do any of the following:
t Try different Settings files (presets) to get your basic amp/cab/mic combination.
t Adjust amp controls.
t Try different cabinets and varying amounts of Speaker Breakup.
t 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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To process and re-record tracks through Eleven MK II:

2. Configure the source audio track by doing the following:
a. Assign the audio track Output a bus (such as Bus 1 if mono, or Bus 1-2 if stereo).
3. Choose Track > New and create one mono audio track.
4. Configure the new audio track by doing the following:
a. Click its track Input selector and choose the Bus 1 (or Bus 1-2).
5. Record enable the new audio track (or enable TrackInput monitoring if using Pro Tools Ultimate and
Studio).
6. Begin playback and start listening.
7. While listening, adjust Eleven MK II’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.
Blending Eleven MK II Cabinets and Amps
You can use Eleven MK II for multi-cabinet and multi-amp 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 Cabinets with Eleven MK II and Eleven MK II Cab Plug-ins
You can use multiple tracks to blend cabinet models with the Eleven MK II and Eleven MK II Cab plug-ins.
To blend cabinets with a single instance of Eleven MK II and one or more instances of Eleven MK II Cab:
1. Insert Eleven MK II on a mono audio (or Auxiliary Input) track. Route your guitar signal to this track.
2. Create a new mono Auxiliary Input (or audio) track and insert Eleven MK II Cab on that track.
3. For the new track input, select Plug-in > Eleven MK II Cab > Pre-speaker out.
Selecting Eleven MK II Auxiliary Output Send as the Input for an Auxiliary Input track
4. Repeat steps 2 and 3 to add more cabinet models for blending.

5. Adjust the settings for Eleven MK II and each instance of Eleven MK II Cab as desired.
6. Play your guitar (or play back pre-recorded audio) and adjust the volume fader on each track to
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blend cabinet models as desired.
Blending Eleven MK II and Eleven MK II Cab between and audio track and an Auxiliary Input track
If You Plan on Blending Cabinets
The Eleven MK II 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.
In each Eleven MK II 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.
g Different amps can also have a different number of stages, which can affect polarity. See "Phase
Considerations with Blending in Eleven MK II" on page 346 for more information.
How Do I Use This?
Here are a few suggested ways you can pair Eleven MK II’s amps and cabinets:
l 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 MK II plug-ins (all the cabinets as well as the active amp).
l For maximum variety, mix and match bypassed amps with active cabinets.
l For realism with the combo amps (such as the Tweed Lux and AC Hi Boost), make sure to use their
default cabinets.
Blending Eleven MK II Amps
You can easily set up tracks and Eleven MK II for multi-amp setups.
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To blend multiple amps:
1. Set up tracks and signal routing as explained in the previous workflow (see "Blending Cabinets with
Eleven MK II and Eleven MK II Cab Plug-ins" on page 343).
2. Remove (or simply bypass) the Eleven MK II plug-in on the source input/track.
g To maximize processing resources, remove the Eleven MK II 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

4. Click to open the Eleven MK II plug-in window on the soloed Aux Input, and do any of the following:
t Make sure the amp and cabinet are active (not bypassed).
t Choose a preset (Settings file).
t Pair any amp with any cabinet.
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 un-solo all the Aux Inputs.
8. Continue playing so you can hear the combined tone of all the amps.
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b. Try different pan positions for each Auxiliary Input track.
10. Evaluate the phase relationships of the combined signals and adjust accordingly (see "Phase
Considerations with Blending in Eleven MK II" below).
Phase Considerations with Blending in Eleven MK II
When multi-tracking guitar, experienced engineers know how to identify and take advantage of the phase
relationships that exist between different signals. Adjusting phase is not just a corrective technique either,
it’s also a powerful creative technique for tone, as well as for special effects.
You can use the TimeAdjuster plug-in to flip phase and to adjust timing in single-sample increments, as
described in the next sections.
Flipping Phase (Polarity)
Electric guitar is often recorded to more than one track, such as one dry or DI track, plus one or more tracks
of a mic’d amp. The different signal paths of direct tracks versus mic tracks affect the timing relationships
of the audio. Depending on the signal chain of each track, the signals can get so out of alignment that they
nearly cancel each other out.
Sending a single source track through multiple, unique amps can pose an additional challenge in that each
tube stage in an amp usually inverts the signal. So, depending on whether the number of tube stages in an
amp is odd or even, that amp will either be inverting or non-inverting, respectively. If you send an identical
signal to two amps and one is inverting while the other is non-inverting, signal cancellation will result. All
amps in Eleven MK II 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 MK II Cabinets and
Amps" on page 343. Make sure each Aux Input has an Eleven MK II plug-in followed by a TimeAdjuster
plug-in.
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 MK II/TimeAdjuster channels.
6. Try combinations of flipped and non-flipped Phase settings to find the ideal relationship for the
currently blended amps and cabinets.
Tweaking Phase
If each of the mics used on a single cabinet are not positioned carefully, comb filtering and other frequency
anomalies can occur. With real amps, the engineer moves one or more mics to find their optimal positions
relative to the source, and to each other.
To hear the effect of small adjustments to the phase relationships of signals, do the following.
To use the Time Adjuster plug-in to control phase:
1. Configure your audio track and Aux Inputs as instructed in "Blending Eleven MK II Cabinets and
Amps" on page 343. Make sure each Aux Input has an Eleven MK II plug-in followed by a TimeAdjuster
(short) plug-in.
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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 MK II Tips and Suggestions

This section leaves you with some tips and suggestions for other ways you can integrate Eleven MK II 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:
l Soft (or clean) tone for the verse, kick in the distortion for the chorus.
l Using tremolo during the intro and the bridge.
l Doubling the rhythm track halfway through the verse to build momentum.
Pro Tools automaton is the key to these and other techniques:
l 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.
l 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 MK II workflow.
n See the Pro Tools Reference Guide to learn about Snapshot automation, Glide, and other
automation features.
l For maximum flexibility, control and variety, use a dry track bussed to multiple Aux Inputs, each with
a different Eleven MK II tone (see "Blending Eleven MK II Amps" on page 344 for instructions).
Configure one for tone A, configure the next Eleven MK II (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 MK II 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 MK II Plug-In Resources
If system resources need to be conserved or minimized, you can “bus record” with effects to commit Eleven
MK II tones to disk. See "Recording Wet: Record Eleven MK II–Processed Track to Disk" on page 340.
Or, use the AudioSuite version to print Eleven MK II tracks to disk. AudioSuite is especially useful when you’re
processing loops or other shorter-form guitar material.
Beyond Eleven MK II: 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 MK II 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
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Reference Guide for mixing and submixing setups and suggestions, and try them out while exploring some
of the following effects suggestions.
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:
l 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 MK II).
l 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 MK II 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.
Eleven MK II Signal Flow Notes

The following figure shows the signal flow through Eleven MK II from its input source to its output
destination.
Signal flow through Eleven MK II
Plug-Ins are Pre-Fader

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 MK II.
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 MK II 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 MK II and use its (Trim) gain control to attenuate the signal.)
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Input Knob and Amp Gain
Eleven MK II actually gives you two separate input gain stages to the plug in:
l The Input knob in the Master section, which affects the signal level before entering the amplifier
model.
l 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, it silences amp output
to the speaker cabinet.
Eleven Effects Harmonic Plug-ins

Eleven Effects Harmonic plug-ins include the following:
l "Black Op Distortion" below
l "DC Distortion" on the next page
l "Green JRC Overdrive" on the next page
l "Tri-Knob Fuzz" on page 351
l "White Boost" on page 352
Black Op Distortion
Inspired by an 80s-era op-amp-based distortion pedal, Black Op Distortion offers massive crunch and
power. Its hard-clipping drive can squeeze aggressive rhythm and lead tones out of soft-sounding vintage
amps, and create surprisingly hard-edged tones when paired with more modern amps.
Black Op Distortion is available as a Mono (or Multi-Mono) plug-in only.
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Distortion
The Distortion knobs control the clipping level.
Cut
The Cut knob adjusts the cutoff frequency for the filter.
Volume
The Volume knob controls final output volume.
DC Distortion
This distortion effect offers a range of overdriven tones, aided by its built-in Bass and Treble EQ, which help
shape the response of the clipping circuit.
DC Distortion is available as a Mono (or Multi-Mono) plug-in only.
Distortion
The Distortion knob controls clipping level.
Treble
The Treble knob adds treble frequencies to the boosted signal.
Bass
The Bass knob adds bass frequencies to the boosted signal.
Volume
Green JRC Overdrive

Green JRC Overdrive was inspired by a low-gain 70s overdrive pedal, loved by blues and blues-rock players
the world over for its sweet, singing sting. It can be used to simply drive an amp’s input section into gentle
clipping, or to supply some dirty glow of its own.
Green JRC Overdrive is available as a Mono (or Multi-Mono) plug-in only.
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Overdrive
The Overdrive knob determines the amount of overdrive applied to the signal.
Tone
The Tone knob lets you change the tonal balance of the effect.
Level
The Level knob sets the overall output volume of the effect.
Tri-Knob Fuzz
The pedal that inspired Tri-Knob Fuzz was a transistor-based unit, originally popular with lead guitarists
searching for ever-higher gain in the ’70s. It shone again in the ’90s grunge rock scene, probably pushed
further into woolly grind than its makers could have anticipated.
Tri-Knob Fuzz is available as a Mono (or Multi-Mono) plug-in only.
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Volume
The Volume knob sets the overall output volume of the effect.
Sustain
The Sustain knob sets the gain of the fuzz circuit.
Tone
The Tone knob sets the tonal balance of the effect, from deep and full of sub-bass to high and shrill.
White Boost
This booster effect, based on a well-loved clean booster pedal with 20 dB of gain boost and a built-in EQ, is
useful for driving the preamp section of any amp model into a gentle (or not so gentle) overdrive. The Bass
and Treble controls boost and cut frequencies as normal, but in this usage, they help shape the overdrive
response as well.
White Boost is available as a Mono (or Multi-Mono) plug-in only.
Gain
The Gain knob controls the amount of boost added to the signal.
Volume
Treble
The Treble knob boosts (turn to the right) or attenuates (turn to the left) treble frequencies to the boosted
signal.
Bass
The Bass knob boosts (turn to the right) or attenuates (turn to the left) bass frequencies to the boosted
signal.
Lo-Fi
Lo-Fi provides “retro,” down-processing effects.
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Lo-Fi features include:
l Bit-rate reduction
l Sample rate reduction
l Soft clipping distortion and saturation
l Anti-aliasing filter
l Variable amplitude noise generator
Lo-Fi is available in DSP, Native, and AudioSuite formats.
Lo-Fi supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Lo-Fi operates as a mono, multi-mono, or stereo plug-in.
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 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).
g 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 sample data bits in an effort to fit the audio into the selected bit
resolution. This imparts a characteristically raunchy sound to the audio that 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 pseudo-white 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
it reached Lo-Fi. Monitor your send or insert signal levels closely to prevent this from happening.
Pro Subharmonic
Avid Pro Subharmonic is an AAX plug-in (DSP, Native, and AudioSuite) that synthesizes low-frequencies
based on the harmonic content of the source audio signal.
Pro Subharmonic supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
The multichannel version of Pro Subharmonic supports stereo audio, and can also add a channel of LFE (.1)
content to any 5.0, 6.0, or 7.0 track. Subharmonic material may also be added to the existing LFE channel
of a 5.1, 6.1, or 7.1 track.
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The mono/multi-mono version should be used for all other channel formats.
c Greater-than-stereo formats are only available with Pro Tools Ultimate and Studio software only.
c 7.0 SDDS to 7.1 SDDS and 7.1 SDDS to 7.1 SDDS instances of Pro Subharmonic are Native-only. All
other channel formats are supported on both Native and DSP.
In addition to standard knob, button, and fader controls, Pro Subharmonic also provides a graphic display
of the synthesized frequencies and of the dynamic curves of the low pass and high pass filters applied to
the low frequencies.
Metering
Pro Subharmonic provides channel-maximum sample peak meters for Input and Output signals.
The Peak Hold indicator appears as a thin line in the meter. This provides highly accurate visual metering
correlation with the audio signal.

l Dark Blue
l Light Blue
l Yellow
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Pro Subharmonic Input meter
Input
Input Level control
Input Level

t Click in the Input Level field and type a value (–18.0 dB to +18.0 dB).
Input Meters
The Input meters show channel-maximum sample peak meters summed from all input channels before
processing (from –90 dB to +6.0 dB).
Output
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Output Level control
Output Level
The Output Level control sets the output level after processing, letting you boost or attenuate gain of the
output signal.

t Click in the Output Level field and type a value (–18.0 dB to +18.0 dB).
t Click the Output Volume control and drag up or down to adjust the Output Volume setting.
Output Meters
The Output meters show channel-maximum sample peak meters summed for all output channels after
processing (from –90 dB to +6.0 dB).
Dynamic Frequency Display

Pro Subharmonic provides a graphic display for the synthesized frequency content, low frequency content
already present in the source audio, and for the low and high pass filters.
Graphic Frequency and Dynamics display
High Pass Filter Curve
Displays the frequency range and dynamic curve for the High Pass filter.
Low Pass Filter Curve
Displays the frequency range and dynamic curve for the Low Pass filter.
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Lower Synthesized Band
Displays the dynamics of the lower range of synthesized frequencies.
Upper Synthesized Band
Displays the dynamics of the upper range of synthesized frequencies.
Direct Bass Band
Displays the dynamics and the frequency range of the direct bass band (the bass frequency content from
the source audio signal).
High Pass and Low Pass Filter Controls

The High Pass and Low Pass filters shape the output of Pro Subharmonic. These filters apply to the direct,
unprocessed low frequency content of the source signal as well as the synthesized frequencies.
High Pass and Low Pass Filter controls
HP Freq
Sets the high frequency for the High Pass filter (5–1000 Hz).
HP Q
Sets the Q for the High Pass filter (0.10–5.00).
LP Freq
Sets the low frequency for the Low Pass filter (5–1000 Hz).
LP Q
Sets the Q for the Low Pass filter (0.10–5.00).
To adjust the Frequency of the High Pass or Low Pass filter, do one of the following:
t Move the cursor over the frequency curve in the Frequency Graph so that it shows the “FREQ” cursor,
then drag to adjust the Frequency value.
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To adjust the Q of the High Pass or Low Pass filter, do one of the following:
t Move the cursor over the frequency curve in the Frequency Graph so that it shows the “Q” cursor,
then drag to adjust the Q value.
Adjusting the Q setting on the Frequency Graph
Subharmonic Frequency Range

Use the Subharmonic Frequency Range buttons to specify the frequency range of the source signal for the
synthesized audio content. For example, in a post production setting, you might use the lowest option to
reinforce an explosion in the LFE channel of a 5.0 source signal. Or, you might use the 60–90 Hz option for a
stereo music mix to give it a little more “bump.” The lower settings can add power to kick drums and
explosions, while the higher ones add fatness to bass lines and low mid-range audio source signals. The
best setting will depend on the audio content.
g You can also use MIDI to tune Pro Subharmonic. This can be especially useful for music production.
See "Tuning Subharmonics with MIDI" on page 363.
To set the frequency range for synthesized subharmonics, click one of the following options:
t 120–180 Hz
t 80–120 Hz
t 60–90 Hz
t 40–60 Hz
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Frequency Range Selector buttons
g Shift-click the Input (Hz) button to automatically adjust the High Pass and Low Pass Frequency
settings to align with the Input (Hz) setting.
Lower, Upper, and Direct Gain Controls

Pro Subharmonic provides gain controls for the lower band, upper band, and direct bass band. You can also
solo each.
Band Gain controls
The Lower Band Gain control corresponds to the lower frequency of the currently selected Subharmonic
Frequency Range setting. Likewise, the Upper Band Gain control corresponds to the upper frequency of the
currently selected Subharmonic Frequency Range setting. The Direct Gain control lets you blend in some of
the low frequency content of the source signal.
To adjust the gain for the Lower Band, Upper Band, or Direct Bass Band, do one of the following:
t Drag the Gain knob up or down, or to the left or right.
t Click the Gain field and type a value between –120 dB and 24 dB.
t Move the cursor over the corresponding Gain curve in the Frequency Graph so that it shows the
“Gain” cursor, and drag to adjust the Gain value.
Adjusting the Gain setting on the Frequency Graph
To solo the Lower Band, Upper Band, or Direct Bass Band:

t Click the corresponding Solo button.
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Drive
Use the Drive control to adjust the level of all three bands. The Drive control applies soft-saturation
distortion, so it can be used to prevent internal clipping. The Drive effect is shaped by both the Low Pass
and High Pass filters.
You can also select any one of six Drive Character options. These settings change the behavior and sound
of Pro Subharmonic by altering the resonances of the internal filters and by adjusting the contour of the
saturation stages in the processed signal path. For the cleanest, most direct sound, use Clean I. For a
sound that adds some color, select one of the Resonant or Distort options.
Experiment with the various Drive Character options and the Drive level setting until you find the tone-
quality you want.
Drive control
To adjust the amount of Drive, do one of the following:

t Drag the Drive knob up or down, or to the left or right.
t Click the Gain field and type a value between –18.0 dB and 18.0 dB.
To set the Drive Character, select one of the following options:
t Clean I
t Clean II
t Resonant I
t Resonant II
t Distort I
t Distort II
Mix Controls
Mono and Stereo tracks provide a single Mix control to balance the wet (added bass) signal and the dry
signal. If any of the bands are soloed, the Mix solo button (the button to the right of the numeric field)
flashes. Click the flashing button to unsolo all soloed bands. You can click the unlit Mix solo button to
effectively mute the dry signal regardless of the Mix setting. This lets you audition only the synthesized and
processed signal.
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Mix control (Mono and Stereo tracks)
To adjust the dry/wet mix, do one of the following:

t Drag the Mix knob up or down, or to the left or right.
t Click the Gain field and type a value between 0% (all dry) and 100% (all wet).
g When Mix is at 50%, both the wet and dry signal paths are at full volume. Setting the Mix control to
values below 50% keeps the dry level at full, but decreases the amount of the wet signal. Conversely,
settings the Mix control to values over 50% keeps the wet level at full, but decreases the level of the
dry signal.
For surround tracks, Pro Subharmonic provides multiple Mix controls.
Mix Output
Lets you mix the output of Pro Subharmonic between the LFE channel and the Left and Right channels.
Wet
Lets you mix the wet signal from 0–100% to the plug-in output.
Dry
Lets you mix the dry signal from 0–100% to the plug-in output.
Wet and Dry Solo Buttons
Let you solo the wet or dry output signals respectively. Only one of these may be soloed at a time (for
example, soloing the Dry signal while the Wet signal is already soloed cancels the solo on the Wet signal).
Mix control (surround tracks)
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Surround Send Controls
When Pro Subharmonic is inserted on a surround track, it provides controls to mix how much of the source
audio is sent from the Front channels, the Center channel, the Surround channels, and the LFE channel (if
present) for Subharmonic synthesis processing. For example, if your source audio has a big sound effect in
the center channel and music in the front left and right channels, and you want subharmonics synthesized
only from the sound effect, turn up the Center channel input and turn down the Front and Surround channel
inputs. On .1 tracks only, Pro Subharmonic provides an LFE control so that you can create subharmonics
using the audio on the LFE channel. Each input source can be soloed.
Front Send
Lets you feed Pro Subharmonic from the front left and right input channels.
Center Send
Lets you feed Pro Subharmonic from the center input channel.
Surround Send
Lets you feed Pro Subharmonic from the surround input channels.
LFE Send (.1 Format Tracks Only)
Lets you feed Pro Subharmonic from the LFE input channel.
Tuning Subharmonics with MIDI

In addition to using the Range Setting buttons, you can use MIDI to tune Pro Subharmonic. Pro Subharmonic
can receive MIDI note numbers 12 through 60. This means that you can use MIDI notes to set the Band
frequencies from 16 Hz up to 262 Hz. Use this to reinforce the low end of your program material based what
key or mode it is in.
When Pro Subharmonic is receiving MIDI, the MIDI In icon highlights, the Range settings are controlled by
the MIDI input, and the Range Setting buttons are bypassed. The Lower band is tuned to the incoming MIDI
note and the Upper band is tuned to a perfect fifth above that.
c VENUE and Media Composer do not support MIDI.

To route MIDI to Pro Subharmonic:
1. Insert Pro Subharmonic on a track.

2. Create a new MIDI track.
3. Route the output of the MIDI track to the corresponding Pro Subharmonic MIDI node.
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Routing MIDI to Pro Subharmonic
To tune Pro Subharmonic with MIDI:
1. Use a MIDI track routed to the specific instance of Pro Subharmonic that you want to control.
2. Record, enter, or import a MIDI sequence on the MIDI track.
c Be sure to only use a monophonic MIDI sequence or live performance to control Pro
Subharmonic.
3. Play back the session. The MIDI In icon highlights when Pro Subharmonic is receiving MIDI.
Routing MIDI to Pro Subharmonic
g The MIDI tuning feature can be used to lock Pro Subharmonic to a bass Virtual Instrument. Routing a
MIDI track to both the input node of a bass Virtual Instrument and the input node of a Pro
Subharmonic that is inserted after the Virtual Instrument creates subharmonics evenly across a four-
octave playing range (from MIDI note number 12 up to MIDI note number 60).
g When using Pro Subharmonic with a bass Virtual Instrument, you may want to transpose the bass part
up or down to better tune Pro Subharmonic. To do this, copy and paste the MIDI sequence to a new
MIDI track and route the track output to Pro Subharmonic. Use Real-Time MIDI properties on the MIDI
track to transpose the sequence as desired.
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MIDI Input to Pro Subharmonic AudioSuite
The AudioSuite version of Pro Subharmonic also supports MIDI input. This can be used to fine-tune the input
frequency range for the AudioSuite version of Pro Subharmonic.
To tune Pro Subharmonic AudioSuite with MIDI:
1. Select the audio you want to process in the Edit window.

2. Open the AudioSuite version of Pro Subharmonic.
3. Create a new MIDI track and rout its output to Pro Subharmonic.
4. Send a MIDI note to Pro Subharmonic via the MIDI track.
5. Process the selected audio with Pro Subharmonic AudioSuite.
Recti-Fi
Recti-Fi provides additive harmonic processing effects through waveform rectification. Recti-Fi features the
following effects:
l Subharmonic synthesizer
l Full wave rectifier
l Pre-filter for adjusting effect frequency
l Post-filter for smoothing generated waveforms
Recti-Fi is available in DSP, Native, and AudioSuite formats.
Recti-Fi supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Recti-Fi operates as a mono, multi-mono, or stereo plug-in.
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 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.
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The range of the Pre-Filter is from 43 Hz to 22 kHz, with a maximum value of Thru (which effectively means
bypass).
Normal waveform
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
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.
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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.
Alt-Max rectification
Recti-Fi Gain Control

Gain lets you adjust signal level before the audio reaches the Post-Filter. This is particularly useful for
restoring unity gain if you have used the Pre-Filter to cut off high frequencies prior to rectification. The
range of this control is from –18dB to +18dB.
Recti-Fi Post-Filter
Waveform rectification, particularly alternating rectification, typically produces a great number of
harmonics. The Post Filter control lets you remove harmonics above the cutoff frequency and smooth out
the sound. This is useful for filtering audio that contains subharmonics. To create classic subharmonic
synthesis effects, set the Pre-Filter and Post-Filter to a relatively low frequency.
The range of the Post-Filter control is 43 Hz to 22 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

it reached Recti-Fi. Monitor your send or insert signal levels closely to prevent this from happening.
Reel Tape Saturation

Reel Tape Saturation is part of the Reel Tape suite of tape-simulation effects plug-ins. 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 is available in DSP, Native, and AudioSuite formats.
Reel Tape Saturation supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Reel Tape Saturation operates as a mono, multi-mono, or stereo plug-in.
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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.
Reel Tape Saturation can be placed on mono, stereo, or multichannel tracks.

Drive
Output
Tape Machine
US
Swiss
Lo-Fi
effect.
Tape Formula
Classic
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Hi Output
Reel Tape Saturation Controls

In addition to the Drive, Output, Tape Machine, and Tape Formula controls, Reel Tape Saturation has the
following controls:
Speed
The Speed control adjusts the tape speed in ips (inches per second). Tape speed affects the frequency
response of the modeled tape machine. Available tape speeds include 7.5 ips, 15 ips, and 30 ips, with a
default setting of 15 ips.
Noise
Reel Tape Saturation produces noise only during playback and recording, and not when the transport is
stopped.
The Noise control adjusts the level of simulated tape noise that is added to the processed signal. The
characteristics of the noise depend on the Speed, Bias, and Tape Machine settings, and the relative level of
the noise depends on the Drive, Cal Adjust, and Tape Formula settings.
Noise is adjustable from Off (–INF) to –24 dB, with the default value being Off.
Bias
The Bias control simulates the effect of 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.
Cal Adjust
Cal Adjust simulates the effect of three common calibration levels on the modeled tape machine and
magnetic tape formulations.
With the evolution of tape formulations, it was possible to increase the fluxivity level, or magnetic strength,
of the signals on tape. Over the years, this resulted in an elevation of recorded levels relative to a standard
reference fluxivity (185 nW/m at 700 Hz). The Cal Adjust value expresses the elevated level in dB over this
standard reference level.
The Cal Adjust control does not affect the overall gain, but does affect the amount of saturation effect for a
given input signal.
Available Cal Adjust values are:

l +3 dB (equivalent to 250 nW/m)
The default value is +6 dB.
Reel Tape Saturation Tips

l Use Reel Tape Saturation on individual tracks to round out sharp transients or add color to sustained
tones.
l Use Reel Tape Saturation on a group of tracks (for example drums) to add cohesiveness to the sound
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of the group.
l 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.
SansAmp PSA-1
SansAmp PSA-1 is a guitar amp simulator plug-in. 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! Tube sound, speaker simulation, warm equalization and cool lo-fi textures—no wonder thousands
of records feature the classic sounds of SansAmp!
SansAmp PSA-1 is available in DSP, Native, and AudioSuite formats.
SansAmp PSA-1 supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
SansAmp PSA-1 operates as a mono, multi-mono, or stereo plug-in.
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 emulation captures the low-order harmonics and sweet overdrive unique to
tube amplifiers. And pushed harder, SansAmp also generates cool lo-fi and grainy sound textures that still
retain warmth.
SansAmp also features a proprietary speaker simulator which emulates the smooth, even response of a
multiple-miked speaker cabinet—free of the harsh peaks, valleys and notches associated with single miking
or poor microphone placement.
Finally, SansAmp provides two extremely sweet sounding tone controls (high and low) that sound great on
most anything.
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PSA-1 Controls
Use the eight knobs to dial in your 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.
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 by ±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
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).
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9 Dither Plug-Ins
Dither plugs-in minimize quantization artifacts when reducing the bit depth of an audio signal to 16-, 18-, or
20-bit resolution. 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.
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 converters of the interface.

l "Dither" below
l "POW-r Dither" on the next page
Dither
Dither is a dither-generation plug-in. The Dither plug-in minimizes quantization artifacts when reducing the
bit depth of an audio signal to 16-, 18-, or 20-bit resolution.
Dither is available in DSP and Native formats.
Dither supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Dither operates as a mono, multi-mono, or stereo 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.
n For more information on using dither plug-ins in Pro Tools, see the Pro Tools Reference Guide.
The Dither plug-in has user-selectable bit resolution and a noise shaping on/off option.
g 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 converters of the
interface.
Dither Controls
The Dither plug-in has a Bit Resolution button and a Noise Shaping button.
Bit Resolution Button

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.
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18-bit
Recommended for output to digital devices with a maximum resolution of 18 bits.
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. 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 in Pro Tools.
w 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.
POW-r Dither
POW-r Dither is a dither-generation plug-in. 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 required when reducing bit depth.
POW-r Dither is available in DSP and Native formats.
POW-r Dither supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
POW-r Dither operates as a mono, multi-mono, or stereo plug-in.
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 20-bit 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.
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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.
c The POW-r Dither plug-in is not appropriate for truncation stages that are likely to be further
processed. It is recommended that POW-r Dither be used only as the last insert in the signal chain
(especially when using Type 1 Noise Shaping).
The POW-r Dither plug-in provides three types of noise shaping, each with its own characteristics. Try each
noise shaping type and choose the one that adds the least amount of coloration to the audio being
processed.
Type 1
Has the flattest frequency spectrum in the audible range of frequencies, modulating and accumulating the
dither noise just below the Nyquist frequency. Recommended for less stereophonically complex material
such as solo instrument recordings.
Type 2
Has a psychoacoustically optimized low order noise shaping curve. Recommended for material of greater
stereophonic complexity.
Type 3
Has a psychoacoustically optimized high order noise shaping curve. Recommended for full-spectrum, wide-
stereo field material.
n For more information on using dither plug-ins in Pro Tools, see the Pro Tools Reference Guide.
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10 Sound Field Plug-Ins
Sound Field plug-ins let you manipulate where sounds appear in the stereo or multi-channel sound field, or
up/down mix material for the current listening environment (such as "down mixing" a 5.1 surround mix to
stereo to be able to listen in stereo).

l "AutoPan" below
l "Down Mixer" on page 383
AutoPan
AutoPan is an automatic panning plug-in that is available in DSP and Native formats. 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. 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.
AutoPan Controls
AutoPan provides output meters, panner controls, LFO controls, tempo controls, and envelope controls.
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)
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The Clip indicator lights red when the channel has clipped. The clip indicator for each channel can be
cleared by clicking it.
AutoPan Panner Controls

The Panner section provides different controls for different output channel configurations. AutoPan in
mono-to-stereo and mono-to-LCR formats provide controls common to all output configurations: Output,
Width, and Manual. 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.
Output
The Output slider lets you cut or boost the output signal level from –24 dB to +12 dB.
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 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
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Panner section, mono-to-5.0, left to right path selected
The Angle slider is only available with mono-to-quad 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-to-quad and mono-to-5.0 formats, and a left to right or right
to left path selected.
Spread
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).
Panner section, mono-to-5.0, clockwise path selected
The Spread slider is only available with mono-to-quad and mono-to-5.0 formats, and a circular path
(clockwise or counterclockwise) selected.
Panning Source
Click LFO or Env to select the source for panning. When the Source is set to LFO, panning is controlled by
the LFO and its controls (see "AutoPan LFO Controls" on the next page). When the Source is set to Envelope
(Env), panning is controlled by the Envelope Detector and its controls (see "AutoPan Envelope Controls" on
page 381). 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 383).
Panning Source buttons
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Panning Display
The Panning display graphically represents the panning field and the location of the sound source within
that field.
Panning display, mono-to-5.0, left to right path selected
Sound Location Indicator
This bright yellow light indicates the location of the sound source.
Panning Field Indicator
This is the grey line on which the yellow Sound Location indicator travels and indicates the panning field.
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
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 the previous page).
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LFO section
g When the Panner section is set to Envelope (Env), the controls in the LFO section have no effect on
panning.
Rate
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 display always shows the current session tempo (see "Tempo Display" 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 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.
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LFO Triggers
l Single
When the Single trigger is selected, the LFO will cycle through the waveform once only and then
stop.
l 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.
l 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.
l 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.
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 display always shows the current session tempo.
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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.
Selecting Duration
Tempo Display
The Tempo display shows the tempo in BPM. The value in the Tempo display can also be edited directly by
clicking it and typing a new value.
Tempo display
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.
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Envelope section
g When Envelope (Env) is not selected as the Panning Source, the controls in this section have no effect
on the sound.
Side-Chain Input
When the Side-Chain Input selector (the key icon) is enabled, the audio for the Envelope Detector is taken
from the side-chain input rather than the current track. Select the Side-Chain Input using the Pro Tools Key
Input selector at the top of the plug-in window.
Side-Chain Input selector enabled
Threshold
The Threshold slider sets the amplitude level required for the Envelope Detector. The LFO Envelope
Detector light blinks brighter when audio is detected above the threshold.
Attack
The Attack slider sets the attack rate of the Envelope Detector.
Release
The Release slider sets the release rate of the Envelope Detector.
Using AutoPan
AutoPan can be used for dynamic panning effects based on a Low Frequency Oscillator (LFO), an
amplitude envelope (ENV), or manual control.
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 AutoPan: panning to the beat for rhythmic
panning effects and surround panning effects for post production.
Panning to the Beat

AutoPan lets you synchronize the LFO to MIDI Beat Clock for rhythmic panning effects.
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To synchronize 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 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 AutoPan Plug-In window, enable Link To Tempo. This sets the LFO rate to follow the session
tempo.
4. Select a duration from the Duration selector. For example, select 2 Beats.
5. Select a 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.
Post Production Panning
(Pro Tools Ultimate and Studio Only)
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.
To pan a mono track to 5.0 with AutoPan:
1. Insert a mono-to-5.0 instance of AutoPan on the mono track containing the audio you want to pan.
The track’s channel width changes from mono-to-5.0.
2. Select a 5.0 output path from the track’s Output selector.
3. In the AutoPan Plug-In window, select a clockwise or counter-clockwise Path.
4. Adjust the Spread and Width sliders.
g Try automating Spread and Width to alter the positioning of the panned sound.
5. From the LFO Waveform selector, select Half Sine.
g Try automating the Manual control instead of using the LFO to create a more erratic panning of
the “mosquito” sound.
6. Adjust the Rate slider.
g Try automating Rate to alter the speed of the panned sound over time.
7. Play back the session to hear the “mosquito” flying around your head.
Using the Side-Chain Input
The Side-Chain Input option in 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 a channel to a bus and setting the side-
chain input on AutoPan to the same bus.
n For more information on using the Side-Chain Input, see the Pro Tools Reference Guide.
Down Mixer
Avid Down Mixer can be used to automatically mix greater-than-stereo multichannel tracks (such as 5.1)
down to stereo (Pro Tools Studio and Ultimate only) or stereo tracks down to mono.
Down Mixer is available in DSP and Native formats.
Down Mixer supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
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Down Mixer (5.1 to Stereo)
Down Mixer supports the following greater-than-stereo multichannel formats:

l LCR
l LCRS
l 5.0
l 5.1
l 7.1 SDDS
l 7.1
l 7.0.2
l 7.1.2
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 (Stereo to Mono)
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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.
Mute
When enabled, the Mute button mutes the channel input to the Down Mixer.
Phase
When enabled, the Phase button inverts the phase of the channel input to the Down Mixer.
Level
You can adjust the level of the channel input to the Down Mixer from –45 dB to +12 dB. For stereo to mono
down mixing, both the Left and Right channels are mixed to summed mono. For greater-than-stereo
multichannel down mixing, the following rules apply:
l All left-channel sources (L, Lc, Ls, Lss, Lsr) feed to the left channel (L) of the down mixer.
l All right-channel sources (R, Rc, Rs, Rss, Rsr) feed to the right channel (R) of the down mixer.
l 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.
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.
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11 Instrument Plug-Ins
Instrument plug-ins are Virtual Instruments (such as synths and samplers, or even a click) that can be
played with MIDI from Instrument and MIDI tracks.

l "AutoPan" on page 375
l "Down Mixer" on page 383
Click II
Click II is a metronome plug-in. The Click II plug-in creates an audible click during session playback that
you can use as a tempo reference when performing and recording. The Click II plug-in receives its tempo
and meter data from the Pro Tools application, letting it follow any changes in tempo and meter in a
session. The Click II plug-in is a Native mono-only plug-in. Several click sound presets are included.
Click II supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
g Click II does not sound if the Click option is disabled (Options > Click) or if the Click track has been
muted.
Click II Controls and Displays

Click II provides displays for the session Meter and Tempo settings (which can be set manually in the
Transport window or can follow the conductor rulers for Meter and Tempo in the session Timeline).
Beat Display
The Beat display shows the number of beats in a bar as determined by the Meter for the session. If the
session contains meter changes, the Beat display shows the number of beats in a bar for the Meter at the
current location of the Playback Cursor.
Follow Meter
When the Follow Meter option is enabled (highlighted), Click II follows the Meter track for the session. If this
option is disabled (un-highlighted), you can set the rhythmic values for Click 1 and Click 2 independently of
the Meter track.
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To enable (or disable) the Follow Meter option:
t Click the Follow Meter toggle below the Beat display.
BPM Display
The BPM display shows the current tempo in the session. If Tempo is set manually, with the Conductor track
disabled, the tempo as set in the Transport window is displayed. If the Conductor track is enabled, the
Tempo at the current location of the Playback Cursor is displayed.
ON Button and MIDI IN LED

The ON button lets you enable (highlighted) or disable (grayed out) the LED that blinks with the click. When
enabled, the LED blinks on and off providing a visual indication of the beat. When disabled, the LED is
grayed out and provides no visual indication of the beat.
Click 1
The Click 1 section provides controls for the downbeat click.
Accent Fader
The Accent fader lets you set the relative strength of the accent (output MIDI velocity) for the downbeat
click.
Click Sound Selector
You can choose from several click sound options using the Click Sound selector.
Click Beat Value
If the Follow Meter option is disabled, you can manually set the rhythmic value for the downbeat click. If the
Follow Meter option is enabled, the Click Beat Value options grayed out.
To set the Click Beat Value independently of the Meter track:
1. Ensure that the Follow Meter option is disabled.

2. Click to select a rhythmic value for the downbeat click (whole, half, quarter, eighth, or sixteenth
note).
3. Click to select (or deselect) the triplet or dot modifier for the beat value.
g The Click Beat Value options can be automated to support the appropriate accent patterns for
different meters for the click.
Click 2
The Click 2 section provides the same controls as the Click 1 section, but for all beats other than the
downbeat.
ReWire Requirements
To use the ReWire plug-in, you will need:
l An Avid-qualified Pro Tools system
l 64-bit ReWire-compatible client software (such as Reason from Propellerheads Software)
c 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.
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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 Ultimate
With Pro Tools Ultimate, the ReWire 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 track channels and ReWire audio streams
that does not exceed the maximum number of possible voices for your system. For example, if you are
playing 96 stereo audio tracks in a 48 kHz/24-bit session on a system that supports 256 voices at that
sample rate, another 64 channels of audio will be available for use with ReWire. However, note that ReWire
only supports a maximum of 64 audio streams per host application.
Using ReWire at higher sample rates will increase the load on the CPU. For example, CPU load at 96 kHz is
double the load at 48 kHz. You can monitor Pro Tools CPU usage in the System Usage window, making sure
to not overtax your system.
g With Pro Tools Ultimate, the standard Hardware Buffer size of 512 samples is recommended for using
ReWire in sessions with sample rates above 48 kHz.
Track Count with Pro Tools and ReWire Clients
When using ReWire-client applications with Pro Tools, 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).
ReWire
Pro Tools supports ReWire version 2.0 technology developed by Propellerheads Software. ReWire is
available in Pro Tools using the ReWire Native plug-in.
c Pro Tools does not support ReWire on Apple Silicon computers.

ReWire provides real-time audio and MIDI streaming between applications, with sample-accurate
synchronization and common transport functionality.
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
Plug-In Insert menu (Instrument category) 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.
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Once the outputs of your software synthesizers and samplers are routed to Pro Tools, you can:
l Process incoming audio signals with plug-ins
l Automate volume, pan, and plug-in controls
l Bounce To Disk
l Take advantage of the audio outputs of your Pro Tools audio interfaces
c Pro Tools does not support sending audio to ReWire client applications.
c 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.
c Exchange of additional metadata such as controller and note names between Pro Tools and
ReWire clients is not supported.
Audio and MIDI signal flow between Pro Tools and a ReWire client application (Reason shown)
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12 MIDI Plug-ins
MIDI plug-ins affect MIDI in real-time on playback or live MIDI input on Instrument tracks. Use MIDI effect
plug-ins to add MIDI effects such as note stacking or velocity processing to enhance parts and
arrangements. MIDI effects plug-ins can make a sequence sound more like a human performance or help
you discover fresh musical and rhythmic creative ideas.
To insert a MIDI plug-in on an Instrument track:
1. In either the Mix or Edit window, ensure that Inserts view is enabled.
2. Click the first available insert and select the MIDI plug-in you want from the MIDI plug-in submenu.
(Note that MIDI plug-ins can be used with any insert.)
3. Insert a virtual instrument plug-in after any MIDI plug-ins inserted on the track (or route the MIDI
output to an external MIDI device).
l "Avid Note Stack" on the next page
l "Avid Pitch Control" on page 392
l "Avid Velocity Control" on page 393
l "Audiomodern Riffer" on page 395
l "Mixed in Key MIDI Plug-ins" on page 411
l "Modalics EONarp" on page 412
l "Pitch Innovations Groove Shaper" on page 416
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Avid Note Stack
Avid Note Stack lets you add MIDI notes to incoming MIDI notes to create note stacks. You can add or
subtract notes from the note stack (1–9) by clicking Note# buttons. You can also adjust the interval between
the added note and the incoming note, and scale its velocity. To add an element of random variation, you
can adjust the probability of whether each note in the stack is added or not. This can be useful for layering
drum samples, such as snare and hand claps, without having to record additional MIDI data, with or
without variation.
Notes 1–9
l Note On/Off (1–9): Turns the generated note On or Off.
l Note Offset: Defines the number of semitones by which the incoming note is transposed.
l Note Velocity (V): Defines the velocity of the generated note relative to the velocity of the incoming
note. At 50%, the note plays at half the velocity and 200% the velocity is doubled (maximum 127).
l Note Probability (P): Defines the probability of a note being triggered each time there is an incoming
note. At 100%, the note always plays, while at 50% there is 50% chance that the note will be played.
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Avid Pitch Control
Avid Pitch Control lets you alter the pitch of incoming MIDI notes. Pitch Control can be useful for triggering
specific notes without having to re-assign MIDI notes in an electronic drum kit or other MIDI controller. For
example, kick and snare are most commonly C1 and D1, but side-stick, claps, and rim shot/accent
assignments vary among drum libraries. Pitch Control can be used to trigger the sounds you want without
having to edit MIDI data or re-assign notes in the controller or virtual instrument. Note Stack can also be
used for percussive patterns to generate different sounds based on probability.
Range
Pitch
When enabled, only notes defined by Pitch Range are processed. When disabled, all incoming notes are
processed.
l Pitch Range Type: Determines whether processing is applied only to notes within the defined range or
to notes outside the defined range.
l Pitch Range Low: Defines the lower limit of the Pitch range. If the lower limit value is smaller than the
upper limit, then the range is inverted.
l Pitch Range High: Defines the upper limit of the Pitch range. If the upper limit value is bigger than the
lower limit, then the range is inverted.
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Velocity
When enabled, only the notes defined by Velocity Range are processed. When disabled, all incoming notes
are processed.
l Velocity Range Type: Determines whether processing is applied only to notes within the defined
range or to notes outside the defined range.
l Velocity Range Low: Defines the lower limit of the Velocity range. If the lower limit value is smaller
than the upper limit, then the range is inverted.
l Velocity Range High: Defines the upper limit of the Velocity range. If the upper limit value is bigger
than the lower limit, then the range is inverted.
Control
l Transpose by Octaves/Semitones: Transposes selected notes by octaves and semitones.
l Transpose From/To: Transposes selected notes by semitones, as expressed by the difference between
the source and destination pitches.
l Set All Notes : Sets all selected notes to the same pitch.
l Quantize to Session Key in Scale Steps: Transposes selected notes up or down by scale steps based
on the Pro Tools Key Signature ruler. When set to 0, only notes outside of scale are transposed.
Avid Velocity Control

Avid Velocity Control lets you alter the velocities of incoming MIDI notes.
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Range
Velocity
When enabled, only the notes defined by Velocity Range are processed. When disabled, all incoming notes
are processed.
l Velocity Range Type: Determines whether processing is applied only to notes within the defined
range or to notes outside the defined range.
l Velocity Range Low: Defines the lower limit of the Velocity range. If the lower limit value is smaller
than the upper limit, then the range is inverted.
l Velocity Range High: Defines the upper limit of the Velocity range. If the upper limit value is bigger
than the lower limit, then the range is inverted.
Pitch
When enabled, only the notes defined by Pitch Range are processed. When disabled, all incoming notes are
processed.
l Pitch Range Type: Determines whether processing is applied only to notes within the defined range or
to notes outside the defined range.
l Pitch Range Low: Defines the lower limit of the Pitch range. If the lower limit value is smaller than the
upper limit, then the range is inverted.
l Pitch Range High: Defines the upper limit of the Pitch range. If the upper limit value is bigger than the
lower limit, then the range is inverted.
Control
Select the desired Control mode for the specified Velocity and Pitch Range.
l Filter Out: filters out (ignores) the selected note range from processing.
l Set to: Sets all velocities to the specified value (1–127).
l Add: Adds to existing velocity values by the specified amount (1–127).
l Subtract: Subtracts from existing velocity values by the specified amount (1–127).
l Scale by: Scales all velocities by a percentage amount (1–400%).
Randomize On/Off
When enabled, the selected Control mode is randomized by the specified percentage value.
l Randomize Percentage: Defines the percentage value for the randomization of the selected Control
mode.
Limit On/Off
When enabled, the result of the selected Control mode is restricted to a minimum and maximum range.
l Limit Low: Defines the lower limit of the Limit range.
l Limit High: Defines the upper limit of the Limit range.
Apply to
Determines whether the Control mode is applied only on Note On, Note Off, or both Note On & Off events.
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Audiomodern Riffer
Riffer by Audiomodern is a smart MIDI tool that generates unique sequences and melodies. Use it to drive
any virtual instrument or hardware MIDI device directly from within Pro Tools. You can select various scales,
levels of complexity, step lengths, starting and ending points, pitch transpositions, number of measures,
and more.
Video Resources
For a guided tour video of Riffer and other MIDI plug-ins in Pro Tools, watch MIDI Plugins.
For an overview of Riffer that also highlights the new features in the latest version, check out this video from
Audiomodern: Riffer by Audiomodern Complete Overview.
QuickStart
Use this QuickStart to learn your way around Riffer and start creating.
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Tracks Setup
To set up tracks to start using Riffer:
1. Insert Riffer in the first slot on an Inst track.

2. Insert the desired virtual instrument plug-in on the next slot of that same track.
g To learn about additional MIDI routing options in Pro Tools, see the Pro Tools Reference Guide (in Pro
Tools choose Help > Pro Tools Reference Guide).
Set Up Riffer
The following steps show one way to get started making music with Riffer. In reality you can do some or all
of these steps, in any order you like.
To configure Riffer and start making music:
1. Click to open the Riffer window.

2. Tell Riffer what Key your song is in by clicking to select a key along the top of the piano roll.
3. Choose a scale (such as major, minor, or any of the dozens of available scales and modes) from the
Scales drop-down.
n You can create and save your own custom scales in Riffer. For more information, see "Scales" on
page 404.
Create a Riff
To have Riffer create a riff:
t Click the Dice icon above the Piano Roll.
Notes are generated in the chosen key and scale, and appear in the Piano Roll.
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Shape a Riff
Do any of the following to modify and shape generated riffs and pattern:
t Length (Steps, Loop, and Number of Steps)
To set the number of steps in the pattern, or set the length of the loop, use the Steps, Loop, and
Number of Steps selectors.
t Range
To set the high and low range of notes to be included in the riff, adjust the Pitch Range slider.
t Octave and Step Transpose
To move the notes up or down in octaves, click the desired button below the Piano Roll. To nudge up
or down in single steps (semi-tones) click the ^ and v up/down arrows to the right of the Octave
buttons.
t Shift
To move the notes earlier or later, click the < and > buttons below the Piano Roll.
t Randomize Duration, Velocity, and Density

Click to select Duration, then click the Dice icon. Duration is randomized. Repeat for Velocity and
Density.
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To protect a parameter from being randomized:
1. Click to select it.
2. Click the Lock icon above the Piano Roll.

t Tie: Turning on the Tie notes setting lets you change repeated notes into held notes.
Sustain
Sustain lets you have Riffer randomly tell certain notes to sustain for longer.
Lock
To anchor specific notes so they are not randomized:
1. Click to enable the Lock Notes button.
2. Click to select the desired notes in the Piano Roll.
Locked notes display a teal (blue/green) color. When you randomize notes, they stay anchored in place.
Note that their Duration and Velocity continue to be randomized.
To prevent Duration and Velocity from being randomized:
1. Click to enable the Lock Step button.
2. Click to select the desired notes.
These notes now display a dark blue color.
g To unlock notes, click the Back arrow.

More to Explore
l You can draw additional notes in the Piano Roll. To make sure they all follow the chosen key and
scale, click the Keyboard icon to conform all notes.
l Polyphonic mode lets you use up to four independent riff generators per instance of Riffer. Click a
Riff button (R1, R2, R3, or R4) to configure up to 4 independent riffs, or link them together. To mute or
unmute a riff, double-click its Riff button.
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l To vary playback of the generated riffs use the Forward, Backward, or Ping-Pong modes. Click the
desired icon to have the riff play forwards (the default), backwards, or alternate between forwards
and backwards (ping-pong).
Top Bar Controls

The top bar displays the current tempo in BPM (beats per minute, including a tempo adjustment slider and a
fine adjustment knob) and other controls.
Info Tab
Click the Info tab to display the Riffer version number and credits. Click the Info window to close.
Piano Roll
Keyboard Mode
See "Keyboard Mode" on the next page.
Tempo
The top bar displays the current tempo in BPM (beats per minute).
Tempo is adjustable only in the iPad version of Riffer. When using the Riffer plug-in, tempo will
automatically sync to the Pro Tools session tempo.
Undo
Click to undo the previous change.
Panic
Click to reset MIDI if you encounter issues, such as hanging MIDI notes.
Learn
Lets you quickly map Riffer parameters to a MIDI controller. See "MIDI Learn" on page 407.
Presets Tab
Lets you save Riffer setups as Presets. See "Saving Presets" on page 407.
Settings Tab
Lets you customize Riffer. See "Settings" on page 410.
Resize Window
Lets you quickly return Riffer to its default window size.
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Piano Roll
Riffer includes 4 independent piano rolls to control Pitch, Duration, Velocity and Density, to send note and
automation data to associated plug-in instruments and hardware. These 4 parameters are displayed on the
vertical axis and time on the horizontal axis.
You can view, edit and activate each piano roll, by selecting any of the R1, R2, R3 and R4 buttons. Double
click or Command-click (Mac) or Control-click (Windows) on R1, R2, R3 or R4 to activate or deactivate a
piano roll.
Enable the ALL button, to apply changes to all 4 piano rolls at once.
Keyboard Mode
Riffer 3.0 and later can be controlled from MIDI messages.
While the Keyboard Mode is enabled, tonality can be modified and the sequence can be turned ON/OFF for
the selected Riff. By enabling the ALL button, the tonality for all four Riffs will be modified. Changing the
tone from MIDI messages (such as keyboard or MIDI note messages from your DAW) will keep playing the
sequence. The entire sequence will be transposed in real-time from the keyboard, but always in key,
depending on the type of Scale and Root.
Enable the ALL button, to modify tonality for all 4 Riffs at once
In order to use the Keyboard Mode through your DAW, activate the Monitoring option, in order to allow the
track's input to be played through its device and heard at its output.
Randomization
Randomize the Pitch, Duration, Velocity, and Density for the selected piano roll by clicking on the
Randomization (Dice) button. You can randomize these parameters for all piano rolls at once by enabling
the ALL button.
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Reverse, Lock, Erase
To the right of the Dice icon are buttons for Reverse, Lock, and Erase.
Reverse
This will reverse all steps for Pitch, Duration, Velocity and Density for the selected piano roll.
Lock
While enabled, the selected tab (Pitch, Duration, Velocity, and Density) will not be randomized by clicking
on the randomization button.
Erase
This will erase all settings for the selected piano roll.
Playback Motion
To the left of the Dice icon are buttons for Forward, Backwards, and Back and Forth. These select the
Motion of the Playback.
Infinity Mode
While enabled, the infinity mode takes full control of the selected riff/piano roll and generates a completely
new pattern each time a new pattern starts (loop) so you can just sit back and have it perform for you.
You can also select how many times the pattern shall remain the same until it re generates anew one.
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For example: if X2 is set, then each pattern shall play two (2) times, until the mode regenerates a new
pattern and so on.
You can set different modes for each riff. While the ALL button is enabled, you can activate the Infinity
mode for all 4 piano rolls at once.
Use the right and left arrow buttons to increase or decrease the Infinity Mode number, or scroll up or down
(drag) the Infinity Mode number to make faster changes.
Lock Notes and Lock Steps
Lock Notes
While the Lock Notes button is enabled, you can select specific notes in the piano roll to be locked (click on
these notes directly on the piano roll). This will prevent the steps for the currently selected tab (from Pitch,
Duration, Velocity and Density) from being randomized by clicking on the Randomization (dice) button. For
example: If the Pitch tab is selected, only the Pitch parameter will be locked for the locked steps. By clicking
on the Randomization button, the Duration, Velocity and Density will be randomized, but the pitch will
remain the same.
Lock Steps
While the Lock Steps button is enabled, you can select specific notes in the piano roll to be locked (click on
these steps directly on the piano roll). This will prevent the Pitch, Duration, Velocity and Density for these
steps from being randomized by clicking on the Randomization (dice) button.
Reset
This unlocks all locked notes and steps in the piano roll.
Piano Roll Controls

Number of Steps
Here you can adjust the number of active notes that your riffs will include by clicking on the Randomization
button.
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Quantization
Through this drop down menu, you can select time signature for the selected piano roll.
Shuffle
Add Shuffle amount to your riff.
Steps and Loop
These two selectors control the step amount of your pattern. You can select from 1 to 64 steps and adjust
the loop.
Root Note
You can set the number of root notes that your new riff will contain. For example: if the Root Notes number
is 4, by clicking on the Randomization button, all riffs that will be generated will contain 4 root notes. You
can deactivate this feature using the On/Off toggle button.
Tie Notes
While enabled, same notes that are next to each other will be played as one sustained note.
Sustain Notes
While enabled, paused notes will be played as sustained.
C – B Keys
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Click any of the C B keys to transpose your riff(s) to a selected key.
Riff Start and End
Select the starting and ending points for your riff.
Octave Buttons
Octave
Move your riff up to +/ – 24 semitones Up or Down.
Semitones
Move the riff one step up or down.
Shift
Move the riff one step to the right or left. Move the riff one step up or down.
MIDI Export
This lets you Drag Out the MIDI pattern directly to your Pro Tools session. You can export the MIDI pattern
for the selected riff, or enable the ALL button to export the MIDI pattern for all 4 riffs.
Scales
Riffer provides the option to create riffs using up to 57 scales, or to create your own custom scales.
Use the Scales drop down menu to select your preferred scale for new patterns that will be generated using
the Randomization button.
To create your own custom scales:
1. Choose the Save new scale option in the Scales drop-down menu.
2. In the piano roll, include the notes in your custom scale.
3. Click Save new scale.
4. Type a name and click Save.
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5. The new scale is included in the Scales menu.
To delete a custom scale:
t Click the delete (Trash) button on the right.
Keyboard Button
By clicking the keyboard button, your current Riff will be transported to the selected scale.
Scales Included:
Chromatic, Blues, Major, Minor, Dorian, Mixolydian, Phrygian, Lydian, Locrian, Harmonic Minor, Melodic
Minor, Pentatonic Neutral, Pentatonic Minor, Pentatonic Major, Dim Half, Dim Whole, Augmented,
Roumanian Minor, Spanish Gypsy, Diatonic,Double Harmonic, Eight Tone Spanish, Enigmatic, Algerian,
Arabian A, Arabian B, Balinese, Byzantine, Chinese, Egyptian, Hindu, Hirajoshi, Hungarian Gypsy, H. Gypsy
Persion, Japanese A B, Persian, Prometheus, Six Tone Symetrical, Super Locrian, Wholetime, Major triad,
Minor triad, Major 7th, Minor 7th, Major Dominant, Minor Major 7th, Diminished, Major 6th 9th Add9
Augmented, Minor 6th 9th Add9 Augmented.
Randomizing the Pitch

Pitch Tab
By selecting the Pitch Tab in the piano roll you will be able to see and edit the Pitch of the notes that are
included in your Riff. You can click the Randomization (dice) button to generate new riffs, or click on the
steps directly in the piano roll, to manually create your own riffs.
Pitch Range
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Adjust the Pitch Range of the piano roll. Click on the + or – buttons to increase or decrease the Pitch Range,
or use the vertical blue slider to make changes.
Set the Same Pitch Value to All Steps
Alternatively, click a note directly in the piano roll, hold Shift and drag to the left or right, to set the same
pitch for more than one step in your riff.
Use the Vertical slider to the right of the Piano Roll to scroll the piano roll up or down.
Randomizing the Duration

Duration Tab
By selecting the Duration Tab, in the piano roll you will be able to see and edit the Duration of the notes that
are included in your Riff.
Use the horizontal blue slider below the Duration button to set the same Duration value to all steps.
Duration Range
When the Duration tab is selected, the vertical slider to the left of the Piano Roll controls Duration Range.
Use it to adjust the Duration Range for the steps. Set min and max values for Duration using the vertical blue
slider.
Randomizing the Velocity

Velocity Tab
By selecting the Velocity Tab, in the piano roll you will be able to see are included in your Riff.
Use the horizontal blue slider below the Velocity button to set the same Velocity value to all steps.
Velocity Range
When the Velocity tab is selected, the vertical slider to the left of the Piano Roll controls Velocity Range. Use
it to adjust the Velocity Range for the steps. Set min and max values for Velocity using the vertical blue
slider.
Randomizing the Density

This feature allows you to split specific steps to 2-8 parts/hits.
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Density Tab
By selecting the Density Tab, in the piano roll you will be able to see and edit the Density of the notes that
are included in your Riff.
Use the selector below the Density tab button to adjust the number of steps that will have Density in your
generated Riffs.
Click on the Reset button to remove density for all steps.
Density Range
When the Density tab is selected, the vertical slider to the left of the Piano Roll controls Density Range.
Adjust the Density Range for the steps. Set min and max values for Density using the vertical blue slider.
Automation
MIDI Learn
MIDI Learn allows you to remote control virtually any on screen parameter with a MIDI controller. The basic
process of pairing a physical control to a Riffer parameter using MIDI Learn is extremely simple.
1. In the Info bar, click the 'Learn' button. The controls that can be automated are highlighted.
2. Select a parameter to be activated for the MIDI Learn mode.
3. Move a slider/knob on your MIDI Controller to associate it. The MIDI CC number of the associated
control will be visible in Learn mode, below the Dice icon.
4. Click 'Learn' to exit the MIDI Learn mode
MIDI CC Mappings
By clicking the Top right Settings (Gear) button, you can have access to the MIDI Mappings list. The MIDI
Mappings list includes all parameters that can be controlled and provides the option to associate a
parameter to your MIDI Controller.
Clear Mappings
By clicking the 'Learn' button to enter the mode, you can click the 'X' button from any associated parameter
clear mapping. Alternatively, long click on the parameter to mapping.
In order to clear all mappings at once in MIDI Learn mode, click the Clear Mappings button at the bottom of
Riffer.
Alternatively, you can clear all mappings by clicking the Clear button that is included in the MIDI CC
Mappings section.
Saving Presets
In Riffer you have two options to store/save your custom presets. The traditional method requires you to
click the save (pencil) button on the top right. Name your preset and click 'Save'.
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Quick-Load Slots
The second option is what we call Quick Load slots.
To show the Quick Load Slots:

t Click the Presets icon in the lower left corner of Riffer.
The Quick-Load slots appear along the bottom of the window.
Once you have a pattern that you want to save, simply by clicking to an empty slot will automatically
save the current pattern to this slot number. Once a saved preset is there, it can be recalled with a
press of a button instantly, rather than choosing it from the ''load presets'' top right menu.
This is extremely helpful in live/jam situations where you can load up to 16 different presets and quick
fire them with a touch of a button. The slot/preset that is currently playing will be displayed with blue
color.
Presets Tab
All your saved presets will be included in the presets menu. In this menu, you can double click to load your
saved presets, export/share to any location, delete, or assign your saved presets to a slot in the Quick Load
presets tab.
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To assign preset to a slot in the Quick-Load Presets tab:
t Click to the left of the Preset name and choose a slot.
Export
This lets you store your presets to another location, or share presets between devices. To export a Preset,
click its Export Preset icon.
Delete Preset
To delete a Preset, click its Trash can icon.
Import Presets
By clicking on the Import Presets button, you can import to Riffer a preset from any location.
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Save as MIDI file
You can save your currently playing riff as MIDI file in any location on your device
Settings
By clicking the top right Settings (Gear) button, you can have access to MIDI settings.
MIDI Input Channel Selector

Determines the MIDI input mode/channel for Riffer.
External MIDI Outputs selector

Through the plug-in MIDI Output drop down menu, you can select which of your 4 Riffs will control an
external instrument plug-in or hardware associated with Riffer. You can also select All to allow all 4 riffs to
control the external software/hardware.
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Mixed in Key MIDI Plug-ins
Pro Tools includes two MIDI plug-ins from Mixed in Key: Human Lite and Captain Chords Lite.
n For more information about these plug-ins, visit Mixed in Key.

Human Lite
Human Lite MIDI from Mixed in Key lets you add an organic feel to your MIDI loops and sequences.
Captain Chords Lite

Captain Chords Lite is a music composition MIDI plug-in that helps you write chord progressions for MIDI
quickly and easily.
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Modalics EONarp
EON-Arp for Pro Tools by Modalics is an arpeggiator MIDI effect plug-in that lets you shape intricate,
musical phrases with ease. EON includes over 100 chords, progressions, and scales to assist you with your
musical creativity.
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MIDI Signal Flow
The Chords section sets the input source and controls the overall MIDI output. After the chords section, MIDI
is affected and shaped by the Arp Grid. The MIDI output is then sent to a Virtual Instrument or an external
MIDI device.
Top Toolbar
The top toolbar provides the following controls.
l Swing Knob: Adjust the amount of swing of the arpeggiator. Swing is global and affects all patterns in
the current preset.
l Swing Reference: Set the rhythmic value for swing reference.
l Preset Menu: Click to open the preset menu. Use the arrows to navigate the presets in order.
l Tempo (BPM): Displays the Pro Tools session tempo in beats per minute.
l About Modalics: Click to learn about Modalics.
Pattern Snapshots
This section lets you select pattern snapshots that follow the trigger and quantize value settings in the
Chords section. Each preset can hold up to 8 pattern snapshots that can be independently triggered using
key switches (MIDI notes). Snapshots are saved dynamically while you are editing the pattern.
To trigger any given Snapshot, do one of the following:
t Click the snapshot you want.
t Play the corresponding MIDI note (key switch). Use key switches to select any snapshot: C0 to C1. The
corresponding MIDI note is displayed in the Right-click menu for each pattern (C0–C1).
The timing for switching patterns is determined by the Q Value setting in the CHORDS section.
Right-click to Copy and Paste one snapshot to another. You can also Right-click to select a pattern.
g Press C and click to copy any pattern. Press V and click another pattern to paste.
Editing Toolbar
Q (Quantize Notes)
Click to quantize the starting points and lengths of selected notes to fill an entire step. If no notes are
selected, all notes on the grid are quantized.
Snap
Click to enable (or disable) snap to grid.
Pattern/Step Length
You can set the number of steps for the Arp grid from 1 to 128.
Division Mode
In Division mode, the values of the Numerator and Denominator determine the overall pattern length,
meaning that the length of steps are adjusted to fit within the measure specified by the Numerator and
Denominator.
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For example, if the pattern is x length, and you divide it by a number of steps, so with a pattern length of
1/4, these are the length of each step:
l 1 step = 1/4 note
l 2 steps = 1/8th note
l 3 steps = 1/8th note triplet
l 4 steps = 1/16 note
l 5 steps = 1/8 note quintuplet
Multiply Mode
In Multiply mode, the values of the Numerator and Denominator determine the length of a single step,
meaning that each step has the length of the measure specified by the Numerator and Denominator. In
Divide mode, if you specify a musical duration for your pattern and then create a division from it. In Multiply
mode you specify how long each step is, and then select the number of steps to work with.
Listen to changes in the Length section against a metronome click to hear the rhythmic context of the Arp
settings. Some settings may feel “off” for a short while and might take a moment to sync with the click. This
can actually result interesting and unique rhythmic grooves.
ARP
Click ARP to toggle bypass of EON-Arp on or off.
Brush Module
Cursor Brush
The default brush. Use the cursor brush to:
l Place notes: Double click to add a step.
l Move notes: Click and drag up or down, or left or right.
l Lasso selection: Click & drag on empty step
l Set step length: Click & drag on step start/end points
Pencil Brush
The pencil brush can be used to “color in” or erase steps quickly and easily.
Velocity Brush
Click and drag up or down to adjust note velocity, you can also lasso select multiple notes and adjust their
velocity simultaneously.
Undo/Redo
Click the Left arrow to undo the last action. Click the Right arrow to redo the last action.
Clear
Click Clear to clear the Arp grid.
Arp Grid
The Arp grid provides a piano roll style interface for drawing and editing notes. Incoming MIDI notes trigger
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When you input MIDI notes into EON-Arp, it triggers playback of the notes in the Arp grid. The notes in the
Arp grid playback relative to the pitch of the incoming MIDI note(s). There are 11 pitch positions in the note
grid: –5 to +5 degrees of transposition. The spread of these degrees of transposition are determined by the
settings in the Chords section.
For example, if you input the MIDI note 60 (C3) into EON-Arp, the Arp Grid places C3 in the center and
spreads the note across all 11 degrees within the octave range dictated in the Chords section. When in full
range, the resulting grid play notes C0–C6.
g The –1 octave is reserved for switching between patterns.

When playing a chord into EON-Arp, such as a C major triad with the notes C2, E2 and G2, EON-Arp
allocates the 3 notes in the chord according to their pitch relative to each other. Their order is determined
by how the chord is played with C2 being the lowest note, followed by E2 and G2. The grid starts in the
same order, but 1 octave above (C3, E3, G3). The same goes for the pitches below the center C2.
b Create several notes in the Arp grid. Then select the notes and Option-click (Mac) or Alt-click
(Windows) and drag to copy the selected notes to another location on the grid. This way you can
create repetition and variation in your melodies/sequences.
Chords
In the Chords section you can set input settings, pattern triggering settings, play or import midi files,
transpose keys and/or set an octave range for the Arp grid.
Input
The input toggle switches between 2 modes of operation:
l EXT: EON responds to MIDI input from Pro Tools.
l FILE: EON responds to input from the MIDI player module. Select one of the built in files or import a
custom MIDI file using the folder icon.
Retrigger
Click Retrigger to toggle On or Off. When enabled (ON), MIDI input re-triggers the current pattern. When
disabled (OFF), the current Arp pattern plays continuously regardless of MIDI input.
Q Value
Select the desired rhythmic value for re-triggering the Arp pattern and for switching patterns: 1/32. 1/16,
1/8, 1/4, 2/4, or 4/4.
MIDI File Player

The MIDI File player comes equipped with more than 100 MIDI files in various styles, you can also load your
own MIDI files onto the player.
Transpose
l Pitch: Controls the overall pitch output from EON in semitones.
l Oct: Controls the overall pitch output from EON in octaves.
Octave
Set the Min and Max values for octaves for the grid to consider when allocating pitch values. Oct numbers
are absolute and relate to the MIDI note numbers (C1, C2, C3, and so on).
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Pitch Innovations Groove Shaper
Groove Shaper Lite for Pro Tools by Pitch Innovations is a creative sequencer for generating a vast variety
of rhythms for hi-hats, bass-lines, short string motifs, and much more.
Header Controls
The following controls are available along the top of the plug-in window.
Time
Click to reveal a list of available time signatures for the Shape Sequencer.
Bars
Click to modify the loop length of the sequencer. Four steps are equal to one bar.
Preset
Click to load sequencer presets. Groove Shaper includes a collection of factory presets in multiple
categories.
Mode
l AUTO (DAW/PERFORM): When selected, Groove Shaper intelligently switches between DAW and
Perform modes based on how you are using the plug-in. When Pro Tools is not playing back, Groove
Shaper enters PERFORM mode, which optimizes for live performance with MIDI devices. Conversely,
when Pro Tools is playing back, Groove Shaper switches to DAW mode, which is ideal for working
with MIDI clips in your Pro Tools.
l PERFORM: This mode is optimized for live playback using MIDI devices. Once the Groove Shaper
receives MIDI input, the sequencer activates instantly, and stops immediately when the MIDI data
input stops. The sequencer will reset to the top when MIDI input is not received. This ensures smooth
and precise performances from your MIDI input device, allowing for seamless play and stop
transitions.
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l DAW: This mode is specifically tailored for use when a MIDI clip is present in Pro Tools (DAW). Similar
to Perform mode, Groove Shaper starts sequencing when it receives MIDI input, but it waits for a brief
buffer period after MIDI input. If no further MIDI input is detected during this buffer, the sequencer
stops, ensuring clean sequences or playback without overlapping notes.
l SHAPE HOLD: In this mode, Groove Shaper loops a single shape when it receives a MIDI input signal.
When MIDI input re-triggers, Groove Shaper moves to the next shape and continues looping that
shape until the user re-triggers it.
Arp
Click to trigger an arpeggiation of each note in the sequence, ascending in pitch.
Settings
Click here to access the settings tab, where you can transpose, select a skin, and modify other settings (see
“Settings” on page 446).
Shape Blocks
Groove Shaper uses shape blocks to represent rhythmic patterns. Groove Shaper lets you choose to view
Shape Blocks as shapes, bars, or rhythms.
Rhythms available are as follows: 1 quarter note (circle), 2 eighth notes (2 overlapping circles), 3 triplet
eighths (triangle), 4 sixteenths (square), 5 quintuplet sixteenths (pentagon), 6 sextuplet sixteenths
(hexagon), 7 septuplet sixteenth notes (septagon), and 8 thirty-second notes (2 overlapping squares).
Sequencer Shapes
Drag and Drop Shapes to the Sequencer

The Shape Sequencer functions as a step sequencer, allowing users add each Shape Block into a step. The
Shape Sequencer is organized in 2 rows, each of which provides slots for rhythmic shapes depending on the
selected Time Signature (meter).
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To add Shape Blocks to the Shape Sequencer:
t Drag and drop any of the lower eight Shape Blocks to the desired step in the Shape Sequencer.
Adding a shape to the 3rd bar of a sequence
Move the mouse cursor over any Shape Blocks to display the rhythm that they represent in Western
classical music notation.
The sequencer begins looping every 4 beats (a bar) and automatically extends loop length when
adding shapes. You can adjust loop length using the bar selector.
c Groove Shaper Lite for Pro Tools is limited to a maximum of 2 bars.

To clear all steps:
t Click Clear All Steps. Note that this option is only visible when the Sequencer settings tab is open.
To clear a single step, do one of the following:
t Select the Shape in the step you want to clear and click Clear Step. Note that this option is only
visible when the Shape settings tab is open.
t Right-click the step you want to clear.
To move Shapes Blocks in the Sequencer:
t Drag and drop the Shape you want to move over another.
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Sequencer Settings
l ON/OFF: Click to bypass or activate the sequencer.

l Velocity: Use this slider to adjust the global velocity of the sequencer.
l Gate: Use this slider to adjust the global gate length of the sequencer.
l Swing: Adjust this slider to infuse your rhythm with the desired amount of swing.
l Dynamics: Adjust to alter velocity values by adjusting the differences between soft and loud notes
(dynamics) using a percentage value, rather than adding or subtracting a fixed amount. This is
similar to the operation of an audio compressor or expander.
l MIDI Capture: Lets you to record the output of Groove Shaper and generate MIDI files, that can then
be dragged and dropped into your Pro Tools session.
To capture MIDI:
t Click CAPTURE MIDI in the MIDI Capture section.
t Play your MIDI device or a sequenced MIDI input from Pro Tools. Recording starts as soon as Groove
Shaper receives a MIDI signal.
t Click STOP CAPTURING to complete recording.
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t Click the DRAG & DROP button and drag the MIDI file to the Pro Tools timeline.
Drag and drop captured MIDI
Shape Settings
Shape settings (Note Length shown)

l Note Velocity: Lets you to fine-tune the velocity of each note within a shape. Each bar corresponds to
an individual note and the number of bars varying based on the shape type. These bars can be
adjusted by dragging them vertically to modify the velocity value, ranging from 0 to 127.
l Enable/Disable Note Repeats: Right-click on a bars to toggle notes on and off.
l Reset Button: Click to reset all velocities to the default value.
l Note Length: Lets you edit the note length for each individual note within a shape. Similar to Note
Velocity, each bar represents a note repeat. Drag these bars vertically to edit the note length, which
ranges from 0 to 100. Right-click on a bar to toggle the notes on and off.
l Note Tie: Click to create a tie between the note and the next adjacent note.
l Note Length/Note Delay: Click to toggle between Note Length and Note Delay.
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l R (Randomize): Click the R button to randomize shape setting properties such as Note Mute, Note
Velocity, and Note Length within the selected shape.
l Note Delay: Adjust to fine-tune the delay of the shape, with a range spanning from –127 to 127 ticks
(measured in milliseconds according to the sample rate of the Pro Tools session).
Shape settings (Note Delay shown)
Magic Dice
The Magic Dice button lets you generate fresh rhythms with six algorithms to choose from. These algorithms
are tailored to different types of instruments: Bass, Hi Hats, Epic Drums, Percussion, Drum Fills, Synth, and
Short Strings.
Magic Dice (Hi Hats)
To generate rhythmic patterns using the Magic Dice:
1. Click the left/right buttons to select the desired instrument category.

2. Click the Magic Dice button.
The Shape Sequencer is populated with algorithmically generated rhythmic patterns according to the
selected instrument category.
Bottom Toolbar
l Undo/Redo: Click the Undo or Redo button to undo or redo the last adjustment you made within the
plug-in.
l Help Text: Displays information about any user interface element when you move the mouse cursor
over it.
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l Shape Views: The default view visualizes rhythms using shapes. An asterisk (*) in the upper right of
the symbol indicates that it has been customized/edited.
Shape view (* indicating custom Shape settings)
l Bars View: This view displays rhythmic patterns with vertical bars. You can edit note on/off
parameter within steps by clicking the bars. Any changes are mirrored in the Shape settings.
Bars view shown with custom Shape settings
l Score View: Shows rhythmic patterns in the Western music notation. An asterisk (*) in the upper right
of the symbol indicates that it has been customized/edited.
Score view shown with custom Shape settings
l BPM (Sequencer Speed): Lets you halve (1/2), match (x1), or double (x2) the speed of the sequencer in
relation to the Pro Tools session tempo.
l Key Limit: Lets you set the input note range for the sequencer. This lets you bypass the sequencer for
any MIDI notes outside the specified range.
Settings
Click the Settings icon in the Header Controls to access the Groove Shaper settings.
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Global Settings
Tempo
l Internal: Lets you set an independent tempo within the sequencer.
l Ext (DAW): The sequencer tempo synchronizes to the Pro Tools session tempo.
Transpose
Adjust to transpose the input MIDI signal up and down by two octaves within the range of –24 to +24
semitones.
Help Text
Enable (or disable) to turn the help text view on (or off).
Advanced Settings
Use MIDI Input Velocity
Enable (or disable) to input (or bypass) MIDI velocity for the internal sequencer.
Display Note Values on Hover
Enable to hover over the shapes with the mouse in the Shape palette.
Select Last played Shape
Enable to automatically open the shape settings of the last played shape in the sequencer.
Sustain Pedal Enable/Disable Seq
Enable to bypass the internal sequencer or bypass the key limit when the sustain pedal is engaged.
Choose Skin
l Aqua Glow: Select for the Aqua Glow user interface color scheme.
l Dark Velvet: Select for the Dark Velvet user interface color scheme.
Send Anonymous Usage Date
Enable to send anonymous usage data to Pitch Innovations.
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13 Other Plug-Ins
Plug-ins in the "Other" category include many useful utility processors including tuners, metering plug-ins,
and more.

l "InTune" below
l "MasterMeter" on page 430
l "Signal Generator" on page 438
l "SoundReplacer" on page 439
l "Time Compression / Expansion" on page 447
l "Trim" on page 448
InTune
InTune is a professional instrument tuner plug-in that is available in DSP and Native formats. It offers the
features and performance of a rack mounted digital tuner in the convenience of a plug-in. 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.
To use InTune with Pro Tools, simply create a new mono audio or Auxiliary Input track in Pro Tools, and
select InTune from the plug-in menu for that track.
When 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.
By default, InTune loads the Chromatic tuner preset. This displays all notes in the scale and automatically
displays the required octave.
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.
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InTune Controls and Displays
InTune Auto Button
Click the Auto button to toggle Automatic Mode on and off. When Automatic mode is active, InTune will
detect the note played and automatically show the pitch for that note.
To enable Automatic mode:

t Click the Auto button to enable Automatic mode. The Auto button highlights.
To tune to a single note and turn off Automatic mode:
t Click the button for a note.
Selecting a note
This turns off automatic mode. InTune will now display pitch relative to the selected note only.
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InTune Test Tone Menu Selector
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.
To hear a test tone:
1. Select Sine, Triangle, or Audible from the Test Tone selector
Selecting a test tone
2. Click the Note button for a note.

3. Adjust the Tone Volume slider.
When a test tone is playing, “Tone Playing” appears in the information display.
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. "Creating InTune Tuning Presets"
on page 428
InTune Meter Selector

The Meter selector lets you use a standard needle style meter or a strobe style display.
To select the Meter display:

t Select Needle or Strobe from the Meter selector.
Selecting Meter display
Strobe Display
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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.
InTune Reference Frequency Control
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.
InTune Note Buttons

The Note buttons provide two functions:
l When in automatic mode, clicking on a note button will turn off automatic mode and InTune will now
display pitch relative to the selected note only.
l 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.
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
Octave buttons
The octave range of 0–8 displayed in InTune is based on middle C being equal to C4. In chromatic presets,
you can select a tuning octave from 0–8 using the arrows at each end of the note display.
InTune Tone Volume

The Tone Volume slider controls the volume of the test tone audio signal.
InTune Information Display

The LCD style information display in InTune displays the following:
l The reference frequency
l The current note to which InTune is tuning
l The number of cents sharp or flat from the current note
l The status of any test tones playing
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InTune Presets
InTune provides a selection of factory presets for stringed instruments. These presets can be selected from
the Plug-In Librarian menu.
Selecting a InTune preset
To make any preset the default when InTune is instantiated:
1. From the Plug-In Librarian menu, select a preset.

2. 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.
n For more information on using plug-in presets in Pro Tools, see the Pro Tools Reference Guide.
Creating InTune Tuning Presets
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 InTune screen, click the Edit button to display the Tuner Programming screen.
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.
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Single Octave Mode
When selected, Single Octave Mode disables the display of octave information with each note on the
main InTune screen. When tuning in this mode, 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
InTune to tune to a higher harmonic of the note instead.
Display Flat Semitones
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, 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
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. 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 InTune screen.
Using InTune
When InTune detects a signal, the meter lights up and displays the relative pitch of the incoming signal.
With stringed instruments, this will vary during the attack and decay of the note.
In Automatic mode, 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 InTune
to the specified note.
The meter will display the frequency of the note detected, and the accuracy is displayed on a scale of
plus/minus 50 cents. In addition, the information display will display the note and the number of cents from
perfect tuning.
When loading factory presets, stringed instruments are laid out from the highest numbered string (usually
the lowest tone) to the highest, from left to right. For example, a six string guitar in standard tuning is shown
as E2, A2, D3, G3, B3, E4, which are the notes and octaves for the sixth string through to the first string
respectively.
For best tuning results with guitars, do the following:
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l Use headphones, as loud monitors can modulate the guitar string.
l Switch your guitar to its rhythm (neck) pickup, if it has one.
l Roll your guitar’s tone knobs all the way off to remove all the highs.
l 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 a Note button to produce the appropriate test tone.
Test tones can be routed to headphones as required for musicians during session.
MasterMeter
MasterMeter is an oversampling meter plug-in that is designed for critical mixing and mastering
applications. MasterMeter is available in DSP and Native formats.
MasterMeter Overview
This section provides an overview of metering and mastering, and how MasterMeter can help you produce
great sounding mixes.
Understanding Digital Distortion

Clients in the music industry regularly demand the loudest possible mixes. In the process of achieving such
a “hot mix,” unwanted distortion can be introduced. Intersample peaks that exceed 0 dB may play without
distortion in a studio environment, but when the same mix is played through a consumer CD player, the
digital to analog conversion and oversampling process can reproduce a distorted mix.
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.
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Sampling
A waveform can be represented in multiple ways during the process of sampling, display and
reconstruction.
The following four figures show how the same complex waveform shown in the previous figure can be
represented in the digital domain.
A complex waveform
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A waveform sampled
Waveform as represented in DAW
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 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 1800s 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.
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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. Until this process has been
completed, the waveform is inaccurately represented by the samples. This is the reason that in most DAWs
the waveform is represented on the screen as a “dot to dot” connection between sample points. They do not
undergo the reconstruction process inside the system, so all that can be represented is the sample points
and for the sake of visual ease, they connect the dots between them with straight lines. They save the
reconstruction process for the digital to analog converters.
Intersample peaks
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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.
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 as loud as possible out of a
consumer’s system. Hence, it is very common for popular music CDs to be full of digital samples that are at,
or nearly at full scale.
The problem is realized in that while going through these digital gyrations and utilizing digital tools to
amplify the signal as much as possible, both during mixing and during mastering, the “peak value” of the
sample points is closely watched to ensure that it does not get to full scale. Since the peak meters in said
DAW and digital mixing systems are inaccurate, and do not actually indicate the peak values of the
resulting waveform, the result is that while the samples themselves do not exceed full scale and are
carefully monitored to ensure this, the resulting waveforms represented by the samples may exceed full
scale throughout any standard CD!
While the digital mixing system is not clipping the music or distorting the music, the digital to analog
converters that have the task of recreating the audio through digital reconstruction filters are clipping
repeatedly throughout most CDs on the market. The result is that most CDs and DVDs end up distorting
with regularity when they are asked to reconstruct and play back audio that appears to be completely
“legal” because not a single sample actually clipped.
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.
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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 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
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.
Using MasterMeter
MasterMeter uses the DSP power of Pro Tools to model the conversion process found in typical consumer
devices. In technical terms, the 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.”
MasterMeter can be used in two different ways during a session: Real-Time Metering or Historical Metering.
Real-Time Metering
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, 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, 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.
MasterMeter Controls and Displays

MasterMeter Browsers
Signal Clip Events Browser
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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 a 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 MasterMeter is being used for historical metering but the session
was started from a point other than the beginning. If 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 MasterMeter reaches
2,000 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 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
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 a
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 MasterMeter is being used for historical metering but the session
was started from a point other than the beginning. If MasterMeter is being used in real time, the timecode
information in this column can be ignored.
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.
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Once MasterMeter reaches 2000 clip events, it ceases to record additional events. Although the meters
remain active and the Peak field continues to be updated, new events will not be added to the browsers. The
Events field flashes ‘2000’ to indicate this condition.
The Oversampling field displays the current oversampling factor in use by the DSP processing. This will vary
between 2x, 4x and 8x oversampling depending on the session sample rate.
The information in this browser is cleared using the Clear button, or is cleared automatically whenever the
Pro Tools transport is started.
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.
MasterMeter Clear Button

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.
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.
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.
MasterMeter Offset Field

The Offset field offsets the values displayed in both the browsers by the value entered. This is useful for
historical metering but the session was started from a point other than the beginning. The Enter key must be
used after a new offset is typed for it to become active. The information shown in the browsers is updated
immediately when the new Offset is entered.
For example, if the session was started from the point 1:03.901 (1 minute 3.901 seconds), this value should be
entered into the Offset to ensure the timecode displayed in both of the browsers matches that of the
Pro Tools session.
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 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.
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Signal Generator
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 is a mono (or multi-mono) plug-in that is available in DSP, Native, and AudioSuite formats.
Signal Generator supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
n Refer to the guide for your audio interface for instructions on using Signal Generator to calibrate that
interface.
g Signal Generator produces a tone as soon as it is inserted on a track. To mute the Signal Generator,
use the Bypass button.
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. Choices are sine, square, sawtooth, triangle, white noise, and pink
noise.
c 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.
g Select the Create Continuous File option for greater flexibility in making audio selections for use with
the Signal Generator plug-in.
g 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.
SoundReplacer
SoundReplacer is an AudioSuite only 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:
l Sound replacement with phase-accurate peak alignment
l Intelligent tracking of source audio dynamics for matching the feel of the original performance
l Three separate amplitude zones per audio event for triggering different replacement samples
according to performance dynamics
l Zoomable waveform display for precision threshold/amplitude zone adjustment
l Crossfading or hard-switching of replacement audio in different amplitude zones for optimum
realism and flexibility
l Online help
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Audio Replacement Techniques
Replacing audio elements during the course of a recording session is a fairly common scenario. In music
production it is often done in order to replace or augment an element that lacks punch. In film or video post-
production it is typically done to improve or vary a specific sound cue or effect.
In the past, engineers and producers had to rely on sampling audio delay lines or MIDI triggered audio
samplers—methods that had distinct disadvantages. Delay lines, for example, support only a single
replacement sample, and while they can track the amplitude of the source events, the replacement sample
itself remains the same at different amplitude levels.
The result is static and unnatural. In addition to these drawbacks, sample triggers are notoriously difficult
to set up for accurate timing.
Similarly, with MIDI triggered samplers, MIDI timing and event triggering are inconsistent, resulting in
problems with phase and frequency response when the original audio is mixed with the triggered
replacement sounds.
The SoundReplacer Solution
SoundReplacer solves these timing problems by matching the original timing and dynamics of the source
audio while providing three separate amplitude zones per audio event. This lets you trigger different
replacement samples according to performance dynamics.
Each replacement sample is assigned its own adjustable amplitude zone. Variations in amplitude within the
performance determine which sample is triggered at a specific time. For example, you could assign a soft
snare hit to a low trigger threshold, a standard snare to a medium trigger threshold, and a rim shot snare to
trigger only at the highest trigger threshold.
Replacement samples that are triggered in rapid succession or in close proximity to each other will overlap
naturally—avoiding the abrupt sound truncation that occurs on many samplers.
In addition to its usefulness in music projects, SoundReplacer is also an extremely powerful tool for sound
design and post production. Morphing gun shots, changing door slams, or adding a Doppler effect can now
be accomplished in seconds rather than minutes—with sample-level precision.
Replacement audio events can be written to a new audio track, or mixed and re-written to the source audio
track. Sample thresholds can be amplitude-switched between the replacement samples, or amplitude
crossfaded for seamless transitions.
SoundReplacer Controls
SoundReplacer Waveform Display
The waveform display shows the audio that you have selected for replacement. When you select audio on
the source track, then open SoundReplacer, the audio waveform will automatically be displayed here.
Waveform display with trigger markers shown
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Once the audio selection is displayed, you can load 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, SoundReplacer automatically updates the waveform
display each time you make a new selection or begin playback.
SoundReplacerTrigger Threshold
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:
l The yellow slider represents amplitude zone 1, the lowest-level trigger.
l The red slider represents amplitude zone 2, the middle-level trigger.
l The blue slider represents amplitude zone 3, the highest-level trigger.
With a replacement sample loaded, drag the Threshold slider to set the 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.
c If you zoom the waveform display below a specific Trigger Threshold slider’s amplitude zone, the slider
will be temporarily unavailable. To access the slider again, zoom back out to an appropriate
magnification level.
SoundReplacer Load/Unload Sound Buttons
Load/Unload Sound
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.
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g SoundReplacer lets you choose whether or not to use sample rate conversion before loading
replacement samples if they are at a different sample rate from the session.
To audition a replacement sample before loading it into SoundReplacer, use the Import Audio command in
Pro Tools. Once you have located and previewed an audio file, you can then load it into SoundReplacer
using the Load/Unload Sound icons.
c SoundReplacer does not load clips that are part of larger audio files. To use a clip as a replacement
sample, you must first save it as an individual audio file.
SoundReplacer Zoomer
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.
l To zoom in on amplitude, click the Up Arrow.
l To zoom out on amplitude, click the Down Arrow.
l To zoom in on time, click the Right Arrow.
l To zoom out on time, click the Left Arrow.
c 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.
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.
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.
n For more information on using Peak Align, see "Getting Optimum Results with SoundReplacer" on
page 444.
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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.
g Setting Mix to 50% and clicking Preview lets you audition source audio and replacement audio
together to check the accuracy of replacement triggering timing.
SoundReplacer Dynamics
Dynamics controls how closely the audio events in the replacement file track the dynamics of the source
file:
l Setting the ratio to 1.00 matches the dynamics of the source file.
l 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.
l 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

To use online help, hold the pointer over any control and an explanation will appear.
Using SoundReplacer
Following are basic guidelines for using SoundReplacer effectively. Also see "Getting Optimum Results with
SoundReplacer" on the next page.
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 an 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.
g 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, "Mapping The Same Sample
Into Multiple Amplitude Zones with SoundReplacer" on page 445
7. To align the amplitude peak in the replacement file(s) to threshold trigger markers in the source
audio, enable Peak Align.
8. Click Preview to audition the replacement audio.
9. Adjust the Threshold sliders to fine tune audio replacement triggering.
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10. Adjust the Dynamics slider to fine tune how SoundReplacer tracks and matches changes in the
source audio’s dynamics.
11. Adjust the Mix slider to set the balance between replacement audio and source audio.
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. Render the selected clip by doing one of the following:
t To render the selected clip only in the track in which it appears, choose Playlist from the
Selection Reference pop-up.
t To render the selected clip in the Audio Clip List only, choose Clip List from the Selection
Reference pop-up.
14. Determine which occurrences of the selected clip you want to render by doing one of the following:
t 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).
t If you do not want to update every occurrence of the selected clip, disable Use In
Playlist.
15. If you have selected multiple clips for rendering and want to create a new file that connects and
consolidates all of these clips together, choose Create Continuous File from the File mode pop-
up menu.
c 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

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

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:
l Turn on Peak Align if you are replacing drum or percussion sounds whose peak level occurs at the
initial attack.
l 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.
To illustrate why Peak Align makes a difference, look at the following two illustrations.
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A fast-peaking kick drum
A slower-peaking kick drum
The first figure shows a fast-peaking kick drum whose peak level occurs at its initial attack.
The second figure shows a slower-peaking kick drum whose peak level occurs after its initial attack.
If you turn on Peak Align and attempt to replace the fast-peaking kick with the slow-peaking kick (or vice-
versa), SoundReplacer will align their peaks—which occur at different points in the sound. The audible result
would be that the replacement audio file (slow-peaking kick) would trigger too early.
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:
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 kick drum with a pre-hit preceding a denser attack
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.
445
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.
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.
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.
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 color-coded trigger markers) and then Command-drag the Threshold slider for
more precise adjustment.
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.
g 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 you often use the same settings and replacement sounds in different sessions, Sound-Replacer 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).
If it is not there, SoundReplacer looks in a folder named Audio Files within SoundReplacer’s Root Plug-In
Settings folder (Plug-In Settings/Sound-Replacer/Audio Files).
If SoundReplacer finds the replacement audio file there, the Settings file will load with the associated audio.
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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.
c Do not create subfolders within SoundReplacer’s Audio Files folder. Files located within subfolders are
not recognized.
Time Compression / Expansion

Time Compression / Expansion is an AudioSuite-only time-processing plug-in. The Time Compression /
Expansion plug-in adjusts the duration of selected clips, increasing or decreasing their length without
changing pitch.
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.
g Normalizing a selection before using Time Compression / Expansion may produce better results.
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.
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Ratio
Sets the destination length in relation to the source length. Moving the slider to the right increases the
length of the destination file, while moving the slider to the left decreases its length.
Crossfade
Adjusts the crossfade length in 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.)
Crossfade length affects the amount of smoothing performed on audio material. This prevents audio
artifacts such as clicks from occurring. Long crossfade times may over-smooth a signal and its transients.
This may not be desirable on drums and other material with sharp transients.
Use the Crossfade slider to manually adjust and optimize crossfade times if necessary. For audio material
with sharper attack transients, use smaller crossfade times. For audio material with softer attack transients,
use longer crossfade times.
Min Pitch
Sets the minimum, or lowest, pitch that will be used in the plug-in’s calculations during the Time
Compression/Expansion process. The control has a range of 40 Hz to 1000 Hz.
This control should be set lower when processing bass guitar or audio material with a low frequency range.
Set this control higher when processing higher frequency range audio material.
Accuracy
Prioritizes the processing resources allocated to audio quality (Sound) or timing (Rhythm). Moving the slider
towards “Sound” generally results in better sonic quality and fewer audio artifacts. Moving the slider
towards “Rhythm” puts the emphasis on keeping the tempo consistent.
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.
Trim
Trim is a mono (or multi-mono) plug-in that is available in DSP and Native formats.
Trim supports 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz sample rates.
Use Trim 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.
448
g Alt-click (Windows) or Option-click (Mac) the Trim selector to open a plug-in window for each channel
of a multi-channel 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.
c 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.
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14 Other AudioSuite Plug-In Utilities
AudioSuite plug-ins do not process audio in real time. Instead, they render audio according to the settings
you configure in each plug-in, replacing the original audio or creating a new variation of it.
The following AudioSuite-only utility plug-ins are installed when you install Pro Tools:
l "DC Offset Removal" below
l "Duplicate" below
l "Gain" on the next page
l "Invert" on page 452
l "Normalize" on page 452
l "Reverse" on page 453
DC Offset Removal
The DC Offset Removal plug-in removes DC offset from audio files. DC offset is a type of audio artifact
(typically caused by miscalibrated analog-to-digital converters) that can cause pops and clicks in edited
material.
To check for DC offset, find a silent section in the audio material. If DC offset is present, a near-vertical
fade-in with a constant or steady-state offset from zero will appear in the waveform. Use the DC Offset
Removal plug-in to remove it.
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.
Duplicate
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.
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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.
The Duplicate plug-in works nondestructively. You cannot choose to overwrite files.
To duplicate an audio selection:
1. Select the audio you want to duplicate.

2. Choose AudioSuite > Other > Duplicate.
4. Click Render.
Gain
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.
Gain
Specifies the gain amount. Set this value by manually adjusting the Gain slider, by entering a numeric
decibel value, or by entering a percentage.
Analyze
When clicked, displays the peak amplitude value of the current selection.
RMS/Peak Toggle
Switches the calibration of gain adjustment between Peak or RMS modes. Peak mode adjusts the gain of
the input signal to the maximum possible level without clipping. RMS mode adjusts the input signal to a level
consistent with the RMS (Root-Mean-Square) value, or the effective average level of the selected clip.
To change the gain of an audio clip:
1. Select the clip whose gain you want to change.

2. Choose AudioSuite > Other > Gain.
3. Adjust the Gain slider.
4. Click Preview to audition your changes.
6. Click Render.
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Invert
The Invert plug-in reverses the polarity of selected audio. Positive sample amplitude values are made
negative, and all negative amplitudes are made positive.
This process is useful for altering the phase or polarity relationship of tracks. The Invert 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.
To invert the phase an audio clip (or selection):
1. Select the clip whose phase you want to invert.

2. Choose AudioSuite > Other > Invert.
4. Click Render.
Normalize
The Normalize plug-in optimizes the volume level of an audio selection. Use it on material recorded with too
little amplitude, or on material whose volume levels are inconsistent (as in a poorly recorded narration).
Unlike compression and limiting, which modify the dynamics of audio material, normalization preserves
dynamics by uniformly increasing (or decreasing) amplitude.
g To prevent clipping during sample rate conversion, Normalize in Peak mode 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). Observe caution when
normalizing in RMS mode, as that mode ofanalysis does not account for instantaneous peaks.
Level
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.
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Channel Mode
When processing a selection that spans across multiple channels or tracks, the Normalize plug-in has two
modes of operation:
Mono
Normalizes each channel independently.
Multi-Input
Normalizes program material across all selected channels together, so that the channels are processed
relative to each other.
Peak/RMS Toggle
Switches the calibration of normalizing between Peak or RMS modes:
Peak
Normalizes the input signal at the maximum possible level without clipping.
RMS
Normalizes the input signal at a level consistent with the RMS (Root-Mean-Square) value, or the effective
average level of the selected clip.
Normalizing Multiple Clips
When multiple clips are selected, the Normalize plug-in can search for peaks in two different modes:
l Clip by Clip
Searches for the peak level on a clip-by-clip basis.
l Entire Selection
Searches for the peak level of all clips in the selection.
To normalize an audio clip (or selection):
1. Select the audio you want to normalize.

2. Choose AudioSuite > Other > Normalize.
3. Adjust the Level slider.
5. Click Render.”
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.
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To reverse an audio clip (or selection):
1. Select the clip you want to reverse.

2. Choose AudioSuite > Other > Reverse.
4. Click Render.
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455
Time Compression/Expansion 447
Index
Aural Exciter plug-in 286
# Drive meter 287

1-2 Band EQ (Focusrite D2) 50 gain structure 291
1-Band EQ (EQ III) 32 Harmonics control 290
4-Band EQ (Focusrite D2) 50 Level control 288
6-Band EQ (Focusrite D2) 50 Mix control 290
7-Band EQ (EQ III) 32 Null Fill control 289
304C Compressor plug-in 58 optimizing 291
compound release circuit 60 Out meter 287
deadness (avoiding) 60 Peaking control 288
overshoot 60 SPR switch 292
304E Equalizer plug-in 31 Timbre control 290
660 (Fairchild Limiter) 103 Tune control 288
670 (Fairchild Limiter) 104 Tune fader 292
A AutoPan plug-in
AAX (Avid Audio Extension) plug-ins 17 Angle slider 376
Audiomodern 395 Attack slider 382
AudioSuite Beat Clock trigger 380
Input Mode selector 182 Duration selector 381
Preview 182 ENV 377
processing preferences 182 Envelope section 381
AudioSuite plug-ins Envelope trigger 380
DC Offset Removal 450 LFO 377-378
Duplicate 450 LFO triggers 379
Gain 451 Link To Tempo option 380
Invert 452 Manual slider 376
Normalize 452 Manual trigger 380
Pitch Shift Legacy 174 Output meters 375
Reverse 453 Output slider 376
SoundReplacer 439 Panner section 376
456
Panning display 378 Drive meter 294
panning examples 382 gain structure 296
Panning Field indicator 378 In/Out switch 295
Path selectors 378 Level control 294
Place slider 377 Link switch 295
Rate slider 379 Mix control 295
Release slider 382 optimizing 296
Side-Chain Input option 383 Out meter 294
Side-Chain Input selector 382 Phase switch 295
Single trigger 380 Solo switch 295
Sound Location indicator 378 Tune control 294
Spread slider 377 C
surround panning 383 Captain Chords Lite 411
synchronizing tempo 382 Channel Strip plug-in 67
Tempo controls 380 bypassing effects modules 72
Tempo display 381 Compressor/Limiter 74, 76
Threshold slider 382 Dynamics Graph display 73
Waveform selector 379 Dynamics section 72
Width slider 376 enabling (or disabling) effects 68
Avid Link 22 Expander/Gate 75
B FX Chain 72
BF-2A plug-in 61 Gain Reduction meters 70
side-chain processing 62 Input meters 70
BF-3A plug-in 63 Input Trim 69
side-chain processing 64 Listen button 68
BF76 plug-in 65 Phase Invert 70
side-chain processing 66 Side Chain
Big Bottom Pro plug-in 293 Detection options 78
AutoTrace switch 295 Filter Frequency control 78
Compression meter 294 Filter Type options 78
Drive control 294
457
Side Chain Processing Graph Dynamics III 84
display 79
Fairchild 660 103
Source selector 78
Fairchild 670 104
Click plug-in 386
Focusrite D3 106
Compressor/Limiter (Dynamics III) 87
Impact 112
D
Maxim 115
D 51
Pro Compressor 120
D3 106
Pro Expander 132
DC Offset Removal plug-in 450
Pro Limiter 143
Dither plug-in 372-373
Pro Multiband Dynamics 152
bit resolution 372
Purple Audio MC77 164
Noise Shaping 373
Smack! 165
Down Mixer plug-in 383
Dynamics III plug-ins 84
Duplicate plug-in 450
common controls 84
flattening a track 450
Compressor/Limiter 87
D-Verb plug-in 193
De-Esser III 93
Algorithm control 194
Expander/Gate 91
Church algorithm 194
side-chain processing 96
clipping indicator 193
E
Diffusion control 194-195
effects models
Hall algorithm 194
distortion
Hi Frequency Cut control 195
DC Distortion 350
input level meters 193
Green JRC OD 350
Low-Pass Filter control 195
Tri Fuzz 351
output level meters 193
White Boost 352
Size control 194
EQ & compressor effects
dynamics
Graphic EQ 49
BF-2A 61
Gray Comp 102
BF-3A 63
modulation
BF76 64
C1 Chor/Vib 261
Channel Strip 67
Flanger 262
458
Orange Phaser 263 single coil 328
Roto Speaker 264 Eleven plug-in 296
Vibe Phaser 264 advanced applications 321, 347
reverb Amp Bypass 308, 335
Eleven SR 196 Amp Type 307, 333
Eleven effects plug-ins navigating 303, 330
Black Op Distortion 349 amps 310, 338
Black Spring reverb 195 controls 307, 334
Black/Shiny Wah 260 list of 307, 333
C1 Chorus/Vibrato 261 AudioSuite 321, 347
DC Distortion 350 Axis (On/Off) 312, 339
Flanger 262 beat clock (see Tempo Sync) 309, 336
Graphic EQ 49 blending
Gray Compressor 102 amps and cabinets 316, 343
Green JRC Overdrive 350 and cab resonance 317, 344
MIDI control 29 and phase 320, 346
Orange Phaser 263 buffer 300
Roto Speaker 264 bypass
Studio Reverb 196 amp only 308, 335
Tri-Knob Fuzz 351 cabinet and mic 311, 338
Vibe Phaser 264 Cab Type 309, 336
White Boost 352 navigating 303, 330
Eleven Lite plug-in 296 Cabinet bypass 311, 338
Eleven Mark II plug-in cabinets 309-310, 337-338
buffer 326 Category 302-303, 329
impedance 326 close mic 312, 339
input comb filtering 320, 346
about guitar amps and levels 324 combining 316, 343
Input LED 332 combo amps 310, 337
LED colors for input calibration 328 condenser 311, 338
Link button 333 cone breakup 311, 338
459
control surfaces and unused Mic Type 311, 338
controls 303, 330
navigating 303, 330
controls 303, 329
mics (microphones) 312, 339
CPU Usage 311
on- and off-axis 312, 339
Depth 309, 336
MIDI 304, 330
DSP 321, 347
Learn 30, 304, 331
dynamic 311, 338
tempo sync 309, 336
flip phase 320, 346
mono / multi-mono 302, 329
format (mono or multi-mono) 302, 329
multichannel 302, 329
Gain 1 308, 335
multiple cabinets 316, 343
Gain 2 308, 335
noise gate 306, 333
gate 306, 333
Output 306, 332
Hardware Buffer for Input
Calibration 300, 326 phase 320, 346
Harmonic 302, 329 polarity (see phase) 320, 346
humbucker 301, 328 Presence 309, 336
impedance 299 presets 305, 332
input Previous/Next arrows 303, 330
about guitar amps and levels 298 pure excess 316, 343
Input (Trim) 305, 332 recording 313, 340
input calibration 301, 328 dry 312, 340
HW Buffer 300, 326 dry and Eleven simultaneously 314,

341
Input LED 305
processing pre-recorded tracks 315,
inserting Eleven 302, 329 342
intensity (Tremolo) 309, 336 Release 306, 333
lamp (bypass) 308, 335 resonance 317, 344
LED colors for input calibration 301 resources 321, 347
load 317, 344 ribbon 311, 338
Manufacturer 302-303, 329 sample rates 302, 329
Master 309, 336 Settings Files 305, 332
Master section 305, 332 signal flow 322, 348
mic placement 312, 339 signal routing and track setups 312, 339
460
single coil 301 670 104
Speaker Breakup 311, 338 Focusrite D2 plug-in 50
Speed 309, 336 bypassing 52
stacking 316, 343 configurations 50
standby 308, 335 curves 51
stereo 302, 329 disabling 52
tap 322, 348 enabling 52
Tempo Sync 309, 336 filter controls 52
Threshold 306, 333 High-Mid Peak Filter 52
tips 321, 347 High-Pass Filter 52
tracks processed through Eleven 313, High-Shelf Filter 52

340
Input Level parameter 51
Tremolo 309, 336
Left Channel button 53
EON-Arp 412
Link button 53
EQ
Low-Mid Peak Filter 52
Channel Strip 67
Low-Pass Filter 53
EQ III 32
Low-Shelf Filter 52
Focusrite D2 50
meters 51
Pultec EQH-2 56
Output Level parameter 51
Pultec EQP-1A 55
Right Channel button 53
Pultec MEQ-5 56
Focusrite D3 plug-in 106
EQ III plug-in 32
Attack 109
1-Band EQ 32
Auto Release button 110
7-Band EQ 32
brick wall limiter 106
Frequency Graph display 33, 79
Compressor 106
gain (inverting) 33
Compressor/Limiter 106
Expander/Gate (Dynamics III) 91
Compressor+Limiter 106
external side-chain processing 111
controls 107, 110
F
External Key control 108, 111
Fairchild plug-ins
Gain Reduction meter 108
660 103
Limit LED 110
461
Limiter 106 InTune plug-in 424
meters 108 Automatic mode 425
Ratio control 109 Chromatic mode 428
RMS detector 107 creating presets 428
side-chain processing 111 Display Flat Semitones option 429
Threshold parameter 109 factory presets 428
G information display 427

Gain plug-in 451 Meter selector 426
Groove Shaper Lite 416 Needle meter 426
guitar amp simulators 296, 370 note entry fields 429
H Note Selection buttons 427
Human Lite 411 Octave range 427
I reference frequency 427
Impact plug-in 112
Single Octave mode 429
Attack 112
Strobe display 427
clip indicator 113
test tones 426
compression ratio 113
Tone Volume slider 427
External On/Off 113
tuner 424
gain (make-up) 113
Tuner Programming screen 426, 428
Gain Reduction meter 113
Invert plug-in 452
Input/Output Meter 113
K
Key Listen On/Off 113 Key Listen parameter 108, 111
Make-Up 113 L
meters 113 limiter
Ratio 112 Dynamics III Compressor/Limiter 87
Release 112 Fairchild 660 103
side-chain processing 113-114 Fairchild 670 104
Threshold 112 Focusrite D3 106
impulse response (IR) 228 Lo-Fi plug-in 352
installing plug-ins 24 adaptive quantization 354
internal side-chain processing 111 aliasing artifacts 353
462
Anti-Alias Filter control 353 Histogram 115-116
Distortion/Saturation controls 354 Input Level control 116
down-processing audio 353 Limiting 116
Linear Quantization control 354 Link button 118
Noise Generator 354 Mix control 118
Output Meter 354 Noise Shaping control 118
Quantization control 354 Online Help 115
Sample Rate control 353 Output control 117
Sample Size control 354 peak levels 115
M Peak limiting 115

MasterMeter plug-in 430 quantization noise 118
Clear button 437 Release control 117
Clip field 437 signal delay 116
Export button 437 signal peaks 115
historical metering 435 Threshold control 117, 119
Offset field 437 X axis of histogram 117
Oversampled Clip Events browser 436 Y axis of histogram 117
Oversampled Level meter 437 MIDI plug-ins 390
real-time metering 435 Mixed in Key 411
Signal Clip Events browser 436 Mod Delay III plug-in 246
Signal Level meter 437 Delay Time 247
View Time menu 437 Feedback setting 247
Maxim plug-in 115 Groove control 248
Attenuation control 117 Input meters 246
Ceiling control 117 Link button 247
controls 116 Low Pass Filter 247
Dither control 118 Meter setting 247
drum limiting 116 Mix control 248
dynamic range of a mix 116 Output Gain control 249
dynamic range of individual Output meters 248
instruments 116
Phase Invert 247
463
Sync option 247 sinusoidal 271
Tempo control 248 sum and difference frequencies 271
Modalics 412 tremolo 271
Moogerfooger 12-Stage Phaser plug-in 268 vibrato 271
allpass 268 Multiband Splitter plug-in 162
analog filters 268 N
modulation 268 Normalize plug-in 452
resonant filters 268 Note Stack 391
tremolo 268 P
whooshing 269 phase 320, 346
Moogerfooger Analog Delay plug-in 249 Pitch II
Bucket Brigade Analog Delay Chips 249 Input Polarity 172
echo (analog delay) 249 Pitch II plug-in 171
lo-fi 251 Effects controls
sound generator 251 Delay 174
Moogerfooger Lowpass Filter plug-in 265 Feedback 174
auto wah 268 Low Pass Filter 174
envelope follower 266 Mix 174
filters 267 Input and Transient controls
resonance filter 265 Clip indicator 172
whistling 267 Follow 173
Moogerfooger plug-ins Input 172
Analog Delay 249 Level indicator 172
Moogerfooger 12-Stage Phaser 268 Range 172
Moogerfooger Lowpass Filter 265 Threshold 172
Moogerfooger Ring Modulator 271 Window 172
Moogerfooger Ring Modulator plug-in 271 Pitch Shift controls
Carrier Oscillator 272 Coarse 173
LFO 271 Fine 173
metallic 273 keyboard 173
oscillator 271 Link 173
464
Ratio 173 Dynamics Graph display 135
Pitch transposition 171 Hold control 139
Pitch Innovations 416 Hysteresis control 139
Pitch Shift Legacy plug-in 174 Input Level 134
POW-r Dither plug-in 373 Input meters 134
bit resolution for Dither plug-in 373 Knee control 138
Noise Shaping 373 Ratio control 138
Pro Compressor plug-in 120 Release control 138
Attack control 127 Side Chain
Attenuation meters 122 Detection options 137
Depth control 128 Filter Type options 142
Dry Mix control 128 Side Chain Processing Graph

display 142
Dynamics Graph display 122
Source selector 140
Input Level 121
Threshold control 138
Input meters 121
Pro Limiter plug-in 143
Knee control 127
Ceiling control 146
Makeup control 128
Character control 147
Ratio control 126
Input Level 144
Release control 127
Input meters 145
Side Chain
Release control 147
Detection options 125
Filter Type options 131
Pro Multiband Dynamics plug-in 152
Side Chain Processing Graph
display 132 Attack control 160
Source selector 129 Automatic Gain control 158
Threshold control 126 band controls 157
Pro Expander plug-in 132 band meters 159
Attack control 138 bypassing bands 158
Attenuation meters 134 Crossover Frequency controls 154
Depth control 139 Depth control 160
Dry Mix control 139 detection options 161
465
disabling bands 158 Pultec MEQ-5 plug-in 56
FFT display 153 pumping 104
FFT display options 154 Purple Audio MC77 plug-in 164
Flip mode 158 R
Gain controls 155, 160 Recti-Fi plug-in 365
linking bands 158 Alternating Rectification 366
meters 156 Alt-Max Rectification 367
Release control 160 Mix control 367
Side Chain 161 negative excursion period of

waveform 367
Slope control 160
Negative Rectification 366
soloing bands 158
Output Meter 367
source linking 156
Positive Excursion 367
Positive Rectification 366
Pro Subharmonic plug-in 354
Post-Filter 367
Drive control 361
Pre-Filter 365
filters 358
Rectification 366
frequency display 357
Subharmonic synthesis 367
frequency range controls 359
zero crossing 367
Gain controls 360
Reel Tape Delay plug-in 251
input level 356
Bass control 253
input meters 356
Feedback control 253
MIDI control 365
Mix control 253
Mix controls 361
Noise parameter 254
output level 357
presets 254
output meters 357
Speed control 253
Surround Send controls 363
synchronizing to session tempo 254
Pultec EQH-2 plug-in 56
Treble control 253
Pultec EQP-1A plug-in 55
Wow Speed parameter 253
phase (EQ filters) 57
Wow/Flutter control 253
program equalization 55
smooth EQ 55
466
Reel Tape Flanger plug-in 273 anechoic chamber 198
Feedback control 275 Attack control 200
Invert parameter 276 Band Breakpoints 203
LFO Depth control 276 Chorus controls 199
LFO Rate control 275 clipping indicator 205
Mix control 276 Crossover sliders 204
Noise parameter 276 Decay Ratio control 199
presets 277 Delay Master control 202
Range control 274 Depth control 199
synchronizing to session tempo 276 Diffusion control 200
Wow/Flutter control 275 Dynamics controls 199
Reel Tape plug-ins 273 early reflections 198
Drive control 252, 274, 368 Early Reflect On 202
Output control 252, 274, 368 Early Reflection control 201
Tape Formula control 252, 274, 368 ER (early reflection) button 204
Tape Machine control 252, 274, 368 ER Settings control 201
Reel Tape Saturation plug-in 367 presets 201
Bias control 369 simulating 201
Cal Adjust control 369 Frequency control 203
Noise control 369 HF Cut control 203
presets 370 HF Damp control 204
removing plug-ins 25 input level meters 205
reverb late reverberation 198
D-Verb 193 Level control 200, 202
Reverb One 196 Master Mix controls 198
ReVibe II 205 meters
Space 221 clipping indicator 205
reverb (about) 223-224 input 205
Reverb One plug-in 196 output 205
100% Wet control 199 Output Level meters 205
acoustic environments 198
467
Pre-delay 198 Contour display 215
presets 200 Depth control 212
Rate control 200 Diffusion control 209
RC (reverb contour) button 204 Early Reflection Level control 210
reverb character 198 Early Reflection Pre-Delay control 210
reverb graphs Early Reflection section 210
editing 202 Early Reflection Spread control 210
Reverb Color 203 Early Reflections button 215
Reverb Contour 204 Early Reflections On button 211
Reverb EQ 203 EQ graph display 213
simulating early reflections 201 Front button 215
Size control 200 Front control 211
Spread control 200, 202 High Frequency Color control 211
Stereo Width control 198 High Frequency control 214
Threshold control 199 High Frequency Crossover 214
Time control 200 High Frequency Ratio 214
tool tips 204 High Frequency Rear Cut control 214
Wet/Dry control 198 High Gain control 214
Reverse plug-in 453 Input control 211
ReVibe II plug-in 205 Input/Output meter 207
100% Dry Mix button 213 Level control 209
100% Wet Mix button 213 Levels section 211
Attack Shape control 210 Low Frequency Color control 211
Attack Time control 209 Low Frequency control 213
bipolar controls 26 Low Frequency Crossover 214
Center control 212 Low Frequency Ratio 214
Chorus On button 212 Low Gain control 213
Chorus section 212 Mix section 212
clip indicator 207 Next and Previous Room Type

buttons 208
Color graph display 214
Pre-Delay control (reverb tail) 209
Coloration control 211
468
Pre-Delay Link button 211 S
Rate control 212 SansAmp PSA-1 plug-in 370
Rear button 215 amp simulation 370
Rear ER control 212 buzz 371
Rear Level Link button 212 cabinet simulation 370
Rear Reverb control 212 crunch 371
Rear Shape control 210 distortion 371
Reverb Contour button 215 equalization 370
Reverb section 208 harmonic generation 370
reverb tail controls 208 lo-fi textures 370
Reverb tail type 208 punch 371
Reverb Type 207 tube sounds 370
Reverb Type menu 208 unity 371
Room Coloration section 211 wah 371
Room Type 207 Sci-Fi plug-in 278
Room Type Category menu 208 Effect Amount control 279
Room Type Name menu 208 Effect Frequency control 279
Size control 209 Envelope Follower 279
Spread control 209 Freak Mod 278
Stereo Width control 213 Input Level 278
supported channel formats 205 LFO (Low Frequency Oscillator) 279
Time control 209 Mod Amount/Mod Rate control 280
Wet/Dry control 213 Mod Slewing control 279
ReWire plug-in Modulation Type control 279
and voices 388 Output Meter 280
client applications 388 Resonator– 279
requirements 387 Resonator+ 279
signal flow for audio and MIDI 389 Ring Mod control 278
software synthesizer 388 ring modulation 278
track count 388 Sample+Hold control 279
Riffer 395 Slewing 279
469
triangle wave 279 Release control 167
Trigger and Hold 279 Side-Chain EQ control 167
side-chain processing 64, 66, 77, 96, 111, side-chain frequency filters 167
114, 129, 139
side-chain processing 170
Signal Generator plug-in 438
threshold and ratio 167
Frequency control 438
unity gain 166
Level control 438
VU meter 169
pink noise 438
SoundReplacer plug-in 439
Signal control 438
Crossfade control 442
white noise 438
Dynamics controls 443
Smack! plug-in 165
Load/Unload Sound icons 441
adjusting input 166
MIDI-triggered samplers 440
Attack control 166
Mix control 443
band emphasis 168
Online Help 443
clip indicator 169-170
Peak Align control 442
compression modes 166
Trigger envelope 441
Distortion control 169
Trigger markers 441
hard limiting ratio setting 167
Trigger Threshold 441
high-pass detector 168
waveform display 440
high-pass filter 169
Zoomer 441-442
Input meter 170
Space plug-in
key input 170
presets 244
level 165
Space plug-in 221
limiting 167
clip indicator 236
Meter Mode button 169
Control group selector 239
Norm mode 166
convolution reverb 224
Opto mode 166
Decay controls 243
output gain 167
Delay controls 240
Output meter 170
Early Reflection controls 241
Ratio control 167
EQ controls 242
reducing waveform distortion 166
impulse computer 226
470
impulse response (IR) 228 Minimum Pitch control 448
installing IR packages 238 Ratio control 448
IR browser 236-237 Source and Destination controls 447
IR channel compatibility 230 Time Shift plug-in 176
IR channel formats 228 Audio Gain control 178
IR filename conventions 229 Audio Mode pop-up menu 178
IR formats 228 Audio Range pop-up menu 178
IR library 245 Audio section 177
IR Name 232 changing pitch 183
Level controls 239 changing time 183
Meters display 235 changing time and pitch 183
Online IR Library 238 Clip indicator 178
Picture Preview mode 233 Decay Rate 181
Preferences mode 235 displays and controls 177
presets 230 Follow button 181
primary controls 239 Formant section 179
Quick browser 233 Input modes 182
Snapshot menu 232 Level indicator 179
Snapshot mode 234 Monophonic mode 178
snapshots 231, 244 Original time 179
system design 225 Pitch section 181
Waveform mode 233 Pitch Shift 182
T Polyphonic mode 178

TCE Trim tool 182 post production workflow 184
Tel-Ray Variable Delay plug-in 255 Processed time 179
Electronic Memory Unit 255 pull up/pull down TCE percentages 184
tuna can 255 Rhythmic mode 178
Time Compression/Expansion plug-in 447 Speed 179
Accuracy control 448 Tempo displays 179
Crossfade control 448 Threshold control 181
destination fields 447 Time section 179
471
timebase 179 Clip indicator 187
Transient section 180 Formant Shift control 190
Transpose 182 Gain control 187
Unit pop-up menu 179 Level indicator 187
Varispeed mode 178 Monophonic mode 187
Window 181 Original time 188
TimeAdjuster plug-in 257 Pitch section 186, 190
DSP delay compensation 258-259 Pitch Shift control 190
phase cancellation 258 pitch shifting a selection 192
Phase Invert button 257 Poly (Faster) mode 187
Trim plug-in 448 Polyphonic mode 187
V post-production workflow 192

variable-mu 103 Processed time 188
Vari-Fi plug-in 184 processing audio 191
Fades controls 185 pull up/pull down TCE percentages 192
Selection controls 185 Sensitivity control 189
Slow Down control 184 TCE Trim tool 191
Speed Up control 184 Tempo (original and processed) 188
vinyl mastering 105 Time section 186-187
Voce plug-ins Time Shift control 188
Voce Chorus/Vibrato 280-281 time shifting a selection 192
Voce Spin 281 Transient section 186, 189
Rotor Balance 283 Transpose control 190
X Unit timebase selector 188
X-Form plug-in 185 Window control 189
2x, 4x, and 8x Range buttons 188 Z
Audio section 186 z (Eleven input impedance) 299, 326
Audio Type pop-up menu 187
AudioSuite Input 191
AudioSuite Preview 191
AudioSuite processing preferences 191
472

Audio Plug-Ins Guide

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Audio and MIDI

For a current and complete list of Avid trademarks visit: www.avid.com/legal/trademarks-and-other-notices.

Avid Audio Plug-Ins 17

Using Plug-Ins in Pro Tools 20

System Requirements and Compatibility for Plug-Ins 20

Conventions Used in This Guide 21

Installing and Authorizing Avid Plug-Ins 23

Installing Plug-ins Bundled with Pro Tools Subscriptions 23

Installing Plug-Ins from the Avid Marketplace 23

Installing Plug-Ins for Pro Tools and Media Composer 23

Authorizing Avid Audio Plug-Ins 24

Adjusting Plug-In Controls 25

Dragging Plug-In Controls 25

Editing Control Values 27

Dragging in Graphic Displays 27

Adjusting Controls with Fine Resolution 28

Resetting Controls to Default Values 28

General Effects Controls 28

304E Equalizer Controls 31

Adjusting EQ III Controls 33

EQ III I/O Controls 35

EQ III EQ Band Controls 36

Eleven Effects Graphic EQ 49

Pultec Tips and Tricks 57

304C Compressor Controls 59

304C Compressor Tips and Tricks 60

Using the BF-2A Side-Chain Filter 62

BF-2A Tips and Tricks 63

BF-3A Tips and Tricks 64

BF76 Tips and Tricks 66

Channel Strip Sections and Panes 67

Channel Strip Input Section 69

Channel Strip Output Section 71

Channel Strip FX Chain 72

Channel Strip Dynamics Section 72

Channel Strip EQ / Filters Section 79

Dynamics III Common Controls 84

Compressor / Limiter III 87

Expander / Gate III 91

Expander / Gate III Threshold Control 92

Dynamics III Side-Chain Input 96

Eleven Effects Gray Compressor 102

Gray Compressor 102

Fairchild Plug-Ins 103

Fairchild 660 103

Fairchild 670 104

D3 Side-Chain Processing 107

Using D3 in Stereo 107

D3 Common Controls 107

D3 Compressor Controls 109

D3 Limiter Controls 110

Using the Side-Chain Input in D3 111

Impact Controls 112

Using the Impact Compressor 113

How Maxim Differs From Conventional Limiters 116

Maxim Controls and Meters 116

Using Maxim 118

Maxim and Mastering 119

Pro Compressor 120

Pro Compressor Metering 120