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DigidesignPlug-Ins Guide

for Macintosh and Windows

Digidesign Inc.3401-A Hillview Avenue

Palo Alto, CA 94304 USAtel: 650·842·7900fax: 650·842·7999

Technical Support (USA)650·842·6699650·856·4275

Product Information650·842·6602800·333·2137

Fax on Demand1-888-USE-DIGI (873-3444)

World Wide Webwww.digidesign.com

Digidesign FTP Siteftp.digidesign.com

Copyright

This User’s Guide is copyrighted ©1999 by Digidesign, a division of Avid Technology, Inc. (hereafter “Digidesign”), with all rights reserved. Under copyright laws, this manual may not be duplicated in whole or in part without the written consent of Digidesign.

DIGIDESIGN, AVID and PRO TOOLS are trademarks or registered trademarks of Digidesign and/or Avid Technology, Inc. All other trademarks are the property of their respective owners.

All features and specifications subject to change without notice.

PN 932707454-00 REV A 9/99

contents

Chapter 1. Getting Started With Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

Plug-In Formats. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

Installing Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

Authorizing Plug-Ins. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Deauthorizing Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

Removing Expired Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

DSP Usage of TDM Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

Allocating Additional Memory to DAE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

Chapter 2. Working with Real-Time Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Plug-Ins as Inserts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Processing Power Requirements of TDM and RTAS Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Delay in Digital Signal Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

Using the Inserts/Sends Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

Using Real-Time Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

Using a Key Input for Side Chain Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

Automating Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

Using the Librarian . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

Creating a Custom User Default Setting. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15

Creating Library Subfolders . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16

Chapter 3. Working with AudioSuite Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

The AudioSuite Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

The AudioSuite Processing Preferences Dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

Using AudioSuite Plug-Ins. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Contents iii

iv

Chapter 4. The D-Fi Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

Lo-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

Sci-Fi. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Recti-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

Vari-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

D-Fi Demo Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Sci-Fi Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Lo-Fi Examples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Recti-Fi Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Chapter 5. The D-fx Plug-Ins. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

D-fx Chorus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

D-fx Flanger . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

D-fx Multi-Tap Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

D-fx Ping-Pong Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

D-fx D-Verb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41

Selecting Tracks/Regions for D-fx Delay or Reverb Processing . . . . . . . . . . . . . . . . . . . . . . . 44

Using D-fx in Stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

Chapter 6. The D-Verb Plug-in . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

About Reverb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

D-Verb Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

Customizing a Reverb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

Using D-Verb AudioSuite . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Chapter 7. The Focusrite D2 Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51

Focusrite D2 Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51

D2 Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52

Editing D2 Parameters from the Keyboard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

Using D2 in Stereo: Left/Right/Link Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

Chapter 8. The Focusrite D3 Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

Focusrite D3 Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

D3 Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59

D3 Compressor Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

D3 Limiter Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62

Digidesign Plug-Ins Guide

Chapter 9. The DINR Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Broadband Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Broadband Noise Reduction Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

Using Broadband Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71

Using BNR AudioSuite . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75

Hum Removal. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76

Chapter 10. The DPP-1 Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

DPP-1 Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

Chapter 11. The Maxim Plug-In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85

About Peak Limiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86

Maxim Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87

Using Maxim Effectively . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90

Maxim and Mastering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90

Chapter 12. The Bruno & Reso Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93

DSP Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

Inserting Bruno/Reso onto an Audio Track. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

Playing Bruno/Reso. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95

Using a Key Input for Side Chain Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98

Bruno Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98

Reso Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104

Bruno/Reso Demo Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111

Chapter 13. The SoundReplacer Plug-In. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

Audio Replacement: Then and Now. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

SoundReplacer Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114

Using SoundReplacer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118

Getting Optimum Results With SoundReplacer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

Using the Audio Files Folder for Frequently Used Replacement Files. . . . . . . . . . . . . . . . . . . 121

SoundReplacer Demo Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121

Contents v

vi

Appendix A. DSP Requirements for TDM Plug-Ins. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125

Appendix B. DSP Delays Incurred by TDM Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . 127

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129


chapter 1

Getting Started With Plug-Ins

Digidesign Plug-Ins provide a comprehen-sive set of digital signal processing tools for professional audio production.

This Guide explains the use of each of the Plug-Ins currently available from Digide-sign on Macintosh and Windows-based Pro Tools systems.

These Plug-Ins include:

◆ Bruno™ & Reso™ cross-synthesis Plug-Ins

◆ D-Fi™ creative sound design Plug-Ins

◆ D-FX™ time-based effects processor

◆ DINR™ intelligent noise reduction

◆ DPP-1™ pitch processor

◆ D-Verb™ digital reverb

◆ Focusrite D2™ multi-band equalizer

◆ Focusrite D3™ compressor/limiter

◆ Maxim™ peak limiter/sound maximizer

◆ SoundReplacer™ drum and sound re-placement Plug-In

☞ For more information on using Plug-Ins dur-ing mixdown with Pro Tools, see the Pro Tools Reference Guide.

System RequirementsTo use Digidesign Plug-Ins you need:

◆ A Digidesign-approved Pro Tools system with version 4.0 software or higher

– or –

◆ A third-party software application that supports the Digidesign TDM, RTAS, or Au-dioSuite Plug-In standard.

For the latest compatibility information, contact your local Digidesign dealer or visit Digidesign's website.

Plug-In FormatsDSP Plug-In modules, or simply Plug-Ins, are special-purpose software add-ons which give additional functionality to Pro Tools. There are three formats of Plug-Ins.• TDM Plug-Ins (real-time)• RTAS Plug-Ins (real-time)• AudioSuite Plug-Ins (non-real-time, file-

based processing)

Chapter 1: Getting Started With Plug-Ins 1

2

TDM Plug-Ins(TDM-equipped Systems only)

Real-time TDM Plug-Ins function as track inserts, are applied to audio during play-back, and process audio non-destructively in real time. TDM Plug-Ins are designed for use on TDM-based Pro Tools systems, and rely on the processing power of Digidesign DSP cards. The number and variety of TDM Plug-Ins that you can use simultaneously in a session are limited only by the amount of DSP available. You can increase available DSP by installing additional MIX Farm or DSP Farm cards in your computer. This power-on-demand aspect is a significant advantage of TDM-based systems.

RTAS Plug-Ins (Pro Tools LE Software only)

RTAS (Real-Time AudioSuite) Plug-Ins pro-vide features and functionality similar to their TDM counterparts, but are designed for use on non-TDM-based Pro Tools sys-tems running Pro Tools LE software. Unlike TDM Plug-Ins, they rely on and are limited by the processing power of your computer. The more powerful your computer, the greater the number and variety of RTAS Plug-Ins that you can use simultaneously. Because of this dependence on the CPU or host processing, the more RTAS Plug-Ins you use concurrently in a session, the greater the impact on other aspects of system per-formance such as available track count, edit density, and automation latency.


AudioSuite Plug-Ins Non-real-time AudioSuite Plug-Ins are not used during playback, but are instead used to process audio files on disk, creating new, rewritten audio files with the effect perma-nently applied. AudioSuite Plug-Ins can be used on all Pro Tools systems.

Installing Plug-InsThe Digidesign Plug-In installer CD-ROM allows you to install and use not only the Plug-Ins which you have purchased, but also a demo version of any other Digide-sign Plug-In. Demo versions of Plug-Ins can be used for up to 15 days, after which they will expire.

Updating Older Plug-InsBecause the Plug-Ins Installer CD-ROM contains the latest versions of all Digide-sign Plug-Ins, you can use it to update any Plug-Ins you already own by installing the desired Plug-In from the CD-ROM. If there is already a valid authorization key on the destination hard disk, you will not be prompted for a new one.

Installation on Macintosh

To install a Plug-In:

1 Insert the Plug-Ins Installer CD-ROM in your CD-ROM drive and double-click the Install Plug-Ins file.

2 Select the Plug-Ins that you wish to in-stall. You can also install 15-day demo ver-sions of any of Digidesign’s other TDM or AudioSuite Plug-Ins.

3 Select a destination drive for the Plug-Ins. (This should be the drive that contains your DAE folder.)

4 Click Install. The Plug-Ins are installed in the DAE Folder, inside the System Folder.

5 If you wish to perform additional instal-lations, click Continue. If you are finished with your installation, click Quit to return to the Finder.

Installation on Windows NT

To install a Plug-In:

1 Insert the Plug-Ins Installer CD-ROM in your CD-ROM drive. The Plug-Ins Installer automatically opens. (If you have disabled the auto-launch option on your computer, navigate to the Plug-Ins Installer CD-ROM and double-click the Setup.exe file.)

2 Select the Plug-Ins that you wish to in-stall and click Next. The Plug-Ins are in-stalled in your DAE folder.

3 When installation is complete, click Fin-ish. Your computer will restart. When you open Pro Tools, you will be prompted to authorize the Plug-Ins.

Send in Your Registration Card to Receive a Backup Authorization Disk

When you send in the Registration Card for a Plug-In you have purchased, Digide-sign will mail you an additional authoriza-tion disk for that Plug-In. If you are an in-ternational customer, refer to the list of international distributors included with your registration package.

Authorizing Plug-InsThe Authorizer floppy disk included with each Plug-In you purchase allows you to authorize a single system to use that Plug-In. This floppy disk also allows you to re-claim the authorization if necessary.

▲ Do not lose your Authorizer floppy disk! You will not be able to complete an installation without it. Keep the disk in a safe place.

Authorization on Macintosh

On Macintosh systems, you can authorize Plug-Ins as they are installed.

To authorize a Plug-In during installation:

■ During installation, insert the appropri-ate Authorizer floppy disk when prompted.

For Plug-Ins that you have not purchased, you can use a demo copy for 15 days, after which the Plug-Ins will expire.

If you purchase a Plug-In after you have al-ready installed a demo version, there is no need to reinstall it. Run the Authorizer ap-plication on the Authorizer floppy disk you received when you purchased the Plug-In, and the 15-day time limit will be removed.

To authorize a Plug-In after installation:

1 Insert the Authorizer floppy disk.

2 Double-click the Deauthorizer applica-tion.

3 Select the drive you want to authorize and click Install. (This should be the drive that contains your DAE folder.)

4 When authorization is complete, click Done.


4

✽ On Macintosh systems, you can use your Authorizer floppy disk as a backup authoriza-tion key when you have used all available au-thorizations. With the Plug-In installed on the desired hard drive, simply insert the floppy disk when Pro Tools prompts you for authoriza-tion. You must do this each time you open the Pro Tools application.

Authorization on Windows NT

When you open Pro Tools for the first time after installing a Plug-In, you will be prompted to authorize the Plug-In.

To authorize a Plug-In:

1 Open Pro Tools.

2 Insert the appropriate Authorizer floppy disk when prompted, and click Next to be-gin authorization.

3 To activate a demo version of a Plug-In, click Try It. You can use the demo copy for 15 days, after which it will expire.

The Plug-In Authorization dialog


Deauthorizing Plug-InsIt is sometimes necessary to remove Plug-In authorization from a hard drive. If you plan to low-level reformat a hard drive con-taining an authorization key, you must first reclaim the authorization or it may be lost. If you want to authorize a different hard drive to use the Plug-In, you must re-claim the original authorization since au-thorization is limited to a single drive.

Macintosh

To deauthorize a Plug-In:

1 Insert the Plug-In Authorizer floppy disk into your floppy drive.

2 Double-click the DeAuthorizer applica-tion.

3 Click Setup. From the Authorization pop-up menu, select Remove, then select the appropriate hard drive. (The drive with the authorization key is indicated by a “key” icon.)

4 Click Remove.

5 When deauthorization is complete, Click Done, then Quit to return to the Finder.

Windows NT

To deauthorize a Plug-In:

1 Locate the DeAuthorizer application for the Plug-In. It is inside the Digidesign Folder on your System drive.

✽ If you cannot locate the DeAuthorizer appli-cation on your hard drive, the Plug-In Authorizer floppy disk contains a copy that you can run.

2 Double-click the DeAuthorizer applica-tion.

3 Insert the Plug-In Authorizer floppy disk.

4 Select the hard drive you want to deau-thorize. (The drive with the authorization key is indicated by a “key” icon.)

5 Click Remove.

6 When deauthorization is complete, Click Done.

Removing Expired Plug-InsIf you allow a Plug-In to expire, you should remove it from your system. Otherwise, each time you open Pro Tools you will be prompted with a message that the Plug-In has expired.

Macintosh

To remove an expired Plug-In:

1 Open the DAE Folder inside your System Folder.

2 Open the Plug-Ins folder or Plug-Ins (Un-used) folder and drag the expired Plug-In to the Trash.

3 Empty the Trash.

The Plug-In DeAuthorizer

Windows NT

To remove an expired Plug-In:

1 From the Start menu, choose Settings > Control Panels and double-click Add/Re-move Programs.

2 Select the expired Plug-In from the list of installed applications and click the Add/Re-move/button.

3 When removal is complete, click OK and close the Add/Remove Control Panel.

DSP Usage of TDM Plug-InsEach real-time TDM Plug-In in a Pro Tools session uses a portion of your system’s total available DSP resources. Since these DSP re-sources reside on the cards that make up your particular Pro Tools hardware config-uration, the amount of DSP available de-pends entirely on the number and type of DSP cards in your system.

The table in Appendix A of this Guide shows the total number of instances of each Digidesign TDM Plug-In that can be powered by a single DSP chip on the two types of Digidesign TDM-based DSP cards (the Pro Tools 24 MIX card and the DSP Farm). DSP usage differs according to card type.

✽ The Show DSP Usage window (in the Pro Tools Windows menu) allows you to see how much DSP is available on your system and how it is currently being used. For details, see the Pro Tools Reference Guide.


6

DSP Manager Allows DSP Sharing Between TDM Plug-InsDigidesign’s MultiShell and DSP Manager technologies allow up to five different types of TDM Plug-Ins to share the same DSP chip at one time. This allows you to si-multaneously use a greater variety of Plug-Ins by efficiently managing the DSP avail-able in your system.

In order to take advantage of this capabil-ity, Plug-Ins must be MultiShell compati-ble. The Focusrite D3 Plug-In features full MultiShell compatibility when used on a Pro Tools 24 MIX system.

If you are not sure whether a specific Third-Party Plug-In is MultiShell-compatible, please check with its developer.

Allocating Additional Memory to DAE(Macintosh Only)

If you plan to use a large number of Plug-Ins in addition to the DigiRack Plug-Ins in-cluded with Pro Tools, allocating addi-tional memory to DAE will help ensure re-liable system performance.

If enough RAM is available in your com-puter, allocate 1-2 megabytes of additional RAM to DAE for each non-DigiRack Plug-In installed on your system.

To allocate additional memory to DAE:

1 Start Pro Tools so that DAE can calculate its basic memory allocation.

2 Go to the Finder, and under the Apple menu, choose About This Computer.


3 If you have 3 megabytes or more of mem-ory available (as indicated in the Largest Unused Block portion of this window), go to step 4. If you have less than 3 megabytes of free memory (3,000k), stop here: Do not allocate additional memory to DAE unless you install additional RAM in your com-puter.

4 Quit Pro Tools.

5 Open the DAE folder inside your System Folder, select DAE, and choose Get Info from the Finder’s File menu.

6 Enter the desired amount of memory above the minimum requirement in the Pre-ferred Size field. For example, if the Pre-ferred Size field currently says “9402k” and you wish to allocate an additional 2 mega-bytes of memory (1 megabyte equals 1000 kilobytes), enter “11402” into the Preferred Size field.

7 Close the Get Info dialog.

8 The next time you start Pro Tools, DAE will use this new memory allocation.

Allocating additional memory to DAE

chapter 2

Working with Real-Time Plug-Ins

Real-time Plug-Ins process audio non-de-structively in real time. They do not alter the original source audio, but only apply their effect during playback.

There are two formats of real-time Plug-Ins, TDM Plug-Ins and RTAS Plug-Ins. TDM Plug-Ins are designed for use on TDM-equipped Pro Tools systems. RTAS Plug-Ins are designed for use on Pro Tools LE soft-ware.

Plug-Ins as InsertsReal-time Plug-Ins are available as track in-serts and must be inserted in-line on an au-dio track, auxiliary return, or a master fader. A maximum of 5 real-time Plug-Ins can be used per track.

When more than one insert is used on a track, they process the audio in series, each effect being added to the previous one, from top to bottom in the Mix window.

Pre-Fader Insert OperationReal-time Plug-Ins function as pre-fader in-serts, meaning that their level is not af-fected by a track’s volume fader (except when used on a master fader). For this rea-

son, clipping can occur if you boost the gain of a Plug-In to extremes. This is partic-ularly true on tracks recorded at high am-plitude. Listen carefully to your tracks and watch on-screen metering to identify and rectify clipping if it occurs.

Mono and Stereo Inserts

Most Plug-Ins can be used in either mono or stereo, and some feature mono-to-stereo capability for creating a stereo image from a mono track. Some stereo Plug-Ins require up to twice as much processing power as mono Plug-Ins. Any inserts that occur on a track after another stereo insert are auto-matically used in stereo.

Processing Power Requirements of TDM and RTAS Plug-InsTDM and RTAS Plug-Ins differ in their pro-cessing power requirements. TDM Plug-Ins use DSP power. RTAS Plug-Ins use CPU power.

Chapter 2: Working with Real-Time Plug-Ins 7

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TDM Plug-Ins and DSP PowerThe number and variety of TDM Plug-Ins you can use at one time depends on how much DSP power is available in your sys-tem. Since the TDM hardware on Pro Tools 24 and Pro Tool 24 MIX cards provide dedicated DSP power on demand for Plug-Ins, Plug-In performance isn’t lim-ited by CPU processing power.

You can add more mixing and processing power to your system by installing addi-tional DSP cards, provided you have un-used PCI expansion slots in your computer or use a Digidesign-approved Expansion Chassis.

The DSP Usage window (Windows > Show DSP Usage) shows how much DSP is avail-able in your system and how it is currently being used.

☞ For more information about DSP usage and allocation, see Appendix A of this Guide and also Appendix A of the Pro Tools Reference Guide.

RTAS Plug-Ins and CPU PowerBecause RTAS Plug-Ins rely on and are lim-ited by the processing power of your com-puter’s CPU, the more RTAS Plug-Ins you use concurrently in a session, the greater

The DSP Usage window


the impact on other aspects of your sys-tem’s performance such as maximum track count, the density of edits possible, and la-tency in automation and recording.

The CPU Load meter in the Automation Enable window (Windows > Show Automa-tion/Performance) shows how much of your computer’s processing power is cur-rently being used, and how much is still available.

Delay in Digital Signal Processing(TDM Plug-Ins only)

DSP processing in digital audio systems in-curs signal delay of varying amounts. Such delays can vary from as short as several mi-croseconds to as long as several millisec-onds, depending on the type of processing applied.

Delays are a matter of concern only if you use a real-time TDM Plug-In on one chan-nel of a stereo or multichannel signal but not the others. This can cause the channels to be slightly out of phase. If you are work-

CPU Load meter

ing with mono tracks, or are processing all channels with the same Plug-Ins, these sig-nal delays should not be a matter of con-cern.

Maxim is the only RTAS-format Plug-In which incurs signal delays of this type.

☞ See Appendix B of this Guide for informa-tion on delays inherent in specific TDM and RTAS Plug-Ins. See also Appendix A of the Pro Tools Reference Guide for a comprehensive guide to calculating DSP-induced delays.

Compensating with TimeAdjuster

You can compensate for TDM Plug-In-in-duced delays by using the TimeAdjuster TDM Plug-In included with Pro Tools. This Plug-In allows you to apply a specific num-ber of samples of delay to the signal path of a Pro Tools track. Using one of the preset li-brarian settings files for TimeAdjuster, you can apply the correct compensation time in samples for delay introduced by a spe-cific Plug-In or Plug-Ins.

☞ For details on using TimeAdjuster, see the DigiRack Plug-Ins Guide.

Using the Inserts/Sends EditorThis floating window appears whenever you click an insert or send button on a track. It allows you to access and edit the parameters for any real-time Plug-In insert or send in use on a track.

Settings menu Allows you to copy, paste, save, and import Plug-In settings.

Track Selector Allows you to access any non-MIDI track in your session.

Librarian Menu Allows you to recall settings files saved and stored in a Plug-In’s root set-tings folder or in the current session’s Plug-In Settings folder. See “Using the Librarian” on page 12.

Insert/Send Selector Allows you to access any insert or send on the current track.

Key Input Selector Allows you to select au-dio on a particular input/bus and route it to trigger the Plug-In. This menu only appears on Plug-Ins that feature side-chain process-ing.

Plug-In Selector Allows you to select any real-time Plug-In installed in the DAE Plug-Ins folder.

Compare Button If you have changed the settings of a Plug-In since you last saved them, clicking this button toggles between

The Inserts/Sends Editor (Focusrite d3 shown)

Click here to show Inserts/Sends Editor

Settings menu

Track selector

Librarian menu

Insert/Sendselector

Plug-In selector

Effect bypass

Automation enable

C
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the original settings and the edited settings in order to compare them.

Bypass Button Clicking this button dis-ables the currently displayed Plug-In. This allows you to compare the track with and without the effect. To re-enable a Plug-In, click this button again.

Enable Automation Parameters Button

Clicking this button allows you to enable individual Plug-In parameters for automa-tion recording. Creating automation is cov-ered in detail later in this chapter.

Using Real-Time Plug-InsTo use a real-time Plug-In in a Pro Tools session, add it to a track as an insert.

To insert a Plug-In on a track:

1 Click the Inserts button on a track and se-lect the Plug-In that you want to use.

2 Click Play on the Transport to begin au-dio playback.

3 Adjust the parameters of the Plug-In for the effect you want.

Adding a Plug-In as an Insert


Keyboard Shortcuts

■ For finer adjustments, hold down the Command key (Macintosh) or Control key (Windows), then drag the desired parame-ter slider.

■ To return a control to its default value, Option-click (Macintosh) or Alt-click (Win-dows) the control.

To remove a Plug-In from a track:

■ Click the Insert Selector and choose No Insert..

Keyboard Input for Plug-In ParametersYou can use your computer keyboard to in-put and edit parameters of Plug-Ins.

To set parameters with your keyboard:

◆ Click in the parameter text field that you wish to edit to activate the field. Type the desired value.

◆ In fields that support values in kilohertz, typing “k” after a number value will multi-ply the value by 1000. (If you wish to enter a value of 8000, type “8k”).

◆ To increase a value, press the Up Arrow on your keyboard. To decrease a value, press the Down Arrow on your keyboard.

Removing an Insert

◆ Press Enter on the numeric keyboard af-ter typing a value to input the value (with-out leaving the selected parameter field).

◆ Press Return (Macintosh) or Enter on the alpha keyboard (Windows) to enter the value and leave keyboard editing mode.

◆ To move downward through the differ-ent parameter fields, press the Tab key. To move upwards, press Shift-Tab.

Using a Key Input for Side Chain ProcessingCertain Plug-Ins, such as the Focusrite D3 Compressor/Limiter, feature side-chain processing capabilities. Side-chain process-ing allows you to trigger a Plug-In from a separate reference track or external audio source. The source used for triggering is re-ferred to as the key input.

In the case of a Plug-In such as the D3 Compressor/Limiter, you can use this capa-bility to control the dynamics of one audio signal with the dynamics of another signal (the key input). A typical use for this fea-ture is to use one instrument to trigger gat-ing of another: a kick drum to gate and tighten up a bass track, a rhythm guitar to gate a keyboard pad, and so on.

Key Input FiltersSome Plug-Ins feature key high-pass and low-pass filters. These controls allow you to define a specific frequency range in the key input signal with which to trigger the Plug-In effect. A common production technique is to use these controls to filter a drum track so that only specific high frequencies (a hi-

hat, for example) or low frequencies (a tom or a kick, for example) trigger the effect.

To use a key input for side chain processing:

1 Click the Side Chain Input menu and choose the input/bus carrying the audio you want to use to trigger the Plug-In.

2 Click the External Key button on the Plug-In. This activates side chain process-ing.

3 To hear the audio source you have se-lected to control side chain input, click the Key Listen button.

4 To filter the Key Input so that only spe-cific frequencies trigger the Plug-In, use the Key HPF and Key LPF controls (if available) to select the desired frequency range.

5 Begin playback. The Plug-In uses the in-put/bus that you chose as a side chain in-put to trigger its effect.

6 Adjust the Plug-In’s Threshold parameter (if available) to fine tune side-chain trigger-ing.

7 Adjust other parameters of the Plug-In to achieve the desired effect.

Selecting an input from the Key Input pop-up


12

Automating Plug-InsYou can use the automation features of Pro Tools to automate changes in Plug-In parameters in real time. Since Pro Tools creates a separate playlist for each Plug-In parameter that you automate, you can later edit and modify each automated parameter individually. This allows you to build up complex automation in stages.

To automate a Plug-In:

1 Choose Window > Show Automation En-able.

2 In the Automation Enable window, select Plug-In to enable automation recording for Plug-Ins.

3 In the Mix or Edit window, select the au-tomation mode for the tracks you are auto-mating. For an initial automation pass, choose Auto Write.

4 Open the Inserts/Sends Editor for the Plug-In and click Automation.

5 Select the name and click Add for each parameter that you want to automate. Click Done when you are finished.

6 Click Play to begin automation recording and adjust the parameters in real time.

7 When you have finished, click Stop. If you are in Touch mode, automation re-cording will stop when you release the con-trol and ramp back to the original value at the rate specified in the AutoMatch Time preference.

To add additional automation:

■ To augment your previous pass, follow the above procedure using Touch mode. This will add new automation only when you actually touch or move a control.


Using the Librarian The Settings Librarian makes it easy to cre-ate your own library of commonly used Plug-Ins settings. Using the Librarian and Settings menus, you can copy, paste, save, and import these settings from Plug-In to Plug-In, or from session to session.

Once you create and save settings files to disk (and let Pro Tools know where to look for them by assigning their root folder) they will appear in the Librarian menu. You can save your settings in the current session or to a different location.

Click the Automation button to access the Plug-In Automation dialog

The Plug-In Automation dialog

✽ Pro Tools always saves the current Plug-Ins settings within your session document. The li-brarian merely allows you to access a perma-nent collection of settings for use in any session.

The Settings menu contains the following commands:

Save Settings Saves the parameter settings of the currently active Plug-In as a file on hard disk. This command overwrites any previous version of the settings that you have saved. The setting then appears in the Librarian pop-up menu. You cannot undo the Save Settings command.

Save Settings As Saves the settings of the currently active Plug-In under a different

The Librarian pop-up menu

The Settings pop-up menu

name, or in a different hard disk location. It is particularly useful if you are experi-menting and want to save different ver-sions.

Copy Settings Copies the current settings of a Plug-In. You can then apply these set-tings to the same type of Plug-in on a dif-ferent track by calling up the track in the Inserts/Sends Editor and pasting the set-tings with the Paste Settings command. Plug-In settings can be copied and pasted during playback. To apply a Plug-In’s set-tings in another session, use the Save Set-tings and Import Settings commands.

Paste Settings Pastes settings copied with the Copy Settings command. Plug-In set-tings can be copied and pasted during play-back. To apply a Plug-In’s settings in an-other session, use the Save Settings and Import Settings commands.

Import Settings Imports a settings file from a location other than your current Root Plug-In Settings folder or Session’s Settings folder. When the setting is loaded, it is ap-plied to the currently active Plug-In. Pro Tools adds the imported setting to the current library. The Import Settings com-mand is particularly useful for loading set-tings files that have been created on or transferred from another Pro Tools system.

Delete Current Settings File Allows you to permanently delete the currently selected settings from disk.

Set As User Default Defines the current set-tings as the User Setting for that type of Plug-In. Once you have done this, you can use the Set Plug-In Default To command to have the Plug-In default to your custom settings whenever you first place it on a


14

track. Normally, Pro Tools defaults to the Factory Default setting.

Settings Preferences

This command allows you to choose sev-eral preferences for saving and importing Plug-In settings. The commands available here are:

Set Plug-In Default To Sets the current Plug-In default settings to either Factory De-fault, (which is the Pro Tools standard de-fault settings for that Plug-In) or to User Setting, (which is your own custom set-ting). Use the Set As User Default command to define a custom User Setting first. Whichever you choose will be the default setting for the Plug-In when it is inserted on a track.

Save Plug-In Settings To Allows you to se-lect the folder where Plug-In settings are saved. You can set this to either Session’s Settings Folder, or Root Settings Folder. If you choose Session’s Settings Folder, Pro Tools will save the settings in a folder called Plug-In Settings Folder within the current session’s main Session Folder. If you choose Root Settings Folder, Pro Tools will save them to a folder that you have specified with the Set Root Settings Folder command.

Set Root Settings Folder Allows you to choose a source folder (referred to as a root folder) for saving and importing Plug-In settings. When you save Plug-In settings, Pro Tools will then save them there. When you import Plug-In settings, Pro Tools will look there first. Since Pro Tools automati-cally creates a Plug-In Setting folder inside the DAE folder, you don’t need to create a


root settings folder unless you explicitly want to use a different folder for your li-brary of Plug-In settings.

Choosing a Destination for SettingsBefore you save any Plug-In settings, select their destination folder.

To set the destination folder for settings:

■ From the Settings pop-up menu, choose Settings Preferences > Save Plug-In Settings To, and choose either Session’s Plug-In Set-tings Folder or Root Settings Folder.

If you choose Root Settings folder, Pro Tools saves to the Plug-In Settings folder within your DAE Folder.

To set a different folder, you must tell Pro Tools where to look for this folder.

To set your Root Settings folder:

1 From the Settings pop-up menu, choose Settings Preferences > Set Root Settings Folder.

2 Select the folder you want to use as your Root folder and click Select “(your folder’s name).”

Saving Plug-In SettingsTo save the current settings of a Plug-In, use the Save Settings command. This will save the settings to the destination folder you specified with the Save Plug-In Settings To command.

To save Plug-In settings:

1 With the Plug-In open, click the Settings pop-up menu and choose Save Settings.

2 Enter a name for the settings file and click OK. The setting now appears in the Li-brarian menu.

To load a previously saved setting:

■ With the Plug-In open, click the Librar-ian pop-up menu and select the setting.

Importing Plug-In SettingsIf you keep banks of settings files on other hard disks, floppy disks, or removable me-dia, you can use the Import Settings com-mand to access these settings.

To import a Plug-In settings file:

1 With the Plug-In open, click the Settings menu and choose Import Settings.

2 Locate the desired settings file and click Open. Pro Tools loads the settings and cop-ies them to your current root destination folder for future use.

Copying Plug-In SettingsThe Copy Settings command allows you to copy the settings from one Plug-In so that you can then paste them into another.

The Librarian menu

To copy Plug-In settings:

■ With the Plug-In open, choose Copy Set-tings from the Settings pop-up menu.

Pasting Plug-In SettingsOnce you have copied a Plug-In’s settings, you can paste them into a destination Plug-In of the same type.

To paste Plug-In settings:

1 Open the destination Plug-In.

2 From the Settings pop-up menu, choose Paste Settings. Pro Tools pastes the previ-ously copied settings into the destination Plug-In.

Creating a Custom User Default SettingThe Set As User Default command allows you to create a custom setting for any cate-gory of Plug-In and use that setting as the default User Setting.

You can make a Plug-In default to this cus-tom setting whenever you first insert it on a track by using the Set Plug-In Default To command. This feature saves you the trou-ble of loading a setting each time you want to use it.

To create a custom User Default setting:

1 Set the Plug-In’s parameters as desired.

2 Click the Settings pop-up menu and choose Save Settings As.

3 Name the settings file and click Save.

4 From the Settings pop-up menu, choose Set As User Default. Pro Tools defines the


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current settings as the User Default for that type of Plug-In.

To make a Plug-In default to your custom setting:

■ From the Settings pop-up menu, choose Set Plug-In Default To > User Setting. Pro Tools now uses the custom settings as the default for the Plug-In.

The Factory Default Setting

The Factory Default setting is always avail-able in the Librarian menu and is used as the default settings for a Plug-In (unless you use the Set As User Default command to change it to something else).

Because the Factory Default setting is al-ways available, choosing it allows you to access the original default settings for your Plug-In as a starting point at any time.

Creating Library SubfoldersPro Tools automatically creates subfolders in the root destination folder for each type of Plug-In. You may wish to subdivide your settings files still further by creating sub-folders for them.

For example, if you are using D-Verb, you may wish to organize settings files/folders by room size or algorithm type. This allows you to keep your library well organized with hierarchical menus for accessing Plug-In settings.

To create a settings subfolder:

1 Set the Plug-In’s parameters as desired.


2 From the Settings pop-up menu, choose Save Settings.

3 Click the New Folder button and enter a name for the subfolder.

4 Name the Plug-In setting and click Save. The new setting is saved within the sub-folder.

The Librarian menu with subfolders for settings

chapter 3

Working with AudioSuite Plug-Ins

Non-real-time AudioSuite Plug-Ins differ from TDM and RTAS Plug-Ins in that they are not used non-destructively in real time, but are instead used to process and modify audio files on disk. Depending on how you configure a non-real-time AudioSuite Plug-In, it will either alter the original source au-dio file or create an entirely new audio source file.

In Pro Tools, non-real-time AudioSuite Plug-Ins are accessed from the AudioSuite menu.

The AudioSuite menu

The AudioSuite WindowWhen you choose a Plug-In from the Pro Tools AudioSuite menu, the AudioSuite window appears. This floating window al-lows you to access and edit the parameters for any AudioSuite Plug-In.

The controls found in this window are as follows:

The AudioSuite controls

Settingspop-up

Librarianpop-up

File modepop-up

Plug-Inselector

Comparebutton

Processmodepop-up

SelectionReferencepop-up

Key Inputselector

Use in Playlistenable

C
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Plug-In Selector

This menu allows you to select any Audio-Suite Plug-In installed in your Plug-Ins folder.

Selection Reference Selector

This menu allows you to aim the selected process at either regions selected in an au-dio track/playlist, or at regions selected in the Audio Regions List.

When you select a region on-screen (in ei-ther a track/playlist or the Audio Regions List) Pro Tools normally selects both occur-rences of the region. Since you may not wish to process both, this menu provides a convenient means of limiting the Audio-Suite process to regions selected in one or the other.

Playlist Applies AudioSuite processing only to regions currently selected in tracks/play-lists in the Edit window. Processing will not be applied to regions in the Audio Regions List.

Region List Applies AudioSuite processing only to regions currently selected in the

The Plug-In Selector pop-up menu

The Selection Reference pop-up menu


Audio Regions List. Processing will not be applied to regions located in tracks/playl-ists in the Edit window.

▲ If an AudioSuite Plug-In supports stereo pro-cessing, the Selection Reference must be set to Playlist for the mono/stereo processing se-lector to be available.

Use in Playlist Button

This button determines whether the Audi-oSuite Plug-In will replace all instances of the selected regions everywhere in the ses-sion, or only the occurrences of the regions that are currently selected.

Use In Playlist Off When Use in Playlist is disabled, only the currently selected region will be replaced.

Use In Playlist On, Reference = Region List When Use in Playlist is enabled and the Se-lection Reference pop-up menu is set to Re-gions List, all copies of the selected region will be replaced everywhere in the session.

Use In Playlist On, Reference = Playlist When Use in Playlist is enabled and the Se-lection Reference menu is set to Playlist, only regions selected in tracks in the Edit window will be replaced. If the session in-cludes other copies of the region used in other playlists, those copies of the original, unprocessed region will not be replaced.

The Use in Playlist button

▲ If the Selection Reference pop-up menu is set to Region List, the Use in Playlist button will automatically be disabled to ensure that you do not accidentally replace every occur-rence of the region in a session.

File Mode Selector

This menu allows you to select whether or not the AudioSuite Plug-In will process the selected audio destructively or non-de-structively, and how the original files will be modified.

There are three options:

Overwrite Processes the selected regions de-structively, overwriting the original audio. Not all AudioSuite Plug-Ins can be used de-structively.

Create Individual Files Processes the se-lected regions non-destructively, creating a new audio file for each region. The new au-dio files are added to the session, leaving the original source audio files in the Re-gions List unchanged. (Whether the pro-cessed audio is added to the current playlist or only to the Regions List, is determined by the Selection Reference setting, as ex-plained earlier.)

Create Continuous File Processes the se-lected regions non-destructively and cre-ates a new audio file consisting of the se-lected regions “glued” into a into a single, unbroken region. This mode is particularly

The File Mode pop-up menu

useful if you are assembling a composite track from multiple takes.

✽ The create continuous file option is not available with some time domain Plug-Ins. To achieve a similar result, use the Pro Tools Du-plicate Plug-In to glue together regions pro-cessed with these Plug-Ins.

Process Mode Selector

If you have made a selection which in-cludes multiple regions, this pop-up menu allows you to specify whether AudioSuite processing is performed on a region-by-re-gion, or entire-selection basis.

Region by Region Analyzes each region in a selection individually, rather than over the entire multi-region selection as a whole.

Entire Selection Uses the entire selection for analysis. All regions will be analyzed and processed relative to the peak level within the entire selection.

Track Process Mode Selector

If you have made a selection which in-cludes regions from multiple tracks, some AudioSuite Plug-Ins allow you to choose whether you wish to perform processing on a track-by-track, or an all-tracks basis.

This pop-up has the same effect as the Pro-cess Mode Selector, with one important

Track Process Mode Selector

Chapter 3: Working with AudioSuite Plug-Ins 19

20

distinction: the Process mode pop-up con-trols region processing within each track. The Track Process Mode Selector, on the other hand, lets you choose to apply the current AudioSuite process to each track in-dividually, or to all tracks collectively.

Peak On Each Track Analyzes each selected track individually. If you apply the Normal-ize Plug-In to multiple tracks in Peak on Each Track mode, for example, each track will be Normalized without regard to the other selected tracks.

Peak From All Tracks Uses all currently se-lected tracks for analysis. If you apply the Normalize Plug-In to multiple tracks in Peak From All Tracks mode, for example, the tracks will be analyzed as a single entity and regions will be normalized relative to the peak level within all selected tracks.

Preview Button

The Preview button allows you to audition the effect of a Plug-In before you process the audio. By adjusting the Plug-In param-eters while you listen to this audio preview, you can fine-tune the effect. Not all Audio-Suite Plug-Ins support this feature.

Before you use Preview, be aware that:

◆ The performance of the Preview function depends on the speed of your CPU. Faster computers preview AudioSuite effects bet-ter than slower computers.

◆ On Macintosh computers, the preview function’s output level is controlled by the Macintosh Sound Manager, not by the fader level of the track containing the re-gion. To adjust Preview volume level, use the Sound Control Panel in the Apple Menu.


◆ Regardless of how many tracks/regions are currently selected, the Preview button will only play the first selected track if a Plug-In is in mono mode, or the first two consecutive tracks if the Plug-In is in stereo mode.

◆ If you are using Region-by-Region pro-cessing mode, the Preview function will preview only the first region within a multi-region selection. To hear all selected regions, temporarily select Create Continu-ous File from the File Mode menu before previewing.

◆ The Preview function always routes au-dio to channels 1–2 of the Audio Interface.

◆ The Preview function is affected by the AudioSuite Buffer Size parameter (on the Processing page of the Pro Tools Prefer-ences dialog). See “The AudioSuite Process-ing Preferences Dialog” on page 22.

Bypass

Clicking this button previews the selected audio without the effect of the current Au-dioSuite Plug-In. When Bypass is enabled, the selected audio is auditioned without AudioSuite processing. The Bypass button applies only to previewing. It does not af-fect actual AudioSuite processing.

Process

Clicking this button begins AudioSuite pro-cessing of the selected audio. Processing can occur during playback (though it may take slightly longer). Processed files are auto-named with the region or audio file’s name plus an acronym for the chosen Au-dioSuite process.

New files are written to the hard disk spec-ified for that track in the Disk Allocation dialog, or to the same drive as the original file if the region is not currently on a track. See “Auto-File Naming of AudioSuite-Pro-cessed Audio” on page 21.

Stereo Processing

If the AudioSuite Plug-In you wish to use supports stereo processing, regions can be selected and processed as a stereo pair as long as they occur at the same time loca-tion on their respective playlists.

✽ Group tracks that form stereo pairs. This will allow you to quickly select them on-screen. Also, name stereo pairs with suffixes such as “L/ R” (for example, “Drums.L” and “Drums.R”). This will keep pairs of audio files/regions together in the Regions List, and make it easy to select them for AudioSuite pro-cessing.

Undoing AudioSuite Processing

If you have processed an audio selection non-destructively, the Undo and Redo commands allow you to undo the selected AudioSuite process. You can undo/redo an AudioSuite process during audio playback.

Since Pro Tools allows a single level of Undo, if you perform any edit operation immediately following a non-destructive AudioSuite process, you will no longer be able to undo that process.

▲ Undo is not available when a Plug-In is con-figured for destructive editing since the pro-cess has already overwritten the source audio file.

Auto-File Naming of AudioSuite-Processed Audio

When new audio files are created as a result of AudioSuite processing, Pro Tools will auto-name these files according to the type of Plug-In used and the type of Selection Reference chosen (either Playlist or Region List). The Selection Reference setting deter-mines the prefix, while the type of Audio-Suite Plug-In determines the suffix.

Auto file-naming follows these rules:

◆ New regions are named beginning with the track name, followed by an abbrevia-tion of the current AudioSuite process, fol-lowed by standard Pro Tools file and region numbering.

◆ If the Selection Reference pop-up is set to Playlist, new region names will begin with the current track name.

◆ If the Selection Reference pop-up is set to Region List, new region names will begin with the original region’s name. For exam-ple, if you use the Reverse Plug-In, the ab-breviation “-RVRS” will be appended to the original region’s name. If the track contain-ing the target region is called “gtr effects,” the resulting region will be named “gtr ef-fects-RVRS.”

◆ If a Plug-In’s File Mode pop-up is set to Overwrite, the original region’s name will not be changed.

◆ If a Plug-In’s File Mode pop-up is set to Create Individual Files, the resulting re-gions will have an abbreviated version of the Plug-In’s name appended to them.


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Other AudioSuite Parameters

In addition to the standard AudioSuite pa-rameters, there are a number of special-pur-pose controls found on certain Plug-Ins:

Plug-In Librarian/Settings Menus The Set-tings and Librarian pop-up menus that ap-pear in some AudioSuite Plug-Ins provide a means of saving, loading, copying, pasting and organizing custom Plug-In settings files. Refer to the section entitled “Plug-In Librarian Functions” in Chapter 2 of this Guide for more information on these fea-tures.

Analyze Button The Analyze button allows you to perform an analysis pass on a selec-tion without actually processing it. For ex-ample, on the DigiRack Gain Plug-In, the Analyze feature allows you to determine the maximum peak level on a track at a specific gain value before you process the audio.

Key Input Selector The Key Input selector allows you to select a track/bus to be used to trigger processing. In order to use this feature, the key input source audio must occur at the same time as the target audio. Side chain processing and the use of a key input are covered in greater detail in Chap-ter 2 in the section entitled “Using a Key In-put for Side Chain Processing.”

The AudioSuite Processing Preferences DialogBefore you begin using AudioSuite Plug-Ins, use the Preferences dialog (Setups > Preferences > Processing) to configure de-fault AudioSuite parameters according to


your needs. These parameters include the default dither setting, a dither on/off checkbox, and the AudioSuite buffer size parameter.

Default Dither AudioSuite Plug-Ins always create files which inherit the session’s bit depth. The Default Dither pop-up allows you to choose the dither algorithm used for dithering higher-bit source files during Au-dioSuite processing. When the Use Default Dither checkbox is enabled, the chosen dither algorithm will be used during any AudioSuite processing that you perform. Using dither with your AudioSuite process-ing is most useful when working with low-level files whose levels fade in or fade out.

Edit Dither Settings Allows you to access any options for the presently selected De-fault Dither Plug-In. In the case of the Digidesign Dither Plug-In, this allows you to turn noise shaping on or off. The default position of the Noise Shape button is On.

The Processing Preferences page

Dither Depth Allows you to choose the de-fault bit depth used by the dither function during AudioSuite processing. Bit depth can be set to 16-, 18-, 20-, or 24-bit.

☞ For a detailed explanation of dither, refer to the Pro Tools Reference Guide.

AudioSuite Buffer Size Allows you to set the size of Pro Tools memory buffer for pro-cessing and previewing AudioSuite Plug-Ins. Choices are Mini, Small, Medium, Large, or Jumbo. In general, choosing a smaller buffer speeds up AudioSuite pre-viewing functions. Choosing a larger buffer speeds up AudioSuite processing of audio files.

To configure the AudioSuite Buffer Size:

1 Choose Setups > Preferences.

2 Click Processing.

3 Select an AudioSuite Buffer Size. Your choice will depend on whether you wish to give priority to the length of the audio pre-view, or to the speed at which the Plug-In can process the selected audio.

4 Click Done.

Though the default setting will work well for most situations, you may wish to set the buffer according to your current task: Be-fore you audition an AudioSuite Plug-In, set this buffer to Mini or Small. Then, when you are ready to process a file, reset it to Large or Jumbo.

Using AudioSuite Plug-InsAudioSuite Plug-Ins can be applied to whole regions, partial regions, or selections

that are made up of whole and partial re-gions across one or more tracks.

When audio selected in the Edit window includes partial regions, the regions will automatically be split into two or more re-gions when they are processed. Processing will occur only on the selection, leaving other regions unchanged.

Selecting Tracks for AudioSuite ProcessingBecause AudioSuite processes are per-formed on the specific regions that you se-lect, it is important that you select only those regions you actually want to process.

Selecting a region in the Audio Regions List will automatically select it in a track (if it currently resides on one). This is Pro Tools default behavior. You can change this, however, using the Region Selection Fol-lows Track Selection, and Track Selection Follows Regions List Selection options in the Editing page of the Preferences dialog. See the Pro Tools Reference Guide for an ex-planation of these Preferences.

Selecting Tracks for Delay or Reverb ProcessingBecause some AudioSuite effects such as de-lay and reverb add additional material to the end of the selected audio (a reverb tail or a delay tap), it is very important that you make a selection that is longer than the original source material so that the Plug-In can write it into the audio file.

If you select only the original material, without leaving additional space at the


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end, a reverb decay or delay that occurs af-ter the end of the region will be cut off.

To accommodate for this, place the region in a track, and select the desired audio plus an amount of blank space at the end of the region equal to the amount of delay or re-verb decay that you have added in the Plug-In. The Plug-In will then have space at the end of the region in which to write the final delay/decay.

To process audio with an AudioSuite Plug-in:

1 Select the desired regions in the target track(s) and/or in the Audio Regions List. (Shift-click to select multiple regions.) Only regions that are selected will be pro-cessed.

2 Choose the desired AudioSuite Plug-In from the AudioSuite menu.

3 Click the Preview button to begin play-back of the selected material.

4 Adjust the Plug-In parameters to achieve the effect that you want. These settings will determine how the file is processed and what effect the processing will have on the original regions. Here are some guidelines:

◆ To process the selected region only in the track in which it appears, choose Playlist from the Selection Reference pop-up. Alter-natively, if you wish to process the selected region in the Audio Regions list only, choose Region List from this pop-up.

◆ If you wish to process and update every occurrence of the selected region through-out your session, enable the Use In Playlist button (and also choose Region List from the Selection Reference pop-up). Alterna-tively, if you do not wish to update every occurrence of the selected region, disable the Use In Playlist button.


◆ To configure the Plug-In for destructive processing, choose Overwrite Files from the File Mode pop-up menu. This will over-write and permanently modify the original source audio files.

◆ Alternatively, to configure the Plug-In for non-destructive processing, choose Create Individual Files from the File Mode pop-up menu. This will create new audio files that have been processed with the AudioSuite Plug-in and leave the original source audio files untouched.

◆ If you have selected multiple regions for processing and wish to create a new file that connects and consolidates all of these regions together, choose Create Continu-ous File from the File Mode pop-up menu.

5 Finally, when you are ready, click the Pro-cess button. The selected audio is processed according to the settings you have speci-fied. Pro Tools appends an acronym to the region’s name indicating the AudioSuite process that has been applied. The new au-dio files then appear in your session.

✽ Grouping tracks that consist of stereo pairs will allow you to quickly select them on-screen.

✽ Name stereo pairs with suffixes such as “L/ R” (for example, “Drums.L” and “Drums.R”). This will keep pairs of audio files/regions to-gether in the Audio Regions List, and make it easy to select them for AudioSuite processing.

chapter 4

The D-Fi Plug-Ins

D-Fi consists of four separate Plug-Ins which form a unique sound design tool kit for processing and deconstructing audio in several retro and synthesis-oriented ways.

Lo-FiLo-Fi provides retro and down-processing effects, including:• Bit-rate reduction• Sample rate reduction• Soft clipping distortion and saturation• Anti-aliasing filter• Variable amplitude noise generator• Lo-Fi can be used as either a real-time

TDM/RTAS Plug-In or as a non-real-time AudioSuite Plug-In.

Sci-FiSci-Fi provides analog synthesizer-type ef-fects, including:• Ring modulation• Frequency modulation• Variable-frequency, positive and nega-

tive resonator• Modulation control by LFO, envelope

follower, sample-and-hold, or trigger-and-hold

• Sci-Fi can be used as either a real-time TDM/RTAS Plug-In or as a non-real-time AudioSuite Plug-In.

Recti-FiRecti-Fi provides additive harmonic pro-cessing effects via waveform rectification, and includes:• Subharmonic synthesizer• Full wave rectifier• Pre-filter for adjusting effect frequency• Post-filter for smoothing generated

waveforms• Recti-Fi can be used as either a real-time

TDM/RTAS Plug-In or as a non-real-time AudioSuite Plug-In.

Vari-FiVari-Fi provides audio speed-up or slow-down effects, to or from a complete stop (such as that produced by starting or stop-ping playback on a record turntable or an-alog tape recorder). Parameters are:• Speed up from a complete stop to normal

speed• Slow down to a complete stop from nor-

mal speed• Vari-Fi is an AudioSuite-only Plug-In.

Chapter 4: The D-Fi Plug-Ins 25

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Why Degrade Audio?The answer is simple—contemporary mu-sic styles, especially hip-hop, make exten-sive used of retro instruments and proces-sors such as vintage drum machines, samplers, and analog synthesizers. The low bit-rate resolutions and analog “grunge” of these devices are an essential and much-de-sired part of their sonic signatures. And that is why Digidesign created D-Fi.

The D-Fi suite of Plug-Ins combines the best of these instruments of the past with the flexibility and reliability of the Pro Tools audio production system. The re-sult is a set of sound design tools that al-lows you to achieve these retro sounds without the trouble and expense of resam-pling audio through 8-bit samplers or pro-cessing it through analog synthesizers.

Lo-FiThe Lo-Fi Plug-In is designed for down-pro-cessing audio by reducing its sample rate and bit resolution. It is ideal for emulating the grungy quality of 8-bit samplers.

Sample Rate

The sample rate slider allows you to adjust an audio file’s playback sample rate in fixed intervals from a high of the session’s cur-rent sample rate (44.1 kHz or 48 kHz) down to a low of 700 Hz. 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.


Anti-Alias Filter

The anti-alias filter works in conjunction with the sample rate control. As you reduce the sample rate, aliasing artifacts are pro-duced in the audio. These produce a char-acteristically dirty sound. Lo-Fi’s anti-alias filter has a default setting of 100%, auto-matically removing all aliasing artifacts as the sample rate is lowered.

This parameter is adjustable from 100% to 0%, allowing you to add precisely the amount of aliasing you desire back into the mix. This slider only has an effect if you have reduced the sample rate with the sam-ple rate slider (explained above).

Sample Size

The sample size slider controls the bit reso-lution 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 a high of 24 bits to a low of 2 bits.

The Lo-Fi Plug-In

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 either linear or adaptive quantization.

Linear Choosing linear resolution causes Lo-Fi to abruptly cut off sample data bits in an effort to fit the audio into the selected bit resolution. This imparts a characteristi-cally raunchy sound to the audio that be-comes 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 per-cussive, rhythmic effects to the audio sig-nal when it falls to lower levels, and a grungy quality as the audio reaches mid-levels.

Adaptive Choosing adaptive resolution causes Lo-Fi to reduce bit depth by adapt-ing 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 ap-parent bit resolution with a raunchiness that differs from the harsher quantization scheme used in linear resolution.

Noise Generator

The noise slider allows you to mix a per-centage of pseudo-white noise into the au-dio signal. Noise is useful for adding grit into a signal, especially when you are pro-cessing 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 sig-nal and the noise are equal in level.

Distortion/Saturation

The distortion and saturation sliders pro-vide signal clipping control. The distortion slider determines the amount of gain ap-plied and allows clipping to 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.

Fans of Spinal Tap will be pleased to know that both of the above controls can be set to eleven.

Sci-FiThe Sci-Fi Plug-In is designed to mock syn-thesize audio by adding effects such as ring modulation, resonation, and sample and hold, which are typically found on older, modular analog synthesizers. Sci-Fi is ideal for adding a bit of a synth edge to a track.

The Sci-Fi Plug-In


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Input Trim

This slider allows you to attenuate signal input level to the Sci-Fi processor. Since some of Sci-Fi’s parameters (such as the Resonator) can cause extreme changes in signal level, the input trim is particularly useful for achieving unity gain with the original signal level. The range of this con-trol is from –12 dB to 0 dB.

Effect Type

Sci-Fi allows you to choose between four different types of effects:

Ring Mod The ring modulator modulates the signal amplitude with a carrier fre-quency, producing harmonic sidebands that are the sum and difference of the fre-quencies of the two signals. The carrier fre-quency is supplied by Sci-Fi itself. The modulation frequency is determined by the effect frequency control. Ring modula-tion adds a characteristic hard-edged, me-tallic sound to audio.

Freak Mod Freak Mod is a frequency modu-lation processor that modulates the signal frequency with a carrier frequency, produc-ing harmonic sidebands that are the sum and difference of the input signal fre-quency 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 fre-quency slider determines the modulation frequency of the Freak Mod effect.

Resonator + and Resonator – Resonator + and Resonator – add a resonant frequency tone to the audio signal. This frequency is determined by the effect frequency slider.


The difference between these two modules is that Resonator – reverses the phase (po-larity) of the effect, producing a hollower sound than Resonator +. The Resonator can be used to produce metallic and flanging effects which emulate the sound of classic analog flangers.

Effect Amount

This slider controls the mix of the pro-cessed sound with the original signal. The range of this control is from 0–100%.

Effect Frequency

This slider controls the modulation fre-quency of the ring modulator and resona-tors. The frequency range is dependent on the effect type. For the ring modulator the frequency range of this control is from 0 Hz to 22.05 kHz. For Freak Mod, the fre-quency 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 5512 Hz.

Modulation Type

The modulation type buttons determine the type of modulation applied to the fre-quency of the selected effect. Depending on the type of modulation you select here, the parameter sliders below it will change to provide the appropriate type of modula-tion controls. If the mod amount slider is set to 0%, no dynamic modulation is ap-plied to the audio signal. In this case, the frequency set by the effect frequency slider will be the primary control that affects the sound.

Envelope Follower

Choosing the envelope follower causes the selected effect to dynamically track the in-put 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 auto-matic wah-wah-type effect. When you se-lect the envelope follower, the mod amount slider changes to a mod slewing slider. Slewing provides you with the abil-ity 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 mod-ulation will be.

Sample-and-Hold

Choosing sample-and-hold causes Sci-Fi to periodically sample a random pseudo-noise signal and apply this to the effect fre-quency. Sample and hold modulation pro-duces a characteristic random stair-step modulation. The sampling rate and the amplitude are determined by the mod rate and mod amount controls, respectively.

Trigger and Hold

Trigger and hold modulation is similar to sample and hold modulation, with one sig-nificant difference: If the input signal falls below a user-selectable 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 then seem to occur in a periodic, yet ran-dom way that varies directly with peaks in your audio material. Think of this type of modulation as having the best elements of

both sample & hold and the envelope fol-lower.

LFO

The LFO (low-frequency oscillator) pro-duces 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, re-spectively.

Mod Amount/Mod Rate

These two sliders control the amplitude and frequency of the modulating signal. The modulation amount ranges from 0% to 100%. The modulation rate available 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 control becomes a mod threshold control, ranging from –95 dB to 0 dB. It will then control the level below which modulation does not occur with the Trigger and Hold function.

If you select Envelope Follower as a modu-lation type, the mod rate control becomes a mod slewing control, which is adjustable from 0% to 100%.


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Recti-FiThe Recti-Fi Plug-In is designed to provide additive synthesis effects via waveform rec-tification. One of the most common uses for Recti-Fi is for multiplying the harmonic content of a track and adding subharmonic or superharmonic tones.

Recti-Fi Controls

Pre-Filter

The pre-filter allows you to filter out high frequencies in an audio signal prior to rec-tification. This is often desirable since 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 recti-fication can improve waveform stability and the quality of the rectification effect. If you wish to create classic subharmonic synthesis effects, set the pre- and post-filter to a relatively low frequency, typically 250Hz or so.

The Recti-Fi Plug-In


Rectification

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.

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.

Alternating Rectification

This alternates between rectifying the phase of the first negative waveform excur-sion to positive, then the next positive ex-cursion to negative, and so on, throughout the waveform. The audible effect is a halv-ing of the audio signal’s frequency, creat-ing a subharmonic tone.

A normal waveform

positive rectification

negative rectification

alternating rectification

Alt-Max Rectification

This alternates between holding the maxi-mum value of the first positive excursion through the negative excursion period, switching to rectify the next positive excur-sion, and holding its peak negative value until the next zero crossing. The audible ef-fect is a halving of the audio signal’s fre-quency, and creating a subharmonic tone with a hollow, square wave-like timbre.

Gain

The gain control allows you to adjust signal level before the audio reaches the Post-Fil-ter. This is particularly useful for restoring unity gain if you have used the pre-filter to cut off high frequencies prior to rectifica-tion. The range of this control is from –18dB to +18dB.

Post-Filter

Waveform rectification, particularly alter-nating rectification, typically produces a great number of harmonics. The post filter allows you to remove harmonics above the cutoff frequency and smooth out the sound. This post-filter is useful for filtering audio that contains subharmonics. To cre-ate classic subharmonic synthesis effects, set the pre- and post-filter to a relatively low frequency.

Mix

This control allows you to adjust the mix of the rectified waveform with the original, unprocessed waveform.

alt-max rectification

Vari-FiVari-Fi is an AudioSuite-only Plug-In that provides a pitch-change effect that is de-signed to emulate the sound of a tape deck or record turntable speeding up from or slowing down to a complete stop. Vari-Fi achieves this effect while preserving the original duration of the audio selection.

Vari-Fi Controls

Speed Up

This option adds a pitch-change effect to the selected audio, similar to that of a reel of tape or record turntable being sped up from a stop. The pitch change effect does this without changing the duration of the audio selection.

Slow Down

This option adds a pitch-change effect to the selected audio, similar to that of a reel of tape or record turntable being slowed down to a stop. The pitch change effect does this without changing the duration of the audio selection.

The Vari-Fi Plug-In


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D-Fi Demo Session D-Fi includes a demo session that illus-trates some of the many effects you can produce with Lo-Fi, Sci-Fi, and Recti-Fi. This following section explains how to open and use this demo session.

The librarian settings that are featured in the demo session are automatically in-stalled with D-Fi and are available for use at any time. We encourage you to experiment to find new and creative uses for these set-ting and for each of the D-Fi Plug-Ins.

About the Demo SessionThe D-Fi demo session is a very simple ses-sion file containing drum, bass, and guitar loops. Memory locations have been pre-defined in the session. These allow you to quickly zero in on a particular loop and ap-ply different D-Fi effects using the In-serts/Sends editor.

Sci-Fi and Lo-Fi have been placed on a ste-reo master fader routed to outputs 1 & 2 of your system. This allows different tracks with different program material to be audi-tioned and processed by the Plug-Ins.

If you experiment with any of the settings and like the results, you can use the Save As command in the Settings menu to save your own versions for later use.

Before you start, please do the following:

1 Move the D-Fi demo from the Pro Tools folder to a Pro Tools audio drive.

2 Open the Demo session.


3 Choose Windows > Show Memory Loca-tions. You are now ready to use the demo session.

Sci-Fi ExamplesThe following examples demonstrate Sci-Fi. Follow the instructions in each sec-tion below to hear some useful applications for this Plug-In.

Hi-Hat Loop

1 Click memory location #1, “Hat Loop.”

2 Click the Sci-Fi insert on the Master Fader to display Sci-Fi.

3 Press the Space bar to audition the Hi-Hat loop. Since the Bypass button is enabled, you will hear the loop without Sci-Fi pro-cessing.

The Memory Locations window

Choosing a Sci-Fi setting from the Librarian pop-up

4 Press the Space bar to stop the Hi-Hat loop.

5 Click the Librarian pop-up and choose the setting named “Res-1/4 note Trig. & Hold.”

6 Click and disable the Bypass button so that you will be able to hear Sci-Fi’s effects.

7 Press the Space bar to audition the Hi-Hat loop.

8 Listen to the effect. Note how trigger & hold is used to allow modulation to follow the amplitude. This provides a much more interesting type of modulation than stan-dard envelope following.

9 Adjust the mod threshold to vary the modulation on 1/4 note accents.

10 From the Librarian pop-up, choose the setting called “Res. -16 note Trig & Hold.” This setting demonstrates a similar type of modulation which occurs on 16th notes.

11 From the Librarian pop-up, choose the setting called “Wah Res-LFO Faux Flange.” This setting demonstrates a basic flanging-type effect. Try changing the rate parame-ter and also switching to the + resonator. Experimenting with the mod type as well for interesting effects.

Drum Kit Loop

1 Click memory location # 2, “Drum Kit Loop.”

2 Click and enable the Bypass button to hear the Drum Kit loop without Sci-Fi pro-cessing.

3 From the Librarian pop-up, choose the setting called “Ring Mod Trig & Hold Kit.”

4 Click and disable the Bypass button so that you will be able to hear Sci-Fi’s effects.

5 Press the Space bar to audition the Drum Kit loop. This setting uses ring modulation and trigger & hold to create ring modula-tion that changes only on audio peaks.

6 From the Librarian pop-up, choose the setting called “Res-Env. Follower.” This set-ting demonstrates the use of the envelope follower to create resonant flanging that modulates and matches the dynamics of the source audio.

7 From the Librarian pop-up, choose the setting called “Freq. Mod Env. F. Kit.” This setting demonstrates the above principle but with frequency modulation.

8 Experiment with the other settings.

9 Finally, click memory location #4, “Bass/Drums Loop.” Try each of the Sci-Fi settings with this loop, which contains bass and drums together.

Wah Guitar Loop

1 Click memory location # 3, “Wah Guitar Loop.”

2 Click and enable the Bypass button to hear this loop without Sci-Fi processing.

3 From the Librarian pop-up, choose the setting called “Freq Mod Env. Follower Wah.”

4 Disable the Bypass button so that you will be able to hear Sci-Fi’s effects.

5 Press the Space bar to audition this loop.

6 Try each of the Sci-Fi settings with this loop.


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Lo-Fi ExamplesThe following examples demonstrate Lo-Fi. Follow the instructions in each section be-low to hear some useful applications for this Plug-in.

Before you begin, click the Bypass button in the Sci-Fi Plug-In to take it out of the mix. Then click the Lo-Fi insert on the Master Fader to display Lo-Fi.

Slam Kit Loop

1 Click the Lo-Fi insert on the Master Fader to display Lo-Fi in the Inserts/Send editor.

2 Click memory location #7, “Slam Kit Loop.”

3 Enable the Bypass button to hear this loop without Lo-Fi processing.

4 Press the Space bar to audition this loop.

5 Disable the Bypass button so that you will be able to hear Lo-Fi’s effects.

6 Try each of the other Lo-Fi settings with this loop.

7 This loop has a hip-hop feel, and demon-strates how Lo-Fi can be used to create in-teresting textures with harder percussive elements.

Drum Kit Loop

1 Click memory location #2, “Drum Kit Loop.”

Choosing a Lo-Fi setting from the Librarian pop-up


2 From the Librarian pop-up, choose the setting called “Lo-rate Distorto Kit.”

3 This loop demonstrates how Lo-Fi can be used to create excellent grunged-up drums.

4 Try experimenting with the sample rate, saturation, and distortion controls to vary the results.

Bass Only

1 Click memory location #6, “Bass Only.”

2 From the Librarian pop-up, choose the setting called “Bass Dirty Amp.”

3 Use the Bypass button to compare the sound of the processed and unprocessed bass.

4 This setting provides a good example of a slightly gritty bass amp with limited high-end. Try adjusting the saturation and dis-tortion controls to study distortion charac-teristics.

5 From the Librarian pop-up, choose the setting called “Trash Bass.” This setting provides some unusual distortion. Try ex-perimenting with bit depth to understand how it affects the audio.

6 From the Librarian pop-up, choose the setting called “Ring Moddy Bass.” This set-ting demonstrates some very extreme Lo-Fi processing.

Recti-Fi ExamplesThe following examples demonstrate Recti-Fi. Follow the instructions in each section below to hear some useful applica-tions for this Plug-In.

Before you begin, go to the Mix window and remove Lo-Fi on the Master Fader, and instead choose Recti-Fi.

Sub Octave Bass

1 Click memory location #6, “Bass Only.”

2 From the Librarian pop-up, choose the setting called “Sub Octave Bass.”

3 In this setting, the pre/post filters are op-timized for standard octave-doubling be-neath the bass.

Sub-Oct. Heavy Bass

1 From the Librarian pop-up, choose the setting called “Sub-Oct. Heavy Bass.”

2 This setting uses alt max rectification to provide more bottom end. Try experiment-ing with the mix parameter and other con-trols.

Drum Kit Loop

1 Click memory location #2, “Drum Kit Loop.”

2 From the Librarian pop-up, choose the setting called “Sub Kit.”

Choosing a Recti-Fi setting from the Librarian pop-up

3 Compare the sound of the processed and unprocessed audio by using the Bypass but-ton.

4 This setting demonstrates how to use sub-octave rectification to enhance low-fre-quencies.

Hat Loop

1 Click memory location #1, “Hat Loop.”

2 From the Librarian pop-up, choose the setting called “Noise Hat.”

3 Compare the sound of the processed and unprocessed audio by using the Bypass but-ton.

4 This setting demonstrates how Recti-Fi can produce a periodic noise version of the hi-hat which varies along with the original audio.

5 Adjust the mix parameter to hear the sig-nal fully wet.

6 Adjust the pre-filter during playback and listen to the results. Automating changes in the pre-filter frequency can produce very interesting and useful effects.

Wah Guitar

1 Click memory location #3, “Wah Gui-tar.”

2 From the Librarian pop-up, choose the setting called “Up Octave Wah.”

3 Audition the audio.

4 This setting produces a signal which is an octave higher than the original and adds some interesting audio artifacts.

5 Experiment with the mix control and pre/post filters and listen to the results.


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Slam Kit Loop

1 Click memory location #7, “Slam Kit Loop.”

2 From the Librarian pop-up, choose the setting called “Trash Kit.”

3 Audition the audio.

4 This setting illustrates the use of Recti-Fi as a basic sound modifier for percussive sounds.

5 Try experimenting with the pre/post controls and listen to the results.

This concludes the D-Fi demo. You should now have an idea of some of the creative applications for D-Fi. We encourage you to experiment and find more of your own.


chapter 5

The D-fx Plug-Ins

D-fx is a set of five non-real-time Audio-Suite Plug-Ins that provide the following time-delay based effects:• Chorus• Reverb• Flanging• Multi-tap delay• Ping-Pong delay

This suite of Plug-Ins is ideal for thicken-ing, spatializing and adding a sense of space and depth to elements in a mix.

D-fx ChorusThe Chorus Plug-In modifies an audio sig-nal by combining a time-delayed, pitch-shifted copy of it with the original signal. It is ideal for thickening and adding a shim-mering quality to guitars, keyboards, and other instruments.

The D-fx Plug-Ins appear in the AudioSuite menu

Input Level The input level slider allows you to adjust the input volume of the cho-rus to prevent the possibility of clipping and/or increase the level of the processed signal. Please note that this slider is set to a default of +3 dB. If your source audio has been recorded very close to peak level, this +3 dB default setting may accidentally cause clipping. Use this slider to reduce the input level.

When you use this Plug-In in stereo mode, a Sum Inputs button appears next to the right channel’s input level slider. If you click and enable this button, D-fx will sum the dry input signals (regardless of whether the input is mono or stereo) before process-ing them.

The result is that the dry signal will appear in the center of the stereo field, and the wet, effected signal will be output in stereo.

The Chorus Plug-In

Chapter 5: The D-fx Plug-Ins 37

38

When the Sum Inputs button is engaged, the LFO waveform on the right channel is automatically phase inverted in order to enhance the mono-stereo effect.

Mix The mix slider allows you to adjust the balance between the effected signal and the original signal, giving you a control over the depth of the effect. This parameter is adjustable from 100% to 0%.

Low-Pass Filter This slider controls the cut-off frequency of the low-pass filter. This al-lows you to attenuate the high frequency content of the feedback signal. The lower the setting, the more high frequencies are removed from the feedback signal.

Delay This slider allows you to set the delay time between the original signal and the chorused signal. The higher the setting, the longer the delay and the wider the chorus-ing effect. This parameter is adjustable from a minimum of 0 milliseconds to a maximum of 20 milliseconds.

LFO Rate This slider allows you to adjust the rate of the LFO (low-frequency oscilla-tor) applied to the delayed signal as modu-lation. The higher the setting, the more rapid the modulation. D-fx allows you to select either a sine wave or a triangle wave as a modulation source, using the LFO waveform selector (explained below).

LFO Width This slider allows you to adjust the intensity of the LFO (low-frequency os-cillator) applied to the delayed signal as modulation. The higher the setting, the more intense the modulation. D-fx allows

The Sum Inputs button sums the dry input signal(s) before processing


you to select either a sine wave or a triangle wave as a modulation source using the LFO waveform selector (explained below).

Feedback This slider allows you to control the amount of feedback applied from the output of the delayed signal back into its input. Negative settings provide a more in-tense effect.

LFO Waveform These buttons allow you to choose between either a sine wave or trian-gle wave for the LFO. This subtly affects the character of the modulation. The sine wave has a somewhat gentler ramp and peak than the triangle wave.

D-fx FlangerThe Flanger Plug-In modifies an audio sig-nal by combining a slightly time-delayed, pitch shifted copy to the original signal. The D-fx Flanger differs from other digital flangers in that it uses a “thru-zero” flang-ing algorithm which results in a truer tape-like flange. This technique delays the orig-inal dry signal very slightly (approximately 256 samples) then modulates the delayed signal back and forth in time in relation to the dry signal—passing through its zero point on the way.

The Flanger is ideal for thickening and add-ing a swirling, moving quality to guitars and other instruments.

Input Level The input level slider allows you to adjust the input volume of the flanger to prevent the possibility of clip-ping and/or increase the level of the pro-cessed signal. Note that this slider is set to a default of +3 dB. If your source audio has

been recorded very close to peak level, this +3 dB default setting may accidentally cause clipping. Use this slider to reduce the input level.

When you use this Plug-In in stereo mode, a Sum Inputs button appears next to the right channel’s input level slider. If you click and enable this button, D-fx will sum the dry input signals (regardless of whether the input is mono or stereo) before process-ing them. The result is that the dry signal will appear in the center of the stereo field, and the wet, effected signal will be output in stereo. When the Sum Inputs button is engaged, the LFO waveform on the right channel is automatically phase inverted in order to enhance the mono-stereo effect.

Mix This slider allows you to adjust the bal-ance between the effected signal and the original signal, giving you a control over the depth of the effect. This parameter is adjustable from 100% to 0%.

High-Pass Filter This slider controls the cut-off frequency of the high-pass filter. This allows you to attenuate the frequency con-tent of the feedback signal and the fre-

The Flanger Plug-In


quency response of the flange effect. The higher the setting, the more low frequen-cies are removed from the feedback signal.

LFO Rate This slider allows you to adjust the rate of the LFO (low-frequency oscilla-tor) applied to the delayed signal as modu-lation. The higher the setting, the more rapid the modulation. D-fx allows you to select either a sine wave or a triangle wave as a modulation source using the LFO waveform selector (explained below).

LFO Width This slider allows you to adjust the intensity of the LFO (low-frequency os-cillator) applied to the delayed signal as modulation. The higher the setting, the more intense the modulation. D-fx allows you to select either a sine wave or a triangle wave as a modulation source using the LFO waveform selector (explained below).

Feedback This slider allows you to control the amount of feedback applied from the output of the delayed signal back into its input. Negative settings give a more in-tense tunnel-like effect to flanging.

LFO Waveform These selectors allow you to choose between either a sine wave or trian-gle wave for the LFO. This varies the char-acter of the modulation.

D-fx Multi-Tap DelayThe Multi-Tap delay Plug-In modifies an audio signal by adding up to four indepen-dently-controllable delays or “taps” to the original signal. By allowing you to individ-ually control the delay time and number of repetitions of each of the four taps, the


40

Multi-Tap delay provides much more flexi-bility than standard single-delay devices.

Multi-tap delay is ideal for adding spatial-ization or complex rhythmic echo effects to virtually any instrument.

Gain The Gain slider for each of the Multi-Tap delay’s four delay lines controls the in-put level of that individual delay tap. By adjusting these sliders you can prevent the possibility of clipping and/or increase the level of the processed signal.

When you use this Plug-In in stereo mode, a Sum Inputs button appears. If you click and enable this button, D-fx will sum the dry input signals (regardless of whether the input is mono or stereo). This will have no effect on the wet signal, since the delay lines always sum the inputs in stereo mode, regardless of whether Sum Inputs button is on or off. The result is that the dry signal will appear in the center of the stereo field, and the wet, effected signal will be output in stereo.

Feedback This slider allows you to control the amount of feedback applied from the

The Multi-Tap Delay Plug-In


output of the delay back into its input. It also controls the number of repetitions of the delayed signal. In order for the feed-back parameter to function, the gain slider must be raised above its lowest setting of –∞. If you don’t do this, there will be no sig-nal for the Multi-Tap Plug-In to apply feed-back to.

Pan When you use the Multi-Tap delay Plug-In in stereo mode, a pan slider appears for each of the four taps, allowing you to control their apparent location in the ste-reo field.

Delay This slider allows you to set the delay time between the original signal and the delayed signal. The higher the setting, the longer the delay. This parameter is adjust-able from a minimum of 0 milliseconds to a maximum of 1500 milliseconds (1.5 sec-onds).


D-fx Ping-Pong DelayThe Ping-Pong delay Plug-In modifies an audio signal by adding a controllable delay to the original signal. It is ideal for adding spatialization, and when used in stereo, panned echo to virtually any instrument. In stereo mode, this Plug-In will feed back

The Sum Inputs Button sums the dry input signal(s) before processing

delayed signals to their opposite channels, creating a characteristic ping-pong echo ef-fect.

Input Level This slider allows you to adjust the input volume of the ping-pong delay to prevent the possibility of clipping and/or increase the level of the processed signal.

When you use this Plug-In in stereo mode, a Sum Inputs button appears. If you click and enable this button, D-fx will sum the dry input signals (regardless of whether the input is mono or stereo) before processing them. The result is that the dry signal will appear in the center of the stereo field, and the wet, effected signal will be output in stereo.


Delay This slider allows you to set the delay time between the original signal and the delayed signal. The higher the setting, the longer the delay. This parameter is adjust-able from a minimum of 0 milliseconds to

The Ping-Pong Delay Plug-In


a maximum of 1500 milliseconds (1.5 sec-onds).

Low-Pass Filter This slider controls the cut-off frequency of the low-pass filter. This al-lows you to attenuate the high frequency content of the feedback signal. The lower the setting, the more high frequencies are removed from the feedback signal.

Feedback This slider allows you to control the amount of feedback applied from the output of the delay back into its input. It also controls the number of repetitions of the delayed signal.

Cross-Feedback When used in stereo mode, this Plug-In provides a cross feedback pa-rameter which feeds the delayed signals to their opposite channel—the left channel delay will be fed to the right channel input and vice-versa. The result is a stereo echo that “ping-pongs” back and forth between the right and left channels.

D-fx D-VerbThe D-Verb Plug-In modifies an audio sig-nal by simulating and adding the complex natural reflections and echoes that occur after a sound has been produced, imparting a sense of space and depth. It is ideal for adding reverberant ambience to virtually any instrument.

Input Level This slider allows you to adjust the input volume of the reverb to prevent the possibility of clipping and/or increase the level of the processed signal.

When you use this Plug-In in stereo mode, a Sum Inputs button appears. If you click and enable this button, D-fx will sum the


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dry input signals (regardless of whether the input is mono or stereo) before processing them. The result is that the dry signal will appear in the center of the stereo field, and the wet, effected signal will be output in stereo.

Wet/Dry This slider allows you to adjust the balance between the dry signal and the effected signal, giving you control over the depth of the effect. This parameter is ad-justable from 100% to 0%.

Algorithm This control allows you to select one of seven reverb algorithms: Hall, Church, Plate, Room 1, Room 2, Ambience, or Nonlinear. Selecting an algorithm changes the preset provided for it. Switch-ing the Size parameter actually changes characteristics of the algorithm that are not altered by simply adjusting the decay time

The D-Verb Plug-In



and other user-adjustable parameters. Each of the seven algorithms has a distinctly dif-ferent 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. When the decay time is set to its maximum value, infinite reverberation can be created, for special effects purposes.

◆ Church: A large, dense, diffuse space sim-ulating a church or cathedral with a long decay time, high diffusion, and some pre-delay. High-frequency content has been re-stricted on the preset setting in order to simulate the characteristics of such an en-vironment. When decay time is set to its maximum value, infinite reverberation can be created.

◆ Plate: This algorithm simulates the acoustic character of a metal plate-based re-verb. This type of reverb typically has high initial diffusion and a relatively bright sound, making it particularly good for cer-tain percussive signals and vocal process-ing. Plate reverb has the general effect of thickening the initial sound itself. Infinite reverb can be created by setting the decay time to its maximum.

◆ Room 1: A medium-sized, natural, rich-sounding room that can be effectively var-ied in size between very small and large, with good results. Infinite reverb can be created by setting the decay time to its maximum.

◆ Room 2: A smaller, brighter reverberant characteristic than Room 1, with a useful adjustment range that extends to “very small.” Infinite reverb can be created by setting the decay time to its maximum.

◆ Ambient: This algorithm has a rather transparent response that is useful for add-ing a sense of space without adding a lot of depth or density. Extreme settings can cre-ate interesting results.

◆ Nonlinear: This algorithm produces a re-verberation with a natural buildup and an abrupt cut off similar to a gate. This unnat-ural decay characteristic is particularly use-ful on percussion, since it can add an aggressive characteristic to sounds with strong attacks.

Size This control, in conjunction with the Algorithm parameter, allows you to adjust 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 parameter). The Size buttons can be used to vary the range of a reverb from large to small. Gen-erally, 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 This control sets the degree to which initial echo density increases over time. High settings result in high initial buildup of echo density. Low settings cause low initial buildup. This control interacts with the Size and Decay parameters to af-fect the overall reverb density. High set-tings of Diffusion can be used to enhance percussion. Use low of moderate settings for clearer and more natural-sounding vo-cals and mixes.

Decay This control allows you to adjust the rate at which the reverb decays after the original direct signal stops. The value of the decay parameter is affected by the Size and

Algorithm parameters. This parameter can be set to infinity (∞) on most algorithms for infinite reverb times.

Hi Frequency Cut This parameter controls the decay characteristic of the high fre-quency components of the reverb. It acts in conjunction with the Low Pass Filter con-trol 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.

Low-Pass Filter This parameter controls the overall high frequency content of the se-lected reverb by allowing you to set the fre-quency above which a 6 dB/octave filter at-tenuates (cuts) the processed signal.

Pre-Delay This parameter controls the amount of pre-delay. It has a maximum setting of 370 milliseconds. Pre-delay sets the amount of time which elapses between the input of a signal and the onset of rever-beration. 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 du-plicate this phenomenon and is used to create a sense of distance and volume within an acoustic space. Relatively long pre-delay settings place the reverberant field further behind rather than directly next to the input material.


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Selecting Tracks/Regions for D-fx Delay or Reverb ProcessingIn order to process a track/region with an AudioSuite Plug-In, you must first select the target track/region in either Pro Tools Edit window, or in the Audio Regions list. The Plug-In will then process the selected portion of the audio.

However, because D-fx effects such as re-verb, multi-tap delay, and ping-pong delay actually add additional material to the end of the selected audio (such as a reverb tail or a delay tap), it is very important that you make a selection that is actually longer than the original source material so that D-fx can write it into the audio file.

If you select only the original material, without leaving additional space at the end, a reverb decay or delay that occurs af-ter the end of the region will be cut off.

To accommodate for this, place your region in a track, and select the desired audio plus an amount of blank space at the end of the region equal to the amount of delay or re-verb decay that you have added in the Plug-In. D-fx will then have space at the end of the region in which to write the fi-nal delay/decay.

Using D-fx in StereoThe D-fx AudioSuite Plug-Ins can be used in either mono or stereo. If you plan to use them in stereo, please be aware of the fol-lowing:


◆ To process a mono track and obtain a ste-reo result, select the desired track/region plus an empty track/region, then set the Plug-In to stereo mode, and enable the Sum Inputs button if you wish to center the dry signal. When you process the audio, the re-sult will be two tracks/regions which repre-sent the right and left channels of the processed audio. You should then pan these tracks hard right and hard left in your mix.

◆ If you set a D-fx Plug-In to stereo mode, then select an odd number of Pro Tools tracks for processing (as opposed to an even number), D-fx will process the se-lected tracks in pairs, in stereo. However, the last odd (unpaired) track will be pro-cessed as mono, using the left channel set-tings of the stereo D-fx Plug-In. If you want the last track to be processed in stereo, you must select an additional track to pair it with—an empty one if necessary.

chapter 6

The D-Verb Plug-in

D-Verb is a studio-quality reverb provided in TDM, RTAS, and AudioSuite formats. The AudioSuite version of D-Verb appears as “D-FX D-Verb” in the AudioSuite Menu of Pro Tools.

About ReverbDigital reverberation processing can simu-late 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 envi-ronment. When you use a reverberation device such as D-Verb, 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 ad-justable mix of the original dry source and the reverberant wet signal. Reverberation can take relatively lifeless mono direct sources and create an acoustic environ-ment (in stereo) that gives the source a per-ceived weight and depth in a mix.

Creating Unique Sounds

In addition, digital signal processing can be used creatively to produce reverberation

characteristics that do not exist in nature. There are no rules that need to be followed to produce interesting treatments. Experi-mentation 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 downright unpleas-ant.) Each real world acoustical venue, from a small room to a large cathedral, has its own unique acoustical character that it imparts to the sound, which excites it.

Reverb Character

The character of the reverberation, which is in effect a mixture of the nature of the excitation source and the characteristic of the acoustical space, 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.

In a typical concert hall, sound reaches the listener shortly after it is produced. The original direct sound is followed by reflec-

Chapter 6: The D-Verb Plug-in 45

46

tions 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 rever-beration. Early reflections provide a sense of depth and strengthen the perception of loudness and clarity. The delay time be-tween 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 re-verberation and repetitive reflections and attenuation of the original sound reflected from walls, ceilings, floors, and other ob-jects. This sound provides a sense of depth or size.

D-Verb provides control over these rever-beration parameters so that extremely nat-ural-sounding reverb effects can be created an applied in the Pro Tools mix environ-ment.

D-Verb ControlsThe following section explains all of D-Verb’s parameters and their use.

Output Meter

The output meter indicates the output level of the processed signal. (In the case of the stereo version of D-Verb, it represents the summed stereo output.) It is important to note that this meter indicates the output level of the signal—not the input level. If this meter clips, it is possible that the signal actually clipped on input before it reached


D-Verb. Be careful to monitor your send or insert signal levels closely to help prevent this from happening.

Clip Indicator

This indicator shows if clipping has oc-curred. It is a clip hold indicator, meaning that if clipping occurs at any time, the clip light will remain on. To clear the clip indi-cator, click it with the mouse. With longer reverb times there is a greater likelihood of clipping occurring as the feedback element of the reverb builds up and approaches a high output level.

Input Level

The input level slider allows you to adjust the input volume of the reverb to prevent the possibility of clipping and/or increase the level of the processed signal.

Wet/Dry

The Wet/Dry slider allows you to adjust the balance between the dry signal and the ef-

The D-Verb Plug-In

fected signal, giving you control over the depth of the effect. This parameter is ad-justable from 100% to 0%.

Algorithm

This control allows you to select 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 parameter actually changes characteristics of the algorithm that are not altered by simply adjusting the decay time and other user-adjustable parameters. Each of the seven algorithms has a distinctly different character, described as follows:

◆ 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. When the decay time is set to its maximum value, infinite reverberation can be created, for special effects purposes.

◆ Church: A large, dense, diffuse space sim-ulating a church or cathedral with a long decay time, high diffusion, and some pre-delay. High-frequency content has been re-stricted on the preset setting in order to simulate the characteristics of such an en-vironment. When decay time is set to its maximum value, infinite reverberation can be created.

◆ Plate: This algorithm simulates the acoustic character of a metal plate-based re-verb. This type of reverb typically has high initial diffusion and a relatively bright sound, making it particularly good for cer-tain percussive signals and vocal process-ing. Plate reverb has the general effect of thickening the initial sound itself. Infinite

reverb is provided when decay time is set to maximum.

◆ Room 1: A medium-sized, natural, rich-sounding room that can be effectively var-ied in size between very small and large, with good results. Infinite reverb is pro-vided when decay time is set to maximum.

◆ Room 2: A smaller, brighter reverberant characteristic than Room 1, with a useful adjustment range that extends to “very small.” Infinite reverb is provided when de-cay time is set to maximum.

◆ Ambient: This algorithm has a rather transparent response that is useful for add-ing a sense of space without adding a lot of depth or density. Extreme settings can cre-ate interesting results.

◆ Nonlinear: This algorithm produces a re-verberation with a natural buildup and an abrupt cut off similar to a gate. This unnat-ural decay characteristic is particularly use-ful on percussion, since it can add an aggressive characteristic to sounds with strong attacks.

Size

This control, in conjunction with the Algo-rithm parameter, allows you to adjust 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 rel-ative value of the Decay parameter). 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 try-ing to create.


48

Diffusion

This control sets the degree to which initial echo density increases over time. High set-tings result in high initial buildup of echo density. Low settings cause low initial buildup. This control interacts with the Size and Decay parameters to affect the overall reverb density. High settings of dif-fusion can be used to enhance percussion. Use low or moderate settings for clearer and more natural-sounding vocals and mixes.

Decay

This control allows you to adjust the rate at which the reverb decays after the original direct signal stops. The value of the decay parameter is affected by the Size and Algo-rithm parameters. This parameter can be set to infinity on most algorithms for infi-nite reverb times.

Pre-Delay

This parameter controls the amount of pre-delay. It has a maximum setting of 370 mil-liseconds. Pre-delay sets the amount of time which elapses between the input of a signal and the onset of reverberation. Un-der natural conditions, the amount of pre-delay depends on the size and construction of the acoustic space and the relative posi-tion of the sound source and the listener. Pre-delay attempts to duplicate this phe-nomenon and is used to create a sense of distance and volume within an acoustic space. Relatively long pre-delay settings place the reverberant field further behind rather than directly next to the input mate-rial.


Hi Frequency Cut

This parameter controls the decay charac-teristic 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 re-verb. When set relatively low, high fre-quencies decay more quickly than low fre-quencies, simulating the effect of air absorption in a hall.

Low-Pass Filter

This parameter controls the overall high frequency content of the selected reverb by allowing you to set the frequency above which a 6 dB/octave filter attenuates (cuts) the processed signal.

Customizing a ReverbThough D-Verb’s seven reverb algorithms provide very useful effects, you may still wish to create your own custom reverbs. You can do this by simply adjusting any of D-Verb’s parameters and saving the set-tings file.

To create a reverb:

1 Choose the type of space you wish to cre-ate by listening to each of D-Verb’s seven algorithms (Hall, Church, Plate, Room 1, Room 2, Ambience, or Nonlinear) and choose the one that sounds closest to what you have in mind.

2 Adjust the Size parameter (Small, Med, or Large) to make the reverb larger or smaller.

3 Adjust the Decay slider to fine-tune the amount of time that it takes for the reverb to die away at the end of a sound.

4 Adjust the Diffusion slider to control how echo density increases over time.

5 Adjust the Pre-delay slider to control the amount of time that elapses before rever-beration takes place.

6 Adjust the Hi Freq Cut slider to set the frequency above which high-frequency sounds decay at a progressively faster rate.

7 Adjust the LP Filter slider to set the fre-quency above which the filter attenuates the processed signal.

Using D-Verb AudioSuite

Selecting Tracks for D-Verb AudioSuite ProcessingTo process a track/region with the D-Verb AudioSuite Plug-In, you must first select the target track/region in either Pro Tools Edit window, or in the Audio Regions list. The D-Verb Plug-In will then process the selected portion of the audio.

However, because D-Verb actually adds ad-ditional material to the end of the selected audio (a reverb tail), it is very important that you make a selection that is actually longer than the original source material so that D-Verb can write it into the audio file.

If you select only the original material, without leaving additional space at the end, a reverb decay that occurs after the end of the region will be cut off.

To accommodate for this, place your region in a track, and select the desired audio plus an amount of blank space at the end of the region equal to the amount of delay or re-verb decay that you have added in the

Plug-In. D-Verb will then have space at the end of the region in which to write the fi-nal decay.

Using D-Verb AudioSuite in StereoThe D-Verb AudioSuite Plug-In can be used in either mono or stereo. If you plan to use it in stereo, there are a few points you should be aware of:

◆ If you wish to process a mono track and obtain a stereo result, select the desired track/region plus an empty track/region, then select the stereo version of D-Verb from the pop-up menu. When you process the audio, the result will be two tracks/re-gions which represent the right and left channels of the processed audio. You should then pan these tracks hard right and hard left in your mix.

◆ If you choose to use the D-Verb Plug-In in stereo mode, and then select an odd number of Pro Tools tracks for processing (as opposed to an even number), D-Verb will process the selected tracks in pairs, in stereo. However, the last odd (unpaired) track will be processed as mono, using the left channel settings of the stereo D-Verb Plug-In. If you wish the last track to be pro-cessed in stereo, you must select an addi-tional track to pair it with — an empty one if necessary.

When you use D-Verb in stereo mode, a Sum Inputs button appears. If you click and enable this button, D-Verb will sum the dry input signals (regardless of whether the input is mono or stereo) before process-ing them. The result is that the dry signal will appear in the center of the stereo field,


50

and the wet, effected signal will be output in stereo.



chapter 7

The Focusrite D2 Plug-In

Focusrite D2 is a high-quality digital equal-izer Plug-In for TDM-capable Pro Tools sys-tems. Developed in cooperation with Fo-cusrite, the D2 is based on the highly- acclaimed Red Range 2™ dual EQ, designed by Rupert Neve.

D2 provides:

◆ Up to six simultaneous bands of EQ, in-cluding: high- pass, low-shelf, low-mid peak, high-mid peak, high-shelf, and low-pass filters

◆ A highly accurate Cartesian graph which displays EQ curves in real-time as EQ pa-rameters are adjusted

◆ 24-bit mono or stereo operation

Focusrite D2

Focusrite D2 ConfigurationsThere are three configurations of the Focus-rite D2 Plug-In, each designed to use a dif-ferent amount of your system’s DSP:

1–2 Band EQ

This module uses the least DSP. It allows you to 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, however, can be used in com-bination with each other (up to two bands total).

4-Band EQ

This module uses more DSP. It allows you to use up to four filters simultaneously. Un-like the 1–2 Band EQ, Any combination of filters can be engaged, up to a total of four bands.

6-Band EQ

This module uses the most DSP. It allows you to use up to six filters simultaneously. By default, the low-pass and high-pass fil-

Chapter 7: The Focusrite D2 Plug-In 51

52

ters are in bypass mode when the 6-Band EQ is first opened.

D2 ParametersThe rotary controls on the D2 Plug-in can be adjusted by clicking and dragging hori-zontally or vertically. Dragging upwards or to the right will increment the control. Dragging downwards or to the left will dec-rement the control.

D2 Input Controls

Drag vertically or horizontally to adjust the D2’s rotary controls

The input section of the D2 Plug-In (mono mode)

Input Level


Input Level

The input level knob allows you to attenu-ate signal input level to the D2. 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 input level knob. If you wish to adjust input lev-els for both channels simultaneously, en-able the Link button.

Output Level

The output level knob 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. If you wish to adjust output lev-els for both channels simultaneously, en-able the Link button.

Output Level

Meters (stereo mode)

Meters

The D2’s high-resolution plasma-style meters indicate signal levels throughout the D2 Plug-In and detect clipping at the input, algorithm, or output stage. When used in stereo, two meters appear, one for each channel.

A red clip indicator appears above the meters. This indicator displays clipping via dynamic brightness level relative to succes-sive samples clipped. Clicking on the clip indicator clears it. Option-clicking it (Mac-intosh) or Alt-clicking it (Windows) will clear both channels when used in stereo.

Metering:

• Green = nominal levels • Yellow = pre-clipping at –6 below full

scale signal• Red = full scale signal/clipping

The Cartesian Graph

The Cartesian Graph displays a visual rep-resentation of the current EQ settings. As you adjust the parameters of any currently active filter, the Cartesian Graph will plot the changes to the EQ curve in real-time. If

Cartesian Graph

you are using D2 in stereo, the Cartesian Graph displays the EQ curve for the right channel in red and the left channel in blue.

✽ Option-clicking (Macintosh) or Alt-clicking (Windows) the Cartesian Graph resets all D2 parameters to their default settings. To reset parameters for both channels when in stereo mode, Option-Shift-click (Macintosh) or Alt-Shift-click (Windows) the Cartesian Graph.

EQ Filter ControlsEach of the six different EQ filters has its own controls and its own icon. The icons act as three-state switches for enabling, dis-abling, or bypassing the specific filter. The current state of a filter is indicated by its color: • White = enabled. In this state the filter is

active, audible, and using available DSP resources.

• Black = disabled. In this state the filter is not using any DSP resources and has no effect on audio.

• 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

High-pass


54

(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 ro-tary controls. The upper rotary control ad-justs the center frequency, variable from 33 Hz to 6.4 kHz. The lower left rotary con-trol adjusts the filter’s amplitude gain or at-tenuation. Amplitude range is ±15 dB from unity (utilizing a reciprocal curve for both gain and attenuation). The lower right ro-tary control adjusts filter “Q”, variable from .7 to 4.

Low-shelf

Low-mid peak


High-mid Peak Filter

The high-mid peak filter provides three ro-tary controls. The upper rotary control ad-justs 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”, vari-able from .7 to 4.

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 atten-uation. Amplitude range is ±15 dB from unity.

High-mid peak

High-shelf

Low-Pass Filter

The 18 dB/octave Low-pass filter provides a rotary control for adjusting the filter’s cor-ner cutoff frequency, variable from 100 Hz to 18 kHz.

Enabling, Disabling and Bypassing EQ Filters■ To disable a filter, Control-click (Macin-tosh) or Start key-click (Windows) its icon. When disabled, the filter icon will appear black.

■ To re-enable a filter, click its icon. When enabled, the filter icon will appear white.

■ To bypass a filter, click its icon a second time. When bypassed, the icon will appear grey.

■ For finer control of a parameter, hold down the Command key (Macintosh) or Control key (Windows) while adjusting it.

▲ If you are using all available bands of the 1–2 Band or 4–Band EQ and wish to change fil-ter types, you must disable one filter before you can enable a different one.

Low-pass

Editing D2 Parameters from the KeyboardEach D2 rotary control has a corresponding numeric entry/display field directly below or beside it. This numeric entry field dis-plays the current value of the parameter. Since adjusting parameter values with a mouse can be cumbersome, you may prefer to input them using your computer key-board.

To edit parameters from the keyboard:

1 Click the displayed value of the parame-ter to highlight it.

2 Type in a new value. You can type “k” for thousands in entries which exceed 1000 Hz. You can increase/decrease the current value by pressing the Up and Down arrow keys respectively.

3 Press Enter to apply the new value and leave the cursor in the current field, allow-ing further editing. Press Return to apply the new value and leave keyboard edit mode.

4 Press Escape to revert the value to its orig-inal setting prior to editing.

5 Press the Tab key to move to the next text-editable parameter, or press Shift-Tab to move to the previous text-editable pa-rameter.


56

Using D2 in Stereo: Left/Right/Link ModeBecause Focusrite D2 has a single set of Fil-ter parameter knobs, when the Plug-In is used in stereo, you must choose which channel, left or right, you wish to edit.

To do this, you will use the Left Channel, Right Channel, and Link buttons. These controls only appear when D2 is used in stereo.

Left Channel and Right Channel Buttons

The Left Channel and Right Channel but-tons are used to select which channel is ac-tive for parameter editing.

Enabling the Left Channel button activates the left channel for editing. Any changes made to EQ parameters will affect the left channel of the stereo signal.

Enabling the Right Channel button acti-vates the right channel for editing. Any

The Left channel, Right channel, and Link buttons


changes made to EQ parameters will affect the right channel of the stereo signal

Link Button

Enabling the Link button allows you adjust parameters for both channels simulta-neously. By default, Link mode is enabled so that you can easily maintain parity be-tween the parameter settings for both channels.

You can also use Link mode to help you maintain a relative offset between parame-ters settings on the two channels.

Maintaining an offset:

1 Disable Link mode.

2 Enable the desired channel, left or right, and adjust the parameters for that channel.

3 Enable Link mode and adjust same pa-rameters for the opposite channel. D2 will maintain the relative offset between the parameter settings for the two channels.

✽ Holding down the Option key (Macintosh) or Alt key (Windows) while linking channels that have offsets will be automatically copy the pa-rameters of the active channel to the opposite channel (including input and output levels).

chapter 8


Focusrite D3 is a high-quality dynamics processor Plug-In for TDM and AudioSuite-capable Pro Tools systems. Developed in cooperation with Focusrite, the D3 is based on the highly acclaimed Red Range 3™ dual mono/stereo compressor & limiter de-signed by Rupert Neve.

Focusrite D3 ConfigurationsThere are two configurations of the real-time TDM Focusrite D3 Plug-In:

◆ Compressor+Limiter. This configuration allows you to use both the compressor and the limiter at the same time. The Compres-sor+ Limiter Plug-In requires twice as much DSP as the Compressor/Limiter.

◆ Compressor/Limiter. This configuration al-lows 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 enabled and the limiter dis-abled.

To enable the limiter:

1 Disable the compressor by Control-click-ing (Macintosh) or Start key-clicking (Win-dows) its icon.

2 Click the limiter icon.


compressor sectioninput section limiter section


58

The D3 CompressorThe D3 compressor reduces the dynamic range of audio signals which exceed a user-selectable threshold by a specific amount. The D3 does this by reducing output levels as input levels increase above the thresh-old.

The amount of output level reduction that D3 applies as input levels increase is re-ferred to as the compression ratio. This pa-rameter is user adjustable. If you set the compression ratio to 2:1, for example, for each 2 dB that the signal exceeds the threshold, the output will increase only by 1 dB. With a compression ratio of 4:1 for example, an 8 dB increase in input will pro-duce only a 2 dB increase in output.

The D3 LimiterThe D3 limiter operates as a fast-attack compressor with a high compression ratio. It doesn’t attack instantaneously or look ahead in order to attack ahead of time, but instead uses a very fast, 1-millisecond at-

The Compressor and Limiter icons

Compressor Limiter


tack time. As such, the D3 is not a brick wall limiter, but limits the overall dynamic range of signals in a useful, sonically-pleas-ing 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.

Side Chain Processing

Both compressors and limiters generally use the detected amplitude of the input sig-nal as a control source. However, other sig-nals can also be used as a control source by using a key input. In the case of de-essing, for example, a frequency-modified version of the input signal is used as a trigger. This is referred to as side chain processing.

The D3’s side-chain processing capabilities allow the D3’s compression or limiting to be controlled by another independent au-dio signal—typically a different Pro Tools track. In this way you can compress or limit one track’s audio using the dynamics of an-other track’s audio.

Using the D3 Plug-In in Stereo

In stereo configurations, all D3 controls other than the Input Level affect both channels of the stereo signal. The D3’s RMS detector (which derives the control signal that drives the dynamics processing) uses a composite of the two channels. Thus, when stereo processing occurs, there is no image shift when signal levels differ be-tween the two channels, since the compos-

ite control signal drives processing for both channels.

D3 ParametersThe rotary controls on the D3 Plug-in can be adjusted by clicking the mouse on the knob and dragging horizontally or verti-cally. Dragging upwards or to the right will increment the control. Dragging down-wards or to the left will decrement the con-trol

The D3 Plug-In

Drag vertically or horizontally to adjust the D3’s rotary controls

compressor sectioninput section limiter section

D3 Input Controls

Input Level

The input level knob allows you to attenu-ate signal input level to the D3 compressor or limiter. The range of this control is from –30 dB to 0 dB.

When you use the D3 Plug-In in stereo, each channel has its own separate input level knob. If you wish to adjust input lev-els for both channels simultaneously, hold the down the Shift key on your computer. Option-Shift clicking (Macintosh) or Alt-Shift-clicking (Windows) either input level knob resets both channels to 0 dB.

The input section of the D3 Plug-In (stereo mode shown)

Input Level


60

Output Level

The output level knob allows you to adjust the overall output gain. Because large amounts of compression can restrict dy-namic range, the output level knob is use-ful for compensating for heavily com-pressed signals and making up the resulting difference in level.

When you use the D3 Plug-In in stereo, 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.

Key On/Off

The key on/off button enables and disables side chain processing. As explained earlier, side chain processing allows the D3 Plug-In to listen to the varying amplitude of a sep-arate reference track and use it to trigger compression or limiting. Chapter 2 of this Guide explains how to set up and use a key input in greater detail.

Output Level

Key On/Off


Key Listen On/Off

The key listen on/off button allows you to listen to (or not listen to) the reference track controlling the side-chain. This is useful in order to fine tune the compres-sor/limiter’s settings to the key input.

Meters

The D3’s meters indicate gain reduction (the top meter) and output level (the bot-tom meter). The gain reduction meter indi-cates 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 ap-pear, one for each channel. A single gain re-duction meter is used for both channels, however, since the D3’s 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 it (Mac-intosh) or Alt-clicking it (Windows) will clear both channels when used in stereo.

Key Listen On/Off

Metering:

• Green = nominal levels • Yellow = pre-clipping at –6 below full

scale signal• Red = full scale signal/clipping

D3 Compressor ControlsThe Compressor icon, which represents a compression curve, acts as a three-state switch for enabling, disabling, or bypassing the compressor. Its current state is indi-cated by its color: • White = enabled. In this state the com-

pressor is active and using available DSP resources.

• Black = disabled. In this state the com-pressor is not using any DSP resources.

• Gray = bypassed. In this state the com-pressor is not active, but is still using available DSP resources.

The compressor section of the D3 Plug-In

Compressor in/out icon

Keyboard Shortcuts


■ To disable either the compressor or lim-iter, Control-click (Macintosh) or Start key-click (Windows) its icon. When disabled, the icon will appear black.

■ To re-enable either the compressor or limiter, click its icon. When enabled, the icon will appear white.

■ To bypass either the compressor or lim-iter, click its icon a second time. When by-passed, the icon will appear grey.

■ If you are using the Compressor/Limiter (which allows you to use only one module at a time), you must disable one module by Control-clicking it (Macintosh) or Start key-clicking it (Windows) before you can enable the other.

Ratio

The ratio knob allows you to set the com-pression ratio. If the ratio is set to 2:1 for example, it will compress changes in sig-nals above the threshold by one half. The range of this control is from 1.5:1 (very lit-tle compression), to 10:1 (heavy compres-sion, bordering on limiting).

Ratio

Threshold


62

Threshold

The threshold knob allows you to set 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 set-ting of 0 dB is equivalent to no compres-sion.

Attack

The attack knob allows you to set the com-pressor 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.

Release

The release knob allows you to control how long it takes for the compressor to be fully deactivated after the input signal drops be-low the threshold level. In general, this set-ting should be longer than the attack time and long enough that if signal levels re-peatedly rise above the threshold, they cause gain reduction only once. If the re-lease time is too long, a loud section of the

Attack

Release


audio material could cause gain reduction that persists through a soft section. The range of this control is from 25 millisec-onds to 2.5 seconds

Auto

The auto release button enables the D3’s automatic release function. In this mode the release knob’s setting has no effect on the compressor’s release time. Instead, the D3 uses a release time value that is program dependent, and based on the audio signal being processed.

D3 Limiter ControlsThe Limiter icon, which represents a lim-iter curve, acts as a three-state switch for enabling, disabling, or bypassing the lim-iter. Its current state is indicated by its color: • White = enabled. In this state the limiter

is active and using available DSP re-sources.

• Black = disabled. In this state the limiter is not using any DSP resources.

• Gray = bypassed. In this state the limiter is not active, but is still using available DSP resources.

Keyboard Shortcuts


Auto

■ To disable either the compressor or lim-iter, Control-click (Macintosh) or Start key-click (Windows) its icon. When disabled, it will appear black.

■ To re-enable either the compressor or limiter, click its icon. When enabled, it will appear white.

■ To bypass either the compressor or lim-iter, click its icon a second time. When by-passed, it will appear grey.

■ If you are using the Compressor/Limiter (which allows you to use only one module at a time), you must disable one module by Control-clicking it (Macintosh) or Start key-clicking it (Windows) before you can enable the other.

Limit LED

When lit, the limit LED indicates that lim-iting is currently taking place. When unlit, it indicates that no limiting is occurring.

The limiter section of the D3 Plug-In

The Limiter in/out icon

Limit LED

Threshold

This control allows you to set the threshold level. Signals that exceed this level will be limited. Signals that are below it will be un-affected. A level setting of 0 dB is equiva-lent to no limiting. The range of this con-trol is from –24 dB to 0 dB. With the threshold set to 0 dB, there will be no lim-iting.

Threshold


64

chapter 9

The DINR Plug-Ins

Digidesign Intelligent Noise Reduction (DINR) consists of a pair of DSP Plug-Ins which provide two types of noise removal:

Broadband Noise Reduction (BNR) The BNR Plug-In provides broad and narrow-band noise reduction for suppressing such un-wanted elements as tape hiss, air condi-tioner rumble, and microphone preamp noise. BNR is available as a real-time TDM and as an AudioSuite Plug-In.

Hum Removal The Hum Removal Plug-In removes pitch-based noise such as guitar pickup buzz, AC line noise, and fluorescent hum. Hum Removal is available as a real-time TDM Plug-In only.

▲ The Hum Removal Plug-In is not compatible with Pro Tools 24 MIX cards. It requires a non-MIX DSP Farm to operate.

Broadband Noise ReductionThe Broadband Noise Reduction module (BNR) removes many types of broadband and narrow-band noise from audio mate-rial. It is best suited to reducing noise whose overall character doesn’t change very much: tape hiss, air conditioner rum-

ble, and microphone preamp noise. In cases where recorded material contains sev-eral types of noise, the audio can be pro-cessed repeatedly according to the specific types of noise.

For noise that has more of a pitched qual-ity, the Hum Removal module may yield more satisfactory results. Hum Removal is covered later in this chapter.

How Broadband Noise Reduction WorksThe Broadband Noise Reduction module uses a proprietary technique called Dy-namic Audio Signal Modeling™ to intelli-gently subtract the noise from the digital audio file. Noise is removed with multiple downward expanders that linearly decrease the gain of a signal as its level falls.

The BNR Plug-In

Chapter 9: The DINR Plug-Ins 65

66

Creating a Noise Signature

The first step in performing broadband noise reduction is to create what is called a noise signature by selecting and analyzing an example of the noise within the source material. Using this noise signature, a noise contour line is created which is used to de-fine the thresholds for the downward ex-panders that will perform the broadband noise reduction. The noise contour repre-sents an editable division between the noise and non-noise audio signals.

At the same time, DINR also creates a model of what the non-noise audio signal looks like. DINR then attempts to pull apart these two models, separating the bad from the good—the noise from the desired au-dio. The noise portion can then be reduced or eliminated.

The noise reduction itself is achieved through the use of multiple downward ex-panders. The threshold of these expanders is set so that the noise signal will fall below them and be decreased while the desired audio signal will remain above them, un-touched.

The Contour Line

Once the signal level has fallen below the specified contour line (which represents BNR’s threshold), the downward expanders are activated and decrease the gain of the signal as its level falls. Over five hundred individual downward expanders are used linearly across the audio spectrum to re-duce the effects of unwanted noise.


Psychoacoustic Effects of Noise Reduction

One of the psychoacoustic effects associ-ated with broadband noise reduction is that listeners often perceive the loss of noise as a loss of high frequencies. This oc-curs because the noise in the higher fre-quency ranges fools the ear into thinking the original signal has a great deal of en-ergy in that range. Consequently, when the noise is removed it feels as if there has been a loss of high frequency signal. DINR’s high-shelf EQ is useful for compen-sating for this effect. The high-shelf EQ is explained later in this chapter.

Limitations of Noise Reduction

It is important to understand that there is a certain amount of trade-off inherent in any type of noise reduction system. Imple-menting noise reduction means that you have to choose the best balance between the following three things:

◆ The amount of noise removed from the signal

◆ The amount of signal removed from the signal

◆ The amount of artifacts added to the sig-nal

DINR gives you a considerable amount of control over the above three parameters and allows you to maximize noise reduc-tion while minimizing signal loss and arti-fact generation. However, as powerful as it is, DINR does have limitations. In particu-lar, there are two instances in which DINR may not yield significant results:

◆ Cases in which the noise components of the audio are so prominent that they ob-

scure the actual signal components of the audio.

◆ Cases in which the noise amplitude of a 24-bit file is less than –96 dB. DINR is not designed to recognize noise that is lower than this level.

Broadband Noise Reduction ParametersThe following section describes the Broad-band Noise Reduction parameters and their use.

The Spectral Graph The Spectral Graph dis-plays the noise signature and the editable noise Contour Line. The Spectral Graph’s horizontal axis shows frequency, which is displayed in Hertz, from 0 Hz to one-half the current audio file’s sample rate. The Spectral Graph’s vertical axis shows ampli-tude, which is displayed in dB, from 0 dB to –144 dB (below full-scale output of the au-dio).

The Noise Signature The jagged line is a graph of noise. This is called a noise signa-ture. It is created when you use the Learn button in the Broadband Noise Reduction window. Once you have the noise signa-ture of an audio file, you will be able to be-gin removing the noise by generating and editing a threshold or Contour Line (covered

The Spectral Graph

next) between the noise and the desired au-dio signal.

The Contour Line The line with a series of square breakpoints is called the noise Con-tour Line. The Contour Line is an editable envelope which represents the division be-tween the noise and the non-noise signal in the current audio file. The Contour Line is created by clicking the Fit or AutoFit but-ton in the Broadband Noise Reduction window after you have learned a section of noise. By moving this envelope up or down, or by moving the individual break-points, you can modify which signals are removed and which remain.

The noise modeling process treats audio below the line as mostly noise, and audio above the line as mostly signal. Therefore, the higher your move the Contour Line up-wards, the more audio is removed. To max-imize noise reduction and minimize signal loss, the Contour Line should be above any noise components, but below any signal components.

To fine tune the broadband noise reduc-tion, move breakpoints at different loca-tions along this line to find out which seg-ments remove the noise most efficiently. Editing the Contour Line to follow the noise signature as closely as possible will also help maximize noise reduction and minimize signal loss. Editing the Contour

The Contour Line


68

Line is explained in greater detail later in this chapter.

Noise Reduction Amount The NR Amount slider controls how much the noise signal is reduced. It is calibrated in decibels. A set-ting of 0 dB specifies no noise reduction. Increasing negative amounts specify more noise reduction. The default value is 0 dB.

In many cases, as much as 20–30 dB of noise reduction can be used to good effect. However, because higher amounts of noise reduction can generate unwanted audio ar-tifacts, you may wish to avoid setting the NR amount slider to its maximum value.

Response The Response slider adjusts how quickly the downward expanders/noise re-duction process responds to the overall changes in the noise in milliseconds. De-pending on the character of the noise, dif-ferent settings of this control will produce varying amounts of artifacts in the signal, as the modeling process attempts to track the noise signal faster or slower.

The Response speed ranges from 0 ms to 116 ms. A setting of 116 ms (slow) specifies that the modeling process should not at-tempt to track very fast changes in the noise character. A setting of 0 ms (fast) specifies that the modeling process should attempt to follow every change in the noise character very closely.

The NR amount, Response, Release, and Smoothing sliders


A faster setting can yield more noise re-moval, but it may generate more artifacts. This is similar to how a noise gate produces chatter when attempting to track highly dynamic material. A slower setting will al-low slightly less noise removal, but will generate much fewer artifacts.

Release The Release slider is used in con-junction with the Response slider. It con-trols how quickly DINR reduces the amount of noise reduction when the amount of noise present in the audio di-minishes. Release times range from 0 ms to 116 ms. Like the Response control, a faster setting can yield more noise removal, but it may also generate artifacts. You may wish to avoid setting this control to its slowest position, since it will cause the noise track-ing to slow to the point that the other pa-rameters seem to have no effect.

Smoothing The Smoothing slider controls the rate at which noise reduction occurs once the threshold is crossed. It allows you to reduce the audibility of any artifacts generated in the modeling process, at the expense of noise reduction accuracy. This is done by limiting the rate of change of the Response and Release controls to the specified Smoothing setting. As soon as the frequency threshold is reached, the full NR amount value is immediately applied ac-cording to Response and Release settings. When the frequency threshold is reached, DINR will ramp to the NR amount level. Settings range from 0 to 100%. A setting of 0% specifies no smoothing. A setting of 100% specifies maximum smoothing.

Hi Shelf EQ

The Hi Shelf filter is a noiseless filter that can be applied after noise reduction has been performed in order to compensate for a perceived loss of high frequency content. It is unique because it operates only on the signal, not on any remaining noise. The Freq slider controls the center frequency of the filter. Values range from 20 Hz to 22 kHz.

The gain slider controls the gain of the fil-ter. Values range from –12 dB to +6 dB. The Hi Shelf filter can be enabled/disabled by clicking the Enable button.

You can also use the High Shelf filter to re-duce the amount of high frequencies in a signal. This is particularly useful if you are working with older recordings that are band-limited, since the high frequency content in these is probably made up of noise and not signal.

Learn

Clicking the Learn button creates a noise signature based on the audio segment cur-rently selected on screen. There are two Learn modes: Learn First Audio Mode and Learn Last Audio Mode.

Hi Shelf EQ

Learn

Learn First Audio Mode Learn First Audio Mode is the default Learn mode. It is de-signed for use with audio that has an iden-tifiable noise-only section that you can lo-cate and pre-select. To use this mode, locate and select the noise-only portion of the audio, click the Learn button, start playback, and BNR will build a noise signa-ture based on the first 16 milliseconds of audio playback. First Audio Learn Mode can be thought of as a trigger-learn mode, since noise capturing is triggered by the first audio that DINR receives.

Learn Last Audio Mode Learn Last Audio Mode is designed to allow you to locate and identify a segment of noise on-the-fly as you listen to audio playback. In this mode, you first Option-click (Macintosh) or Alt-click (Windows) the Learn button, then initiate audio playback. When you hear the portion of audio that contains the noise you wish to identify and remove, click the Learn button a second time. BNR will build a noise signature based on the last 16 milliseconds of audio playback. The Spectral Graph displays data in real-time in Learn Last Audio Mode.

Fit

The fit button computes a noise Contour Line with approximately 30 breakpoints to fit the shape of the current noise signature. The Contour Line can then be edited to more closely fit the noise signature or to re-duce specific frequency bands by dragging, adding or deleting breakpoints.

Fit


70

Pressing the Up Arrow or Down Arrow keys on your computer keyboard allows you to raise or lower the entire Contour Line, or a selected portion of the Contour Line. The Left/Right arrows allow you to move a se-lection left/right. To select a portion of the Contour Line with multiple breakpoints, hold down the Command key (Macintosh) or Control key (Windows) and click and drag the cursor to highlight the desired area.

After you use the fit function, BNR will au-tomatically boost the entire Contour Line 6 dB above the noise signature so that all noise components of the audio file are be-low the Contour Line. You may wish to ad-just the Contour Line downwards as needed to modify the character of the noise reduction.

Super Fit

The super fit button creates a noise Con-tour Line consisting of over five hundred breakpoints in order to follow the shape of the noise signature more precisely.

Auto Fit

The auto fit function is designed to gener-ate a noise curve for audio that lacks a noise-only portion for DINR to learn. Clicking auto fit computes this generic noise curve based on the points contained within the currently selected audio, then

Super Fit

Auto Fit


fits the Contour Line to it. To use the auto fit function, you must first make a selection in the Spectral Graph by holding down the Command key (Macintosh) or Control key (Windows), then clicking and dragging the mouse.

If the selected audio has both noise and de-sired sound components, you can generate an approximate noise-only Contour Line by selecting a frequency range that appears to be mostly noise, then pressing the auto fit button. You can then edit the resulting noise Contour Line to optimize the noise reduction.

Scroll Left/Right

These buttons scroll the Spectral Graph to the left or right, respectively. The left ar-row/right arrow keys of your computer key-board will also scroll the Spectral Graph.

Zoom Out/In

Clicking on these buttons zooms in or out of the Spectral Graph. This allows you to view and edit the noise contour with greater precision. If you have selected a breakpoint or breakpoints, press Control-Option-Plus (Macintosh) or Alt-Start key-Plus (Windows) to zoom the beginning of the selection to the center of the screen. Press Control-Option-Minus (Macintosh) or Alt-Start key-Minus (Windows) to zoom back out.

Scroll Left/Right

Zoom Out/In

Move Breakpoints Up/Down/Left/Right

These arrows behave differently depending on whether or not there is a selection of points along the Contour Line.

No Selection: When there is no selection, the up and down arrows move the entire Contour Line up/down by 1 dB, respec-tively, and the Left and Right arrows scroll the display left and right.

With a Selection: Clicking these buttons moves a selected breakpoint or breakpoints up, down, right, or left. If there is currently a selection in the Spectral Graph, clicking the left and right arrow buttons will move the selected breakpoints left or right. The up and down arrows will move the selected breakpoints up or down, respectively. Holding down Control-Option (Macin-tosh) or Alt-Start key (Windows) and press-ing the arrow keys on your computer key-board performs the same function.

Undo

Clicking the Undo button undoes the last edit to the Spectral Graph Display. The Undo button does not undo changes made to slider positions.

Move Breakpoints Up/Down/Left/Right

Undo

Using Broadband Noise ReductionBefore you start using BNR, take a moment to think about the nature of the noise in your session and where it’s located: Is it on a single track, or several tracks? Is it a single type of noise, or several different types? The answers to these questions will affect how you use BNR.

If there is a single type of broadband noise on a single track, simply insert the BNR Plug-In onto the offending track. We sug-gest that you solo the track to make it easier hear as you remove the noise. If a single track contains different types of noise, you may need to use more than one DINR in-sert to remove the other types of noise. If multiple tracks contain the same noise, you may wish to bus them all to an auxil-iary input so you can use a single DINR Plug-In insert. This will minimize the amount of DSP you use.

▲ The Macintosh version of the BNR Audio-Suite Plug-In is designed for use on Power Macintosh computers only.

To use Broadband Noise Reduction:

1 Click the Inserts pop-up on the track with the noise and select BNR. The Broad-band Noise Reduction window appears.

2 In the Pro Tools Edit window, select the noisiest portion of the track—ideally, a seg-ment with as little of the desired signal as possible. This will make it easier for BNR to accurately model the noise. If the track contains a segment comprised of noise only, select that portion.


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3 Start audio playback, and in the Broad-band Noise Reduction window, click the Learn button. BNR samples the first 16 mil-liseconds of the selected audio and creates its noise signature.

4 Alternatively, you can conveniently lo-cate and identify noise on the fly, during playback, using BNR’s Learn Last Audio Mode. To do this, Option-click (Macintosh) or Alt-click (Windows) the Learn button. Begin playback, and when you hear the segment that you wish DINR to sample as noise, click Learn a second time. BNR will build a noise signature based on the 16 mil-liseconds of audio immediately preceding the second click.

5 Click the Fit button. BNR will fit a Con-tour Line to the noise signature just cre-ated. If you wish to create a Contour Line that follows the noise signature even more precisely, click the Super Fit button. A Con-tour Line with five hundred breakpoints is created.

6 To audition the effects of the noise reduc-tion interactively, in the Edit window, se-lect a portion of audio containing the noise. Then select Loop Playback from Pro Tools’ Options menu and press the Space bar to begin looped audio playback.

7 Adjust the NR amount slider to reduce the noise by the desired amount. To com-pare the audio with and without noise re-duction, click the Bypass button.

8 To fine tune the effects of the noise re-duction, adjust the Response, Release, and Smoothing sliders to achieve optimal re-sults.

9 To further increase noise reduction, edit the Contour Line. The quickest way to do this is to move the entire Contour Line up-


wards. In the Spectral Graph, hold down the Command key and select the entire waveform range. Then click the Move Breakpoint Up button. The higher you move the Contour Line above the noise signature, the more noise is removed. For more detailed instructions on this tech-nique, refer to the section entitled, Editing the Contour Line, later in this chapter.

10 If you feel that some of high end fre-quencies of the audio have been lost due to the noise reduction process, try using the Hi Shelf EQ to compensate. To do this, click BNR’s Hi Shelf button and adjust the fre-quency and gain sliders until you are satis-fied with the results.

11 If you are happy with the results of the noise reduction, use the Settings and Li-brarian menus to save the settings so that you can use them again in similar sessions.

Keyboard Shortcuts

■ To enable Learn Last Audio mode, Op-tion-click (Macintosh) or Alt-click (Win-dows) the Learn button. This button flashes red when armed for Learn Last Au-dio mode. When you hear the target noise, click Learn a second time.

Performing Noise Reduction on Audio that Lacks a Noise-Only PortionIdeally, audio that you wish to perform noise reduction on will have a noise-only portion at the beginning or end of the re-cording which DINR can analyze and learn. Unfortunately this is not always the case, and in many recordings some amount of signal is always mixed with the noise. Ob-viously, analyzing such audio will produce

a noise signature that is based partially on signal. Luckily, DINR has provisions for cases such as this, and this is where the Auto Fit feature comes in.

If your audio file lacks a noise-only portion for DINR to analyze, you can still obtain reasonable results by selecting and learning a segment of audio which has a relatively low amount of signal and a high amount of noise (as in a quiet passage). By then select-ing a frequency range of the noise signa-ture and using the Auto Fit function to gen-erate a generic noise curve, you can recompute the Contour Line based on this selection.

Some editing of the newly generated Con-tour Line will probably be necessary to yield optimum results, since is not based entirely on noise from your audio file. You’ll learn more about how to edit the Contour Line in the section entitled, Edit-ing the Contour Line, later in this chapter.

To generate a Contour Line for audio that lacks a noise-only portion:

1 In the Edit window, select a segment of audio with a relatively low amount of sig-nal and a high amount of noise.

2 Click the Inserts pop-up on the track with the noise and select BNR. The Broad-band Noise Reduction window appears.

3 Click the Learn button to create a prelim-inary noise signature.

4 Click the Fit button to fit a Contour Line to it.

5 In BNR’s Spectral Graph, hold down the Command key (Macintosh) or Control key (Windows) and make a selection. Select points where the high-frequency noise components are most evident. In general,

the flatter areas of the Spectral Graph, are better, since they represent quieter areas where there is probably less signal and more noise. See the illustration below.

6 Click the Auto Fit button. DINR com-putes a generic noise curve and corre-sponding Contour Line based on your selection. If you wish to remove the selec-tion in the Spectral Graph Display, hold down the Command key (Macintosh) or Control key (Windows) and click once.

7 Follow the steps given in the previous section removing the noise using the NR Amount slider and other controls.

8 Since the Contour Line is not based en-tirely on noise from your audio file, you may also wish to edit its envelope in order to fine-tune the noise reduction. Please re-fer to the section entitled, Editing the Con-tour Line, later in this chapter for details on how to do this.

Editing the Contour LineOne of the most effective ways to fine tune the effects of broadband noise reduction is to edit the Contour Line. The Contour Line treats audio below the line as mostly noise, and audio above the line as mostly signal. Therefore, the higher your move the Con-tour Line upwards, the more audio is re-moved.

Noise components are more evident in flatter areas of the Spectral Graph


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To maximize noise reduction and mini-mize signal loss, the Contour Line should be above any noise components, but below any signal components. To fine tune the broadband noise reduction, try moving in-dividual breakpoints at different locations along this line to find out which segments remove the noise most efficiently. For more dramatic results, try moving the en-tire Contour Line upwards. One drawback of the latter technique is that it will typi-cally remove a considerable amount of sig-nal along with the noise.

Remember that high-frequency noise com-ponents are typically more evident in the flatter, lower amplitude areas of the Spec-tral Graph. Try editing the Contour Line in these areas first.

Before you Begin

To hear the effects of the changes you make to the Contour Line in real time, do the fol-lowing:

1 Select the target audio in Pro Tools’ Edit window. Make sure the selection is at least a second or two in length. If the selection is too short, you won’t be able to loop play-back.

2 Select Options > Loop Playback.

3 Press the Space bar to begin playback.

To edit the Contour Line:

1 To move a breakpoint, click directly on it and drag it to the desired position. Moving a breakpoint higher increases noise reduc-tion at that range. Moving a breakpoint lower decreases noise reduction at that range.


2 To move multiple breakpoints, hold down the Command key, then click and drag to select the desired breakpoints. Click the appropriate Move Breakpoint button (below the Spectral Graph) to move the se-lected breakpoints in 1 dB increments. Command-Shift-drag (Macintosh) or Con-trol-Shift-drag (Windows) to extend your selection.

3 To move the entire Contour Line, hold down the Command key, then select the entire range. Click the appropriate Move Breakpoint button (below the Spectral Graph) to move the selected breakpoints in 1 dB increments. The higher you move the Contour Line above the noise signature, the more noise is removed.

Moving a breakpoint by dragging it

Moving selected breakpoints up, down, left or right

4 To create a new breakpoint, click on the Contour Line.

5 To delete a breakpoint, Option-click (Macintosh) or Alt-click (Windows) the breakpoint. As long as the mouse button is held down, you will delete all breakpoints that the cursor passes over.

Using BNR AudioSuiteBNR AudioSuite is identical to the real-time version of BNR, with the addition of two new features to enhance the noise reduc-tion process:

Audition Noise Allows you to listen specifi-cally to the noise portion being removed from the target material. This makes it eas-ier to fine tune noise reduction settings to maximize noise reduction and minimize signal loss.

Artifact Removal Applies post-processing to the audio file to help remove undesirable artifacts that are a result of noise reduction.

To enable either of these features, click the corresponding button. To disable them, click again.

BNR AudioSuite

BNR AudioSuite Features

To process a region with the BNR AudioSuite Plug-in:

6 Select the desired regions in the target tracks and/or in the Audio Regions List. Only tracks/regions that are selected will be processed.

7 From the Pro Tools AudioSuite menu, choose BNR.

8 Click Learn to capture the noise signa-ture of the selected material. If you have se-lected more than one track/region, BNR will build the noise signature based on the first selected track/region when used in mono mode, or the first two selected tracks/regions when used in stereo mode.

9 Click Fit or Super Fit to create a Contour Line that matches the noise signature.

10 Click Preview to begin playback of the selected material.

11 Adjust BNR parameters and fine tune the noise reduction using the techniques explained earlier in this chapter.

12 To hear the noise components that are being removed, click Audition Noise. Ad-justing BNR’s parameters while toggling this on and off will allow you to fine tune the noise reduction. It also allows you to hear exactly how much signal is being re-moved with the noise, and adjust your pa-rameters accordingly.

13 If unwanted artifacts are generated by the noise reduction process, click Artifact Removal. For best results, set the Response and Release parameters to zero.

To begin AudioSuite processing:

1 Adjust the AudioSuite file parameters. These settings will determine how the file is processed and what effect the processing


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will have on the original regions. Here are some guidelines:

◆ To process the selected region only in the track in which it appears, choose Playlist from the Selection Reference pop-up.

– or –

To process the selected region in the Audio Regions list only, choose Region List from this pop-up.

◆ To process and update every occurrence of the selected region throughout your ses-sion, enable the Use In Playlist button (and also choose Region List from the Selection Reference pop-up).

– or –

If you do not wish to update every occur-rence of the selected region, disable the Use In Playlist button.

◆ If you have selected multiple regions for processing and wish to create a new file that connects and consolidates all of these regions together, choose Create Continu-ous File from the File Mode pop-up menu.

▲ BNR AudioSuite does not allow destructive processing, so the Overwrite Files option is not available in the File Mode pop-up menu.

2 From the Destination Track pop-up, choose the destination for the replacement audio.

3 Click Process.

Hum RemovalDINR’s Hum Removal module is designed to remove many types of harmonic noise from audio material. It is especially effec-tive on noise such as AC line hum, light


dimmer buzz, generator noise, guitar pickup noise, fluorescent hum, passing car drones, and noise from computer monitor emissions. The Hum Removal Plug-In does this by utilizing a 200-notch comb filter to remove the individual frequency compo-nents—fundamental and associated har-monics—from the audio material.

How Hum Removal WorksThe Hum Removal module provides har-monic filters specifically designed to re-move harmonic pitched noise. Harmonic noise is noise that not only contains a fun-damental frequency component, as in the 60 Hz component of AC line noise, but also contains significant components at the odd and/or even harmonics of the funda-mental frequency. If you were to remove only the 60 Hz component of AC line noise, you would still hear significant amounts of noise at 120 Hz (the second harmonic), 180 Hz (the third harmonic), and so on.

Harmonic filters will remove not only the fundamental frequency they are set to, but also all odd and even harmonics of that fre-quency. The advantage of this type of filter is that it effectively creates hundreds of dig-ital notch filters at exactly the frequencies needed to remove certain types of noise.

However, while a harmonic filter is ex-tremely efficient, it also has drawbacks. The disadvantage of this type of filter is that it can alter the sound by imparting a ringing or phase canceling quality to the sound. The harmonic filters used in the Hum Re-moval module give you the ability to choose the best balance between noise re-

duction and the amount of undesired sound alteration that occurs.

Hum Removal ParametersThe following section describes the Hum Removal parameters and their use.

Center Frequency The Center Frequency slider sets the fundamental frequency of the Harmonic filter. Though the Learn function automatically sets the center fre-quency to the fundamental frequency of the learned noise, this parameter can also be manually edited. The other notches in the harmonic filter are automatically set to the related harmonic frequencies.

Cut Level The Cut Level slider determines the depth of the notches within the Har-monic Filter. This allows you to specify the amount of hum removal that takes place when the signal falls below the threshold. Higher dB settings result in deeper notch filters and more noise reduction. Lower dB settings produce fewer undesired alter-

The Hum Removal Plug-In

Center Frequency

ations of the audio. A cut level of 0 dB ap-plies no harmonic filtering. A cut level of –∞ applies maximum hum removal.

Threshold The Threshold slider sets the sig-nal level at which the harmonic filtering occurs. If the input signal level is below the threshold, the harmonic filter is activated and hum is reduced. When the signal level rises above the threshold, the harmonic fil-ter is bypassed, since loud signals psychoa-coustically mask the audibility of noise components.

Setting the threshold to 0 dB (its highest level) will cause the harmonic filter to al-ways be active. For optimum hum removal and a minimum of artifact generation, ad-just the Threshold so that it is just above the noise level.

Attack The Attack slider controls how long it takes the Harmonic Filter to reach its full effect once the signal has fallen below the threshold you have set. The default attack value is 15 ms.

Release The Release slider controls how long it takes the Harmonic Filter to switch off once the signal has increased above the threshold (determined by your Threshold setting). The default release value is 15 ms.

Learn The Learn button analyzes the se-lected audio and automatically sets the center frequencies of the filters for maxi-mum hum removal. The Learn function saves you the trouble of manually sweep-ing through the frequency controls in or-der to find the appropriate fundamental center frequency for hum removal. There are two Learn modes: Learn First Audio Mode and Learn Last Audio Mode.


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Learn First Audio Mode Learn First Audio Mode is the default Learn mode. It is de-signed for use with audio that has an iden-tifiable noise-only section that you can lo-cate and pre-select. To use this mode, locate and select the noise-only portion of the audio, click the Learn button, start Pro Tools playback, and the Hum Removal Plug-In will set an appropriate fundamen-tal center frequency for hum removal based on the first 16 milliseconds of audio playback. First Audio Learn Mode can be thought of as a trigger-learn mode, since noise capturing is triggered by the first au-dio that DINR receives.

Learn Last Audio Mode Learn Last Audio Mode is activated by Option-clicking (Mac-intosh) or Alt-clicking (Windows) the Learn button. It is designed to allow you to locate and identify a segment of noise on-the-fly as you listen to audio playback. In this mode, you first Option-click (Macin-tosh) or Alt-click (Windows) the Learn but-ton, then initiate audio playback. When you hear the portion of audio that contains the noise you wish to identify and remove, click the Learn button a second time. The Hum Removal Plug-In will set an appropri-ate fundamental center frequency for hum removal based on the last 16 milliseconds of audio playback.

In order for Hum Removal to learn the au-dio signal from live inputs, Pro Tools must be in playback mode.

Signal Present LED The signal present LED is lit when an audio signal is received.


Using Hum RemovalAs in the BNR module, you must have Hum Removal learn the target noise in an audio track. To learn the noise, the Hum Removal module sweeps through the material’s fre-quency spectrum, scanning the noise to find the most prominent frequency charac-teristic of the audio signal. When this is completed, it automatically configures it-self for maximum hum removal by setting its center frequency parameter to the fun-damental frequency of the most prominent hum.

Before You Begin

Before you start using Hum Removal, think about the nature of the noise and where it’s located: Is it on a single track, or several tracks? Is it a single type of noise, or several different types? The answers to these ques-tions will affect how you use Hum Re-moval.

If there is a single type of hum on a single track, simply insert the Hum Removal Plug-In onto the offending track. We sug-gest that you solo the track to make it easier hear as you remove the noise. If a single track contains different types of noise, you may need to use more than one DINR in-sert to remove the other types) of noise. If multiple tracks contain the same noise, you may wish to bus them all to an auxil-iary input so you can use a single DINR Plug-In insert. This will minimize the amount of DSP you use.

To use Hum Removal:

1 Click the Inserts pop-up on the track with the noise and select Hum Removal. The Hum Removal window appears.

2 In the Edit window, select the portion of the track where the hum is strongest—ide-ally, a segment with as little of the desired signal as possible. This will make it easier for the Hum Removal module to accurately analyze the hum. If the track contains a segment comprised of hum only, select that portion.

3 Begin audio playback, and in the Hum Removal window, click the Learn button. The Plug-In samples the first 50 millisec-onds of the selected audio and searches for the most prominent hum frequency.

4 Alternatively, you can conveniently lo-cate and identify hum on the fly, during playback, using the Hum Removal mod-ule’s Learn Last Audio Mode. To do this, Option-click (Macintosh) or Alt-click (Win-dows) the Learn button. Begin playback, and when you hear the hum that you wish to identify, click Learn a second time. The Plug-In will identify the hum based on the 50 milliseconds of audio immediately pre-ceding the second click.

5 Adjust the Cut Level slider to in-crease/decrease the degree to which the hum frequency is cut. Although the Hum Removal module automatically sets the Cut Level to maximum hum removal, you may still wish to adjust this parameter to minimize any undesired side-effects. To compare the audio with and without noise reduction, click the Bypass button.

6 To fine tune the effects of the hum re-moval, try adjusting the Threshold, Attack, and Release sliders.


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chapter 10

The DPP-1 Plug-In

DPP-1 is a real-time TDM pitch processing Plug-In designed for a wide range of audio production applications ranging from pitch correction of musical material to full-blown sound design.

DPP-1 provides:• Up to 4 octaves of chromatic, equal-tem-

perament pitch transposition• Up to 125 milliseconds of delay per

channel• 24-bit mono or stereo operation

About Pitch ProcessingDigital pitch processing is an important component of modern audio production. It has a wide range of useful applications, including tuning correction of musical ma-terial, general thickening of specific audio elements in a mix (typically broadening the element’s presence in the stereo field), and special effects applications involving extreme pitch transposition of audio mate-rial.

How DPP-1 Works

In its simplest form, pitch processing uses the technique of varying sample playback rate to achieve pitch transposition. Because changing audio sample playback rate re-

sults in the digital equivalent of vari-speed-ing with tape, this is an unsatisfactory method since it changes the overall dura-tion of the material.

Pitch transposition with DPP-1 involves a much more complex technique: digitally adding or subtracting portions of the audio waveform itself, while using de-glitching crossfades to minimize undesirable arti-facts. The result is a processed signal that is transposed in pitch, but still retains the same overall length as the original, unproc-essed signal.

DPP-1 ParametersInput Level This slider allows you to reduce or attenuate the input volume of DPP-1 to help prevent the possibility of internal clip-ping. This parameter ranges from 0 (no at-tenuation) to full off (infinity). Because Pro Tools Inserts are pre-fader, clipping can oc-cur if gain is boosted too high. This type of clipping can’t be rectified by adjusting the Input Level attenuator since this control af-fects only internal pitch processing levels.

Signal Present Indicator (green LED) This meter indicates the presence of an input signal.

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Clip Indicator (red LED) This indicator shows if clipping has occurred. 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 with the mouse. With longer delay times and greater feed-back settings, there is an increased likeli-hood of clipping as the feedback element builds up and approaches a high output level.

Be careful to monitor your signal levels closely to prevent this from happening. It is important to note that the Clip Indicator is based on the output level of DPP-1—not the input level. It is possible for a signal to actually clip on input before it reaches DPP-1. In this case, DPP-1 has no way of in-dicating or controlling clipping that may have occurred in the signal chain prior to its input.

✽ During mixdown, using the DPP-1 in a stan-dard send/return arrangement, you may wish to use a Master Fader assigned to your send buss so that you can monitor input signal lev-els and watch for clipping.

Mix This slider allows you to adjust the ra-tio of dry signal to effected signal in the output. In general, this control should be set to 100% wet, unless you are using DPP-1 in-line on an Insert for an individual track or element in a mix. This control can be adjusted over its entire range with little or no change in output level.

Delay This slider allows you to set the delay time between the original signal and the pitch-shifted signal. It has a maximum set-ting of 125 milliseconds. You can use the delay slider in conjunction with the Feed-back control to generate a single pitch-


shifted echo, or a series of echoes which climb in pitch.

Feedback This slider allows you to control the amount and type (positive or negative) of feedback applied from the output of the delay portion of DPP-1 back into its input. It also controls the number of repetitions of the delayed signal. It allows for effects which spiral up or down in pitch, with each successive echo shifted in pitch.

Coarse This slider allows you to adjust the pitch of a signal in semitones over a two oc-tave range. Pitch changes are indicated both in the Semitones field and in the Mu-sical Staff Relative section below this slider. Using the –8va and +8va buttons in con-junction with the Coarse slider provides a full 4-octave range of adjustment.

–8va and +8va Buttons Clicking the –8va button adjusts pitch down one octave from the current setting of the coarse and fine pitch controls. Clicking the +8va button adjusts pitch up one octave from the cur-rent setting of the coarse and fine pitch controls.

Relative Pitch Entry (Musical Staff) Click-ing on any note on this ascending musical staff allows you to quickly select a relative pitch transposition value that will be ap-plied to an audio signal. If Middle C is illu-minated, it indicates that no pitch transpo-sition has been selected. If a pitch transposition is selected, the note interval corresponding to the selected transposition value is indicated in yellow. Option-click-ing (Macintosh) or Alt-clicking (Windows) on the staff will set the coarse pitch change value to zero.

Fine This slider allows you to adjust the pitch of a signal in cents (hundredths of a semitone) over a 100 cent range. The range of this slider is –49 to +50 cents. Precise Pitch change values are indicated in the Cents field. The flat, natural, and sharp signs below this slider indicate deviation from the nearest semitone.

Ratio This indicator shows the ratio of transposition between the original pitch and the selected transposition value.

Crossfade This slider allows you to manu-ally adjust the crossfade length in millisec-onds to optimize performance of DPP-1 ac-cording to the type of audio material you are processing. DPP-1 performs pitch trans-position by replicating or subtracting por-tions of audio material and very quickly crossfading between these alterations in the waveform of the audio material.

Crossfade length essentially affects the amount of smoothing performed on audio material to prevent audio artifacts such as clicks as the audio is looped to generate the desired pitch shift. In general, small, nar-row-range pitch shifts require longer cross-fades and large shifts require smaller ones. The disadvantage of long crossfade times is that they smooth the signal, including any transients. While this is often desirable for audio material such as vocals, it is not ap-propriate for material with sharp transients such as drums or percussion.

The default setting for this parameter is Auto. In this setting, crossfade times are set

The relative pitch entry parameter

automatically, according to the settings of the Coarse and Fine pitch controls. The Auto setting is appropriate for most appli-cations. However, you can still manually adjust and optimize crossfade times using the Crossfade slider if necessary. For audio material with sharper attack transients, we suggest that you use shorter crossfade times. For audio material with softer attack transients, use longer crossfade times.

Minimum Pitch This slider allows you to set the minimum fundamental pitch that DPP-1 will recognize when performing pitch transposition. This allows you to op-timize DPP-1’s performance by adjusting this parameter based on the lowest funda-mental pitch of the audio material that you wish to process.

On audio material with a low fundamental pitch frequency content (such as an elec-tric bass guitar) setting this control to a lower frequency such as 30 Hz will improve DPP-1’s performance. The most important thing to remember when using this control is that the fundamental frequency of audio material you wish to process must be above the frequency you set here. This slider has an adjustment range of 15 Hz to 1000 Hz. The default setting is 60 Hz. Adjustment of this parameter is tied to the current setting of the Maximum Pitch parameter such that the minimum/maximum range is never less than one octave, and never more than five octaves.

Maximum Pitch This slider allows you to adjust the maximum fundamental pitch that DPP-1 will recognize when performing pitch transposition. This allows you to op-timize DPP-1’s performance by adjusting this parameter (and the Minimum Pitch

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parameter) based on the highest funda-mental pitch of the audio material that you wish to process. This slider has an adjust-ment range of 30 Hz to 4000 Hz. The de-fault setting is 240 Hz.


chapter 11

The Maxim Plug-In

Maxim is a unique and powerful peak-lim-iting and sound maximizing Plug-In pro-vided in TDM, RTAS, and AudioSuite for-mats. Maxim is ideal for critical mastering applications, as well as standard peak-lim-iting tasks.

Maxim offers several critical advantages over traditional hardware-based limiters. Most significantly, Maxim takes full advan-tage of the random-access nature of disk-based recording to actually anticipate peaks in audio material and preserve their attack transients when performing reduc-tion. This makes Maxim more transparent than conventional limiters, since it pre-serves the character of the original audio signal without clipping peaks or introduc-ing distortion.

Maxim provides:

◆ “Perfect attack-limiting” through look-ahead analysis to accurately preserve tran-sient attacks and the character of original program material

◆ A full color histogram which plots input dB history during playback, providing vi-sual feedback for setting threshold level

◆ A user-adjustable ceiling which allows material to be level-optimized for record-ing

◆ 24-bit mono or stereo operation

• Online Help (accessed by clicking on a pa-rameter name) which provides descriptions of each parameter’s function

Maxim

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About Peak LimitingPeak limiting is an important element of contemporary audio production. It is es-sentially the process of preventing signal peaks in audio material from clipping by limiting their dynamic range to an abso-lute, user-selectable ceiling and not allow-ing them to exceed this ceiling.

Limiters do this by allowing the user to se-lect a threshold in decibels. If an audio sig-nal peak exceeds this threshold, gain re-duction is applied, and the audio is attenuated by a user-selectable amount.

Limiting has two main uses in the audio production cycle: • Adjusting the dynamic range of an entire

final mixdown for premastering pur-poses

• Adjusting the dynamic range of individ-ual instruments for creative purposes

Limiting a MixdownThe primary purpose of applying limiting during final mixdown is to flatten out any large peaks remaining in the audio material in order 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 ra-tio, and a smoother mix.

Limiting Individual InstrumentsThe 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 producers do this by apply-ing heavy limiting to flatten the snap of the attack portion of a drum hit and, by ad-justing the release time of the limiter, bring up any room tone contained in the decay portion of the drum sound.

In some cases, this type of limiting can ac-tually 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, how-ever, since it can also bring noise levels up in the source audio if it is present in the lower levels of the audio signal.

How Maxim Differs From Conventional LimitersMaxim 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 be-fore applying limiting. Maxim is then able to instantly apply limiting before a peak builds up. The result is extremely transpar-ent limiting that faithfully preserves the at-tack transients and retains the overall char-acter of the original unprocessed signal.

In addition, Maxim provides a histogram, which displays the distribution of wave-form peaks in the audio signal. This pro-vides a convenient visual reference for comparing the density of waveform peaks at different decibel levels and choosing

how much limiting to apply to the mate-rial.

▲ Maxim introduces 1024 samples of delay into any processed signal. (This is equal to 23.27 milliseconds of delay at 44.1 kHz and 21.33 milliseconds of delay at 48 kHz). If you wish to preserve phase synchronicity between multiple audio sources when Maxim is only ap-plied to one of these sources, use the Time Adjuster Plug-In (included with Pro Tools) to compensate. See the DigiRack Plug-Ins Guide for more information on using Time Adjuster to compensate for TDM-based signal delays.

Maxim ParametersMaxim features the following parameters and indicators:

Input Level

This meter displays the amplitude of input signals prior to limiting. Unlike conven-tional meters, Maxim’s input meter dis-plays the top 24 dB of dynamic range of au-dio 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.

Histogram

The histogram displays the distribution of waveform peaks in the audio signal. Maxim draws this graph based on audio playback. If you have selected and played back a short loop it will build the histo-gram on that data. If you have selected and played back a longer section, it will build

the histogram on that. Maxim holds peak data until you clear the histogram by click-ing on it.

The histogram provides you with a conve-nient visual reference for quickly compar-ing the density of waveform peaks at differ-ent dB levels. You can then base your limiting decisions on this data.

The X axis of the histogram shows the number of waveform peaks occurring at specific dB levels. The Y axis shows the spe-cific 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 num-ber 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 down-wards, you can visually adjust the level at which limiting will occur. Maxim displays the affected range in orange.

The histogram

dB level of waveformpeaks

density of waveformpeaks at each level


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Threshold

This slider allows you to set the threshold level for limiting. Signals that exceed this level will be limited. Signals that are below it will be unaffected. Limited signal peaks are attenuated to match the threshold level, so the value that you set here will de-termine the amount of reduction that is ap-plied.

Output

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.

Ceiling

This slider determines the maximum out-put 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.

Attenuation

This meter displays the amount of gain re-duction being applied over the course of playback, with the maximum peak dis-played in the numeric readout at the bot-tom of the meter. For example, if the nu-merical 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 dur-ing playback and still know the maximum gain reduction value when you come back. To clear the numeric readout, click it with the mouse.


Release

This slider allows you to adjust how long it takes for Maxim to ease off of its attenua-tion after the input signal drops below the threshold level. Because Maxim has an at-tack time of zero milliseconds, the release slider has a very noticeable effect on the character of limiting. In general, if you are using heavy limiting, you should use pro-portionally longer release times in order to avoid pumping that may occur when Maxim is forced to jump back and forth be-tween 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/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 parameter has a default value of 1 millisec-ond.

Mix

This slider allows you to adjust the ratio of dry signal to limited signal. In general, if you are applying Maxim to a main output mix, you will probably wish to set this con-trol to 100% wet. If you are applying heavy limiting to an individual track or element in a mix to modify its character, this pa-rameter is particularly useful since it allows you add precisely the desired amount of the processed effect to the original signal.

Link Button

When depressed, this button (located be-tween the Threshold and Ceiling numeric readouts) links the Threshold and Ceiling parameters. These two sliders will then

move proportionally together. As you ad-just the Threshold lower and lower, the output level ceiling is lowered as well. When these parameters are linked you can conveniently compare the effect of limit-ing at unity gain by simply clicking the By-pass button (on the Inserts/Sends editor).

Dither

When depressed, this button turns Maxim’s built-in dither function on. Dither is a special form of randomized noise that masks quantization noise that sometimes occurs in very low-amplitude audio signals. The most common applica-tion of dither is to use it as the last proces-sor in the signal path on a master output mix to ensure clean, low-level audio qual-ity in signals with fade-outs or reverb tails when reducing bit-depth.

Applying dither helps reduce quantization noise that can occur when you are mixing from a 24-bit TDM environment 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 func-tion saves you the trouble and DSP re-sources of having to use a separate Dither Plug-In.

If Dither is disabled, the Noise Shaping and Bit Resolution parameters will have no ef-fect.

The Link button

link button

Noise Shaping

When depressed, this button changes the character of the dither from conventional dither to noise-shaped dither. Noise shap-ing biases the dither noise to less audible high frequencies so that it is not as readily perceived by the ear. Dither must be en-abled in order to use Noise Shaping.

Bit Resolution

These buttons allow you to choose one of three possible resolutions for dither pro-cessing. As a general rule, you should set this parameter to the maximum bit resolu-tion of your destination media.

◆ 16-bit is recommended for output to dig-ital devices such as DAT recorders and CD recorders since they have a maximum reso-lution of 16-bits.

◆ 18-bit is recommended for output to an-alog devices if you are using a Pro Tools 888 I/O or 882 I/O Audio Interface since the 18-bit setting allows you to obtain the maximum quality available from the 18-bit digital-to-analog converters of these de-vices.

◆ 20-bit is recommended for output to dig-ital devices that support a full 20-bit re-cording data path such the Sony PCM-9000 optical mastering recorder, or the Alesis ADAT XT 20. This setting is also recom-mended for output to analog devices if you are using a Pro Tools 882/20 Audio Inter-face.

It is also recommended for use with digital effects devices that support 20-bit input and output, since it provides for a lower noise floor and greater dynamic range


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when mixing 20-bit signals directly into the TDM environment.

Using Maxim EffectivelyMaxim is as simple and easy to use as a tra-ditional limiter but with added benefits that only a Plug-In can provide. The sec-tion that follows provides a few suggestions for using Maxim most effectively.

To use Maxim:

1 With Maxim applied to the desired track, locate the sections of the audio where the most prominent peak levels occur.

2 Loop playback and look at the data dis-played by both the histogram and the at-tenuator meter.

3 Click the Link button to link the Thresh-old and Ceiling controls. This will allow you to adjust these parameters together proportionally and, using the Bypass but-ton on the Inserts/Sends Editor, compare the audio with and without limiting.

4 Adjust the Threshold slider downwards until you hear and see the limiting occur-ring, then bring the Threshold back up slightly until you have roughly the amount of limiting you want.

5 Periodically clear the attenuator meter by clicking on it so that you always see how much attenuation is occurring. A good rule of thumb on pop-oriented material is 2 dB to 4 dB of attenuation on occasional peaks. This doesn’t mean 2 dB to 4 dB of continu-ous attenuation throughout the entire pro-gram material, but only on occasional peaks.


6 Use the Bypass button to compare the processed and unprocessed sound and to check if the results are acceptable. In most cases you should try to achieve a good, transparent sound when adjusting the amount of limiting.

7 Avoid pumping effects when you are us-ing heavier limiting by setting the Release slider to longer values.

8 Once you’ve got the effect you want, dis-able the Link feature, and adjust the Ceil-ing slider to raise the output level up to maximize signal levels without clipping. In general, a value of 0.5 dB or so is a good maximum ceiling. We recommend that you don’t set the ceiling all the way to zero because some DAT recorders and many CD player DACs will actually clip at or slightly below zero.

9 If you are using Maxim on an output mix that will be faded out, enable the dithering options that you desire in order to improve the signal performance of the material as it fades to lower amplitudes.

Maxim and MasteringIf you intend to deliver your audio material as a 24-bit file on disk to a professional mastering engineer who will then be mak-ing final adjustments, be aware that many engineers want material delivered without dither and without levels being optimized or normalized.

These 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, you may

wish to just apply the limiting that you find creatively appropriate (adding a little punch to certain instruments in the mix, for example).

However, if you intend to output the mate-rial to a DAT recorder, we suggest that you use appropriate limiting and add dither since the extra bits achieved when mixing and processing in the 24-bit Pro Tools mix environment will otherwise be discarded by the 16-bit environment of the DAT re-corder or CD-R.

Doing so will allow you to optimize the dy-namic range and maximally preserve the activity of the lower, or least significant bits in the audio signal, smoothly dithering them into the 16-bit output that is sent to the DAT deck or CD-R.


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chapter 12

The Bruno & Reso Plug-Ins

Bruno and Reso are a pair of real-time TDM Plug-Ins that process audio using an inno-vative sound generation technique known as cross-synthesis.

Cross-synthesis generates complex sound textures by using an audio track as a tone source and then applying a variety of syn-thesizer-type effects to that tone source.

Bruno and Reso each use a different sound generation method:

◆ Bruno uses time-slicing, a technique whereby timbres are extracted from the source audio during playback and cross-faded together. This crossfading between signals can create a rhythmic pulse in the sound as the timbre changes.

◆ Reso uses a resonator, which adds har-monic overtones to the source audio through a short signal delay line with a feedback loop.

In both cases, the processed sound can then be played—either in real time (with an on-screen keyboard or an external MIDI controller) or sequenced using the MIDI re-cording and playback capabilities of Pro Tools.

Bruno features:

• Time-slice tone generation with adjust-able crossfade

• Up to 24 voices of polyphony• Multi-voice detuning • Editable ADSR envelope generator• Portamento• Velocity-sensitive gain and detuning• Time-slice switching using envelope trig-

gering or MIDI beat clock• Voice-stacking• Side-chain input for control via an exter-

nal audio source• Online help

Reso features:

• Harmonic resonance generation• Up to 24 voices of polyphony• Multi-voice detuning• Resonant low-pass filter• Editable ADSR envelope generator• Portamento• Velocity-sensitive resonance, damping,

gain, and detuning• Harmonic switching using envelope trig-

gering or MIDI beat clock• Voice-stacking

Chapter 12: The Bruno & Reso Plug-Ins 93

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• Side-chain input for control via an exter-nal audio source

• Online help

DSP RequirementsBruno and Reso each require one full DSP chip on a Pro Tools 24 MIX card or DSP Farm card.

DSP and Voice PolyphonyThe maximum number of Bruno/Reso voices available per DSP chip depends on the type of DSP cards in your system. Bruno and Reso are optimized for use on the MIX card.

MIX Cards

A maximum of 24 voices of polyphony are available per chip on MIX cards. MIX Farm cards are equipped with six DSP chips. Of these, three are the type required by Bruno/Reso. For this reason, a maximum of three Bruno/Reso Plug-Ins can be used per MIX Farm card.

DSP Farm Cards

A maximum of 8 voices of polyphony are available per chip on DSP Farm cards. All four DSP chips on a DSP Farm card can be used by Bruno/Reso (if available).

▲ If you use Bruno/Reso on a MIX card-equipped Pro Tools system and later open that session on a DSP Farm-only system, Bruno/Reso’s voice polyphony will be automat-ically reduced from 24 voices to 8 voices per Plug-In.


Inserting Bruno/Reso onto an Audio Track To use Bruno/Reso in a Pro Tools session, you must add it to a track as an insert. Once Bruno/Reso is inserted on the target track, you can adjust its parameters to achieve the effect that you want, then play the Plug-In using the on-screen keyboard, an external MIDI controller, or a MIDI track.

To add Bruno/Reso as a track Insert:

1 Click the Inserts pop-up on the desired track and select Bruno or Reso.

2 Click Play on the Pro Tools Transport to start audio playback.

3 Play Bruno/Reso with the on-screen key-board or by MIDI control. (See “Playing Bruno/Reso” for details.)

4 Adjust Bruno/Reso parameters to achieve the effect you want. (Bruno and Reso’s pa-rameters are explained later in this chap-ter.)

Keyboard Shortcuts

■ For finer adjustments, Command-drag (Macintosh) or Control-drag (Windows) the control.

Inserting Bruno onto an audio track

■ To return a control to its default value, Option-click (Macintosh) or Alt-click (Win-dows) the control.

To remove Bruno/Reso from a track:

■ Click the Insert Selector and choose No Insert.

▲ Each time you insert or remove Bruno/Reso on a track (or open a session which uses Bruno/Reso), Pro Tools will notify you that your MIDI setup has changed. This is because Pro Tools must create and assign a “virtual node” for MIDI communication with Bruno/Reso.

Keyboard Input of ParametersYou can use your computer keyboard to in-put and edit Bruno/Reso’s parameters.

To set Bruno/Reso parameters with your computer keyboard:

◆ Click the text box for the parameter and type the value.

◆ For values in kilohertz, you can type “K” after a number to multiply it by 1000. (For example, to enter a value of 8000, type “8k”).

Removing an Insert

◆ To increase a slider value, press the Up Arrow on your keyboard. To decrease a slider value, press the Down Arrow.

◆ Press Enter on the numeric keypad after typing a value to input the value without leaving the selected parameter field.

◆ On the main keyboard (not the numeric keypad) Press Return (Macintosh) or Enter (Windows) to enter the value and leave keyboard editing mode.

◆ To move downward through the differ-ent parameter fields, press the Tab key. To move upwards, press Shift-Tab.

Playing Bruno/ResoIn order to generate sound, Bruno/Reso must be played while audio playback is en-gaged. You can play Bruno/Reso in two ways:

◆ In real time, using either the built-in on-screen keyboard or an external MIDI con-troller.

◆ By recording or importing a MIDI se-quence onto a Pro Tools MIDI track.

Using the On-Screen KeyboardThe simplest way to play Bruno/Reso is to use its on-screen keyboard. While the mouse physically limits you to clicking one note at a time, the keyboard latch feature allows keys to remain depressed after you release the mouse button—making it possi-ble to assemble chords.

▲ Notes played with the on-screen keyboard are triggered at a MIDI velocity value of 92.


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To play Bruno/Reso with the on-screen keyboard:

1 With Bruno/Reso inserted on the desired track, open the Plug-In Editor window by clicking the corresponding Insert Selector.

2 Click Play on the Pro Tools Transport to start audio playback.

3 Click the on-screen keyboard to play the audio material. Bruno/Reso will only pro-duce sound while audio plays on the source track.

To latch keys on the on-screen keyboard

1 Click the Latch bar, then click the desired key(s). Chords can be played in this way.

2 To turn off a latched key, click it a second time.

3 To turn off key latching entirely, click the Latch bar a second time.

✽ Saving a Bruno or Reso setting with the Li-brarian while keys are latched will save the latched keys as part of the settings file.

Using MIDI If you have a MIDI keyboard controller, you can play Bruno/Reso live or record a MIDI track and route it to Bruno/Reso for playback.

On Macintosh systems, Pro Tools requires OMS for MIDI communication with Bruno/Reso. If OMS is not currently in-stalled and configured on your system, re-fer to your Pro Tools System Installation Guide for instructions.


▲ On Macintosh systems, Mark of the Uni-corn’s FreeMIDI software cannot be used to control Bruno/Reso using MIDI since it does not support virtual nodes.

To configure Bruno/Reso for MIDI input with Pro Tools:

1 If your MIDI interface and MIDI key-board controller are not already connected and configured for use with Pro Tools, con-nect and configure them now. (Refer to your Pro Tools System Installation Guide and MIDI Controllers Guide for detailed setup in-structions.)

2 Choose Options > MIDI Thru.

3 With Bruno/Reso inserted on the desired track, open the Plug-In Editor window by clicking the corresponding Insert Selector.

4 Choose File > New Track and create a new MIDI track. You will use this track to route the MIDI controller to Bruno/Reso.

▲ You must create a separate MIDI track for each Bruno/Reso Plug-In that you want to con-trol via MIDI.

5 On the new MIDI track, click the Output pop-up and choose Bruno or Reso. (If you are using multiple Bruno/Reso Plug-Ins in a session, they will be numbered in this pop-up. Be sure that you route the MIDI track to the correct one.)

6 Record-enable the MIDI track.

7 To test your MIDI connection, play a few notes on your MIDI keyboard. The corre-sponding notes should highlight on Bruno/Reso’s on-screen keyboard.

To play Bruno/Reso with a MIDI controller:

1 Start audio playback.

2 Play your MIDI keyboard as the audio material plays back. Bruno/Reso will only produce sound while audio plays on the source track.

Using a MIDI TrackYou can also use a MIDI track to trigger Bruno/Reso. You have the choice of record-ing a new MIDI track, or importing an ex-isting sequence.

▲ You must create a separate MIDI track for each Bruno/Reso Plug-In that you want to con-trol via MIDI.

Recording a New MIDI Track

To record a new MIDI track for triggering Bruno/Reso:

1 Make sure that your MIDI setup is config-ured according to the instructions given in the preceding section, “Using MIDI.”

2 Insert Bruno/Reso on the desired track.

3 Make sure that MIDI Thru is enabled in the Options menu.

Routing MIDI output to Bruno

4 Choose File > New Track and create a new MIDI track.

5 Click the MIDI track’s Output pop-up and choose Bruno or Reso. (If you are using multiple Bruno/Reso Plug-Ins in a session, they will be numbered in this pop-up. Be sure that you route the MIDI track to the correct one.)

6 Record-enable the MIDI track.

7 Click Record and Play on the Pro Tools Transport.

8 Play your MIDI controller. Your perfor-mance will trigger Bruno/Reso as it is re-corded.

9 When you have finished, stop playback.

10 Replay the session to hear the sequence trigger Bruno/Reso.

✽ MIDI tracks need not be record-enabled dur-ing track playback—only during MIDI input.

Importing a MIDI Sequence

To import a MIDI sequence for triggering Bruno/Reso:

1 Choose File > Import MIDI/Track.

2 Locate the desired sequence and click Im-port.

3 Click the new MIDI track’s Output pop-up and choose Bruno or Reso. (If you are using multiple Bruno/Reso Plug-Ins in a session, they will be numbered in this pop-up. Be sure that you route the MIDI track to the correct one.)

4 Play the session to hear the sequence trig-ger Bruno/Reso.


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Using a Key Input for Side Chain ProcessingBruno and Reso feature side-chain process-ing capabilities. Side-chain processing al-lows you to trigger certain parameters from a separate reference track or external audio source. The source used for triggering is re-ferred to as the key input.

You can use this capability to control the rate at which Bruno performs sample switching or Reso toggles its harmonics back and forth via the dynamics of another signal (the key input).

Typically, a rhythm track such as a drum kit is used to trigger these parameters and create rhythmic timbral changes that match the groove of the key input.

To use a key input for side chain processing:

1 Click the Side Chain Input selector and choose the input/bus with the audio you want to use to trigger the Plug-In.

2 Click the Key Input button (the button with the key icon above it). This activates side chain processing.

3 To hear the audio source you have se-lected to control side chain input, click the Key Listen button (the button with the ear icon above it).

Selecting an input from the Key Input pop-up


4 Begin playback. The Plug-In uses the in-put/bus that you chose as a side chain in-put to trigger its effect.

5 Adjust other parameters to achieve the fi-nal desired effect.

Bruno ParametersBruno uses time-slicing for tone genera-tion, extracting timbres from the audio track during playback and cross-fading them together at a user-selectable rate.

This crossfading can create a rhythmic pulse in the sound as the timbre changes. This makes Bruno ideal for creating tonal effects with a continuously shifting tim-bre—similar to the wave sequencing found on synthesizers such as the PPG, Prophet VS, Korg Wavestation, and Waldorf XT.

By carefully choosing the type of source au-dio, the crossfade length, and the type of switching, you can use Bruno to create complex sound textures that are both evoc-ative and unique.

On-Screen Keyboard

To generate sound, Bruno must be played while audio is playing back. You can do this by clicking a key on the on-screen key-board.

The on-screen keyboard also provides a latch feature which allows keys to remain engaged even after you release the mouse button. In this mode, multiple keys can be engaged at once.

Notes played with the on-screen keyboard are triggered at a MIDI velocity value of 92.

Bruno

The on-screen keyboard

Timbre Parameters

Crossfade

The Crossfade control sets the rate at which Bruno extracts timbres from the source au-dio and crossfades from one time-slice to the next. The range of this control is from 2 to 40 Hz (cycles per second).

The higher the crossfade frequency, the smaller the time slice, and the faster Bruno moves between slices. A higher frequency crossfade would retain more characteristics of the original audio source and would have a pulsed or wave-sequenced feel.

The lower the crossfade frequency, the larger the time slice, and the slower Bruno moves between slices. A lower frequency crossfade would have fewer characteristics of the original source and a more rounded or gradually evolving sound.

Switch

The Switch control causes Bruno to switch directly between time-sliced samples with-out crossfading them. This adds a distinct rhythmic pulse to the timbral changes.

Bruno’s Timbre parameters


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Switching can be controlled either by trig-gering (using the dynamics of the source au-dio itself, or those of an external key input) or by MIDI clock.

External Key This enables you to trigger switching from a separate reference track or external audio source. The source used for triggering is referred to as the key input and is selected via the Side Chain Input pop-up. You can assign either an audio input chan-nel or a TDM bus channel.

Typically, a drum track is used as a key in-put so that switching occurs according to a definite rhythmic pattern.

Key Listen When enabled, this allows you to monitor the source of the key input. It is often useful to do this in order to fine tune Bruno’s settings to the key input.

See “Using a Key Input for Side Chain Pro-cessing” for details.

Threshold This sets the level in decibels above which switching occurs. When the audio input level rises above the Threshold level, Bruno will switch directly to a new time-slice. The range of this control is from a low of –48 dB (maximum switching) to a high of 0.0 dB (no switching). If no key in-put is used, the dynamics of the source au-dio will trigger switching. If a key input is used, the dynamics of the key input signal will trigger switching. Threshold-based switching can be used at the same time as Key Input-based switching.

MIDI Clock This allows you to trigger switching in sync with a MIDI beat clock signal. This creates a very regular, highly rhythmic wave sequencing effect that is ideal for sessions arranged around MIDI beat clock. This parameter can be set to


quarter, eighth, or sixteenth notes, or dot-ted/triplet values of the same.

✽ For quick numeric entry of MIDI beat clock values, type “4,” “8,” or “16” for quarter notes, eight notes, or sixteenth notes. Add “t” for triplets, or “d” for dotted note values. Typ-ing “4t” for example, enters a quarter note triplet value. Typing “16d” enters a dotted six-teenth note value.

▲ Pro Tools version 4.3.1 and earlier cannot send MIDI beat clock information to Bruno/Reso. However, other products that support MIDI beat clock can send that infor-mation using OMS. Refer to your MIDI se-quencer User Guide for details.

TimbrometerThis multicolor waveform display shows the amplitude and duration of the audio signal generated by Bruno as well as the fre-quency of timbral changes and whether they are crossfaded or switched.

Red and blue waveform segments indicate timbral changes that are crossfaded. Green waveform segments indicate timbral changes that are hard switched.

Amplitude Parameters

Gain Amount

Gain Amount allows you to attenuate out-put level gain. Since some of Bruno’s pa-rameters can cause extreme changes in sig-nal level, this is particularly useful for preventing clipping and achieving unity gain with the original signal level. This pa-rameter is adjustable from a low of –96 dB (no gain) to a high of 0.0 dB (maximum gain).

Gain Velocity

Gain Velocity sets the velocity sensitivity of the Gain Amount parameter. This gives you touch-sensitive control over Bruno’s volume via a MIDI keyboard.

This parameter is adjustable from a low of –24 dB (maximum velocity sensitivity) to a high of 0.0 dB (no velocity sensitivity).

If you set Gain Velocity to –24 dB, a soft strike on a key will reduce gain up to –24 dB. A hard strike will have a maximum

Bruno’s Amplitude parameters

output level equal to the current dB setting of the Gain Amount parameter.

Conversely, if Gain Velocity is set to 0.0 dB, Bruno’s volume will not change no matter how hard or soft you strike a key on your MIDI controller.

▲ Gain Velocity only has an effect when you play Bruno with a MIDI controller with velocity sensitivity.

Mix

The Mix control allows you to adjust the mix of the processed audio with the origi-nal, unprocessed audio.

Spread

When Bruno is used in stereo, the Spread control can be used to pan multiple voices within the stereo field. This parameter is adjustable from 0% (no stereo spread) to 100% (maximum stereo spread).

Voice stacking has a direct effect on stereo Spread. For example, setting Voice Stack to “1” and Spread to “100%” will randomly pan each note played. Setting Voice Stack to “4” and Spread to “100%”, will pan two of the voices hard left, and two voices hard right.

ADSR Envelope Generator

The ADSR (attack, decay, sustain, release) Envelope Generator allows you to control Bruno’s amplitude envelope. This ampli-tude envelope is applied to a sound each time a note is struck.


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The four envelope parameters can be ad-justed by dragging the appropriate break-point, or by typing in a numeric value.

Attack Controls the amount of time in milliseconds that the sound takes to rise from zero amplitude to its full level. The longer the attack, the more time it takes for the sound to reach maximum volume after the a note is struck. This parameter is ad-justable from 0.0 to 5000 milliseconds.

Decay Controls the amount of time in mil-liseconds that the sound takes to fall from its peak attack level to the sustain level. This parameter is adjustable from 0.0 ms to 5000 ms.

Sustain Level Controls the amplitude level in dB that is reached after the decay time has elapsed. The amplitude level stays con-stant as long as a MIDI note remains de-pressed. This parameter is adjustable from –96 dB (no sustain) to 0.0 dB (maximum sustain)

Release Controls the amount of time in milliseconds that the sound takes to fall from the Sustain level to zero amplitude af-ter a note is released. This parameter is ad-justable from 0.0 ms to 5000 ms.


Pitch Parameters

Glide

Glide, also known as portamento, deter-mines the amount of time it takes for a pitch to glide from the current note to the next note played. This effect is commonly found on synthesizers.

This parameter is adjustable from a low of 0.0% (no glide) to a high of 100% (maxi-mum glide). A setting of 100% will take the longest time to travel from the current note to the next note played. The effect is also dependent on the interval (distance of pitch) between the two notes: the larger the interval, the more noticeable the effect.

Bend Range

This allows you to set the maximum inter-val of pitch bend that can be applied to Bruno with a MIDI controller’s pitch bend wheel. This parameter is adjustable from 0 semitones (no bend) to 12 semitones (1 oc-tave).)

Bruno’s Pitch parameters

Master Tune

The Master Tune control can be used to tune the pitch of Bruno’s output to another instrument. By default, this parameter is set to 440.0 Hz It can be adjusted from a low of 430.0 Hz to a high of 450.0 Hz.

Detune Amount

Detuning is a common sound-thickening technique used on synthesizers and many effects devices. Bruno’s Detune Amount parameter allows you to set the maximum amount of pitch detuning that occurs when multiple voices are stacked together using Voice Stacking. Using a combination of voice stacking and detuning, you can create timbres that are exceptionally fat.

Voices can be detuned up to 50.0 cents. (One cent is equal to 1/100th of a semi-tone.)

Detune Velocity

Detune Velocity controls how MIDI key ve-locity affects voice detuning. This parame-ter gives you velocity-sensitive control over voice detuning when you play Bruno with a MIDI keyboard.

This parameter is adjustable from a low of 0.0 cents (no velocity-sensitive detuning) to a high of 50.0 cents (maximum velocity-sensitive detuning).

If Detune Velocity is set to 0.0 cents, detun-ing will not change no matter how hard you strike a key on your MIDI controller. Conversely, if you set Detune Velocity to 50.0 cents, a hard strike will detune voices a maximum of 50.0 cents (in addition to

the detuning specified with the Detune Amount parameter).

▲ Detune Velocity only has an effect when you play Bruno with a velocity-sensitive MIDI con-troller.

Voice ParametersThese parameters control Bruno’s voice po-lyphony and allocation.

Mode

Mono (monophonic)

Monophonic mode can be described as sin-gle-key mode, since it only allows Bruno to play a single note at a time (though multi-ple voices can be stacked on the same note using the Voice Stacking parameter). Monophonic mode gives voice priority to the most recently played note.

Poly (polyphonic)

Polyphonic mode allows you to play multi-ple notes simultaneously, to create chords.

Bruno’s Voice parameters


104

The number of notes that can be played si-multaneously depends on the Voice Stack-ing setting you have chosen. A voice stack setting of 1, for example, will allow you to play 24 individual notes simultaneously (a 24 note chord, if you will). A voice stack setting of 24, on the other hand, will allow you to play only one note at a time, but will stack all 24 voices on that same note for an extremely fat sound.

Voice Stack

This selects the number of voices that are used, or stacked when you play a single note. The number of voices that you choose to stack will directly affect polyph-ony. Selecting a larger number of stacked voices will reduce the number of notes that you can play simultaneously.

You can simultaneously play up to:

◆ 24 notes in a 1-voice stack



◆ 3 notes in an 8-voice stack

◆ 1 note in a 24-voice stack

If all available voices are being used, play-ing an additional note will replace the note in the chord that was played first.

▲ A maximum of 8 voices of polyphony are available per Bruno/Reso Plug-In on DSP Farm cards.

Online Help

Online help


To use online help, click the name of any control or parameter and an explanation will appear. Clicking the Online Help but-ton itself provides further details on using this feature.

Reso ParametersReso synthesizes new harmonic overtones from the source audio signal, creating har-monically rich timbres with a metallic, synthesizer-like character.

On-Screen Keyboard

To generate sound, Reso must be played while audio is playing back. You can do

Reso

The on-screen keyboard

this by clicking a key on the on-screen key-board.

The on-screen keyboard also provides a latch feature which allows keys to remain engaged even after you release the mouse button. In this mode, multiple keys can be engaged at once.

Notes played with the on-screen keyboard are triggered at a MIDI velocity value of 92.

Timbre Parameters

Resonance Amount

Resonance Amount controls the intensity of harmonic overtones produced by the Resonator. Increasing the Resonance Amount will increase the overall harmonic content of the sound while increasing the sustained portions of the generated har-monics.

The frequency content of the input signal largely determines what harmonics are generated by the resonator. For this reason, the character of the resonance will change

Reso’s Timbre parameters

according to the type of audio that you process.

Resonance Velocity

Resonance Velocity allows you to increase or decrease resonance according to how hard a MIDI key is struck and how much resonance is initially specified with the Resonance Amount parameter.

Resonance Velocity is adjustable from a low of –10 to a high of +10. With positive values, the harder the key is struck, the more resonance is applied. With negative values, the harder the key is struck, the less resonance is applied.

The effectiveness of this parameter de-pends on the Resonance Amount setting. For example, if Resonance Amount is set to “0”, setting the Resonance Velocity to a negative value will have no effect, since there is no resonance to remove. Similarly, if the Resonance Amount parameter is set to 10, setting Resonance Velocity to +10 will have no effect since the resonance is al-ready at its maximum.

For optimum effect, set the Resonance Amount to a middle value, then set Reso-nance Velocity accordingly for the desired effect.

▲ Resonance Velocity only has an effect when you play Reso with a velocity-sensitive MIDI controller.

Damping Amount

Damping causes the high-frequency har-monics of a sound to decay more rapidly than the low-frequency harmonics. It al-lows you to control the brightness of the


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signal generated by Reso's Resonator and is particularly useful for creating harp or plucked string-like textures.

The range of this parameter is from 0 (no damping) to 10 (maximum damping). The greater the amount of damping, the faster the high frequency harmonics in the audio will decay and the duller it will sound.

Damping Velocity

Damping Velocity allows you to increase or decrease damping according to how hard a MIDI key is struck and how much damping is initially specified with the Damping Amount parameter.

Damping Velocity is adjustable from a low of –10 to a high of +10. With positive val-ues, the harder the key is struck, the more damping is applied. With negative values, the harder the key is struck, the less damp-ing is applied (which simulates the behav-ior of many real instruments).

The effectiveness of this parameter de-pends on the Damping Amount setting. For example, if Damping Amount is set to zero, setting the Damping Velocity to a negative value will have no effect, since there is no damping to remove. Similarly, if the Damping Amount parameter is set to 10, setting Damping Velocity to +10 will have no effect since damping is already at its maximum.

For optimum effect, set the Damping Amount to a middle value, then set Damp-ing Velocity accordingly for the desired ef-fect.


▲ Damping Velocity only has an effect when you play Reso with a velocity-sensitive MIDI keyboard controller.

Harmonics

A resonator adds harmonic overtones to the source audio signal that are integer multiples of the fundamental frequency of the signal.

The Harmonics parameter allows you to choose whether you want to use all of these harmonics, or just the odd-numbered in-tervals. Your choice will affect the timbre of the sound that is produced.

All This adds all of the harmonic overtones generated by the resonator. In synthesizer parlance, this produces a somewhat buzz-ier, sawtooth wave-like timbre.

Odd This adds only the odd-numbered harmonic overtones generated by the reso-nator. In synthesizer parlance, this pro-duces a somewhat more hollow, square wave-like timbre.

Toggle

Reso can automatically toggle between the All and Odd harmonics settings, producing a rhythmic pulse in the timbre.

Harmonic toggling can be controlled either by triggering (using the dynamics of the source audio itself, or those of an external key input) or by MIDI clock.

External Key This enables you to toggle the harmonics from a separate reference track or an external audio source. The source used for toggling is referred to as the key in-put and is selected via the Side Chain Input

pop-up. You can assign either an audio in-put channel or a TDM bus channel.

Typically, a drum track is used as a key in-put so that toggling occurs according to a definite rhythmic pattern.

Key Listen When enabled, this allows you to monitor the source of the key input. It is often useful to do this in order to fine tune Reso’s settings to the key input.

✽ Setting up and using a key input is covered in greater detail in the section entitled “Using a Key Input for Side Chain Processing.”

Threshold This sets the level in decibels above which toggling occurs. When the au-dio input level rises above the Threshold level, Reso will toggle its harmonics setting. The range of this control is from a low of –48 dB (maximum toggling) to a high of 0.0 dB (no toggling). If no key input is used, the dynamics of the source audio will trigger toggling. If a key input is used, the dynamics of the key input signal will trig-ger toggling. Threshold-based switching can be used at the same time as Key Input-based switching.

MIDI Clock This allows you to trigger tog-gling in sync with a MIDI beat clock signal. This creates a very regular, highly rhythmic wave sequencing effect that is ideal for ses-sions arranged around MIDI beat clock. This parameter can be set to quarter, eighth, or sixteenth notes, or dotted/triplet values of the same.

✽ For quick numeric entry of MIDI beat clock values, type “4,” “8,” or “16” for quarter notes, eight notes, or sixteenth notes. Add “t” for triplets, or “d” for dotted note values. Typ-ing “4t” for example, enters a quarter note triplet value. Typing “16d” enters a dotted six-teenth note value.

▲ Pro Tools version 4.3.1 and earlier cannot send MIDI beat clock information to Bruno/Reso. However, other products that support MIDI beat clock can send MIDI beat clock using OMS. Refer to your MIDI se-quencer User Guide for details.

Amplitude Parameters

Gain Amount

Gain Amount allows you to attenuate out-put level gain. Since resonation can cause extreme changes in signal level, this is par-ticularly useful for preventing clipping and achieving unity gain with the original sig-nal level. This parameter is adjustable from

Reso’s Amplitude parameters


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a low of –96 dB (no gain) to a high of 0.0 dB (maximum gain).

Gain Velocity

Gain Velocity sets the velocity sensitivity of the Gain Amount parameter. This gives you touch-sensitive control over Reso’s vol-ume via a MIDI keyboard.

This parameter is adjustable from a low of –24 dB (maximum velocity sensitivity) to a high of 0.0 dB (no velocity sensitivity).

If you set Gain Velocity to –24 dB, a soft strike on a key will reduce gain up to –24 dB. A hard strike will have a maximum output level equal to the current dB setting of the Gain Amount parameter.

Conversely, if Gain Velocity is set to 0.0 dB, Reso’s volume will not change no matter how hard or soft you strike a key on your MIDI controller).

▲ Gain Velocity only has an effect when you play Reso with a velocity-sensitive MIDI key-board controller.

Mix

The Mix control allows you to adjust the mix of the processed audio with the origi-nal, unprocessed audio.

Spread

When Reso is used in stereo, the Spread pa-rameter can be used to pan multiple Reso voices within the stereo field. This parame-ter is adjustable from 0% (no stereo spread) to 100% (maximum stereo spread).

Voice stacking affects stereo Spread. For ex-ample, setting Voice Stack to 1 and Spread


to 100% will alternately pan each note played right and left. Setting Voice Stack to 4 and Spread to 100%, will pan two of the five voices hard left, and two voices hard right.

ADSR Envelope Generator

The ADSR (attack, decay, sustain, release) Envelope Generator allows you to control Bruno’s amplitude envelope. This ampli-tude envelope is applied to a sound each time a note is struck.

The four envelope parameters can be ad-justed by dragging the appropriate break-point, or by typing in a numeric value.

Attack Controls the amount of time in milliseconds that the sound takes to rise from zero amplitude to its full level. The longer the attack, the more time it takes for the sound to reach maximum volume after the a note is struck. This parameter is ad-justable from 0.0 to 5000 milliseconds.

Decay Controls the amount of time in mil-liseconds that the sound takes to fall from its peak attack level to the sustain level. This parameter is adjustable from 0.0 ms to 5000 ms.

Sustain Level Controls the amplitude level in dB that is reached after the decay time has elapsed. The amplitude level stays con-stant as long as a MIDI note remains de-pressed. This parameter is adjustable from –96 dB (no sustain) to 0.0 dB (maximum sustain)

Release Controls the amount of time in milliseconds that the sound takes to fall from the Sustain level to zero amplitude af-ter a note is released. This parameter is ad-justable from 0.0 ms to 5000 ms.

Pitch Parameters

Glide

Glide, also known as portamento, deter-mines the amount of time it takes for a pitch to glide from the current note to the next note played. This effect is commonly used on synthesizers.

This parameter is adjustable from a low of 0.0% (no glide) to a high of 100% (maxi-mum glide). A setting of 100% will take the longest time to travel from the current note to the next note played. The effect is also dependant on the interval (distance of pitch) between the two notes: the larger the interval, the more noticeable the effect.

Bend Range

This allows you to set the maximum inter-val of pitch bend that can be applied to Reso with a MIDI controller’s pitch bend wheel. This parameter is adjustable from 0

Reso’s Pitch parameters

semitones (no bend) to 12 semitones (1 oc-tave).)

Master Tune

The Master Tune control can be used to tune the pitch of Reso’s output to another instrument. By default, this parameter is set to 440.0 Hz It can be adjusted from a low of 430.0 Hz to a high of 450.0 Hz.

Detune Amount

Detuning is a common sound-thickening technique used on synthesizers and many effects devices. Reso’s Detune Amount pa-rameter allows you to set the maximum amount of pitch detuning that occurs when multiple voices are stacked together using Voice Stacking. Using a combination of voice stacking and detuning, you can create timbres that are exceptionally fat.

Voices can be detuned up to 50.0 cents. (One cent is equal to 1/100th of a semi-tone.)

Detune Velocity

Detune Velocity controls how MIDI key ve-locity affects voice detuning. In essence, this parameter gives you touch-sensitive control over voice detuning when you play Reso with a MIDI keyboard.

This parameter is adjustable from a low of 0.0 cents (no velocity-sensitive detuning) to a high of 50.0 cents (maximum velocity-sensitive detuning).

If Detune Velocity is set to 0.0 cents, detun-ing will not change no matter how hard or soft you strike a key on your MIDI control-ler). Conversely, if you set Detune Velocity


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to 50.0 cents, a hard strike will detune voices a maximum of 50.0 cents.

▲ Detune Velocity only has an effect when you play Reso with a velocity-sensitive MIDI key-board controller.

LPF/Voice Parameters

LPF (Low Pass Filter)

Reso’s low-pass filter is a single resonant fil-ter that is applied to all of Reso’s voices.

Frequency

The Frequency control allows you to set the cutoff frequency of the low pass filter in Hertz. All frequencies above the selected cut-off frequency will be attenuated.

The range of this parameter is from 20 Hz to 20 kHz.

The Q Parameter

Q, sometimes referred to as “resonance” on synthesizers, allows you to adjust the

Reso’s Filter and Voice parameters


height of the resonant peak that occurs at the filter’s cutoff frequency.

Increasing the Q increases the volume of frequencies near the filter’s cutoff fre-quency (suppressing the more remote fre-quencies) and adds a nasal quality to the audio. High Q settings allow you to create wah-wah type effects, particularly when the filter is swept with the Follower.

The range of this parameter is from 0 to 10.

Follower

The Follower is an envelope follower which allows the filter cutoff frequency to dynam-ically follow the amplitude of the source audio signal.

The range of this parameter is from a low of –10 to a high of +10. With positive values, the louder the source audio, the higher the cutoff frequency and the wider the filter will open for a brighter sound. With nega-tive values, the louder the source audio, the lower the cutoff frequency and the more the filter will close for a duller sound.

The effectiveness of the Follower depends on the filter’s Frequency setting. For exam-ple, setting the Follower to +10 and select-ing a low Frequency setting will sweep the filter wide on loud passages. However, if the cutoff frequency is at its maximum, set-ting the Follower to +10 will not sweep the filter at all since it is already completely open.

When used with high Q settings and a rel-atively low cutoff frequency, the Follower can be used to produce an automatic wah-wah-type effect.

Voice

Mono (monophonic)

Monophonic mode can be described as sin-gle-key mode, since it only allows Bruno to play a single note at a time (though multi-ple voices can be stacked on the same note using the Voice Stacking parameter). Monophonic mode gives voice priority to the most recently played note.

Poly (polyphonic)

Polyphonic mode allows you to play multi-ple notes simultaneously, to create chords. The number of notes that can be played si-multaneously depends on the Voice Stack-ing setting you have chosen. A voice stack setting of 1, for example, will allow you to play 24 individual notes simultaneously (a 24 note chord, if you will). A voice stack setting of 24, on the other hand, will allow you to play only one note at a time, but will stack all 24 voices on that same note for an extremely fat sound.

Voice stack

This selects the number of voices that are used, or stacked when you play a single note. The number of voices that you choose to stack will directly affect polyph-ony. Selecting a larger number of stacked voices will reduce the number of notes that you can play simultaneously.

You can simultaneously play up to:




◆ 3 notes in an 8-voice stack

◆ 1 note in a 24-voice stack

If all available voices are used up, depress-ing an additional key will steal the “oldest” note in the chord.

▲ A maximum of 8 voices of polyphony are available per Bruno/Reso Plug-In on DSP Farm cards.

Online Help


Bruno/Reso Demo Session Bruno and Reso include a demo session that illustrates some of some useful appli-cations for these Plug-Ins and the many ef-fects you can produce with them.

Before you open the demo, move it to a Pro Tools audio drive. You will not be able to play it otherwise.

Online help


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chapter 13

The SoundReplacer Plug-In

SoundReplacer is an AudioSuite Plug-In de-signed to replace or augment audio ele-ments such as drums, percussion, and sound effects in Pro Tools tracks with alter-nate sounds. SoundReplacer can quickly and intelligently match the timing and dy-namics of the original performance mate-rial, making it ideal for both music and au-dio post production applications.

SoundReplacer features:

• Sound replacement with phase-accurate peak alignment

• Intelligent tracking of source audio dy-namics for matching the feel of the orig-inal performance

• Three separate amplitude zones per au-dio event for triggering different replace-ment samples according to performance dynamics

• Zoomable waveform display for preci-sion threshold/amplitude zone adjust-ment

• Crossfading or hard-switching of re-placement audio in different amplitude zones for optimum realism and flexibil-ity

• Online help

System RequirementsSoundReplacer is designed for use exclu-sively with Pro Tools software and requires version 4.3.1 or higher of the Pro Tools ap-plication.

For the latest compatibility information, contact your local Digidesign dealer or visit Digidesign's website.

Audio Replacement: Then and NowReplacing audio elements during the course of a recording session is a fairly com-mon scenario. In music production it is of-ten done in order to replace or augment an element that lacks punch. In film or video post-production it is typically done to im-prove 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 which had distinct disadvantages. Delay lines, for example, support only a single replace-ment sample, and while they can track the amplitude of the source events, the replace-ment sample itself remains the same at dif-ferent amplitude levels. The result is static

Chapter 13: The SoundReplacer Plug-In 113

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and unnatural. In addition to these draw-backs, sample triggers are notoriously diffi-cult to set up for accurate timing.

Similarly, with MIDI triggered samplers, MIDI timing and event triggering are in-consistent, resulting in problems with phase and frequency response when the original audio is mixed with the triggered replacement sounds.

The SoundReplacer SolutionSoundReplacer solves these timing prob-lems by matching the original timing and dynamics of the source audio while provid-ing three separate amplitude zones per au-dio event. This allows you to trigger differ-ent replacement samples according to performance dynamics.

Each replacement sample is assigned its own adjustable amplitude zone. Variations in amplitude within the performance de-termine which sample is triggered at a spe-cific time. For example, you could assign a soft snare hit to a low trigger threshold, a standard snare to a medium trigger thresh-old, 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—avoid-ing the abrupt sound truncation that oc-curs on many samplers.

In addition to its usefulness in music projects, SoundReplacer is also an ex-tremely 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-writ-ten to the source audio track. Sample thresholds can be amplitude-switched be-tween the replacement samples, or ampli-tude crossfaded for seamless transitions.

SoundReplacer Parameters

Waveform Display

The waveform display shows the audio that you have selected for replacement. When you select audio on the source track, then open SoundReplacer, the audio waveform will automatically be displayed here.

SoundReplacer

The waveform display with trigger markers shown

Once the audio selection is displayed, you can load the desired replacement samples and adjust their trigger thresholds while viewing the waveform peaks. Trigger mark-ers then appear in the waveform, indicat-ing the points at which the samples will be triggered. The color of each marker indi-cates which threshold/replacement sample will be triggered. The Zoomer (covered later in this section) allows you to increase or de-crease waveform magnification here to help accurately set trigger thresholds.

If you change your audio selection on the source track, you must click Update to up-date the waveform display. If Auto Update is on, SoundReplacer automatically up-dates the waveform display each time you make a new selection or begin playback.

✽ If you are often changing selections or start-ing/stopping playback, you may wish to turn off Auto Update to prevent too-frequent re-draws, which can be time consuming.

Trigger Threshold

The color-coded trigger threshold sliders al-low you to set a total of three amplitude

The threshold controls

zones (one for each replacement audio file) for triggering replacement samples:• The yellow slider represents amplitude

zone 1, the lowest-level trigger. • The red slider represents amplitude zone

2, the middle-level trigger. • The blue slider represents amplitude

zone 3, the highest-level trigger.

With a replacement sample loaded, drag the threshold slider to the desired ampli-tude level. Color-coded trigger markers will appear in the Waveform at points where the source audio signal exceeds the thresh-old set for that amplitude zone. The re-placement sample will be triggered at these points.

The color of the trigger markers correspond to the matching threshold slider. This al-lows you to see at a glance which replace-ment samples will be triggered and where they will be triggered.

▲ If you zoom the waveform display below a specific trigger threshold slider’s amplitude zone, the slider will be temporarily unavailable. To access the slider again, zoom back out to an appropriate magnification level.

Load/Unload Sound

Clicking the Load/Unload Sound icons al-lows you to load or unload replacement samples for each of the three trigger thresh-old amplitude zones. Clicking the floppy

The Load/Unload Sound icons


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disk icon loads a new sample (or replaces the current sample). Clicking the trash can icon unloads the current sample.

To audition a replacement sample before loading it into SoundReplacer, use the Im-port Audio command in Pro Tools. Once you have located and previewed the de-sired audio file, you can then load it into SoundReplacer using the Load/Unload Sound icons.

▲ SoundReplacer does not load regions that are part of larger audio files. To use a region as a replacement sample, you must first save it as an individual audio file.

The Zoomer

The Zoomer allows you to increase or de-crease magnification of the waveform data currently visible in the center of the wave-form display so that you can more accu-rately set sample trigger thresholds. • To zoom in on amplitude, click the

Up Arrow.• To zoom out on amplitude, click the

Down Arrow.• To zoom in on time, click the

Right Arrow.• To zoom out on time, click the

Left Arrow.

The Zoomer


▲ If you zoom the waveform display below a specific threshold slider’s amplitude zone, the slider will be temporarily unavailable. To ac-cess the slider again, zoom back out to an ap-propriate magnification level.

CrossfadeWhen Crossfade is on, SoundReplacer crossfades between replacement audio files in different amplitude zones. This helps smooth the transition between them.

When Crossfade is off, SoundReplacer hard switches between replacement audio files in different amplitude zones.

Crossfading is particularly useful for add-ing 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.

Peak AlignWhen Peak Align is on, SoundReplacer will align 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 will align the beginning of the replacement file with the trigger threshold point.

Depending on the characteristics of your source and replacement audio files, turning Peak Align on or off can significantly affect the timing of audio events in the replace-ment file. It is essential that you choose the option most appropriate to the material that you are replacing.

☞ For details on using Peak Align, See “Get-ting Optimum Results With SoundReplacer” on page 119.

UpdateWhen you click Update, the waveform dis-play is redrawn, based on the audio cur-rently selected on the source track. Each time you make a new selection on a source track, you must click Update for Soun-dReplacer to draw the waveform of the se-lection.

Auto UpdateWith Auto Update on, SoundReplacer au-tomatically updates the waveform display each time you make a new selection on a source track. This may not be desirable if you are frequently changing selections, however, since redrawing the waveform can be time consuming.

MixThe Mix control allows you to adjust the mix of the replacement audio file with the original source file. Higher percentage val-ues weight the mix toward the replacement audio. Lower percentage values weight the mix toward the original source audio.

The Mix button provides a quick means of toggling on/off the Mix control. When Mix is toggled off, the balance is instantly set to 100% replacement audio.

✽ Setting Mix to 50% and clicking Preview al-lows you to audition source audio and replace-ment audio together to check the accuracy of replacement triggering timing.

DynamicsThe Dynamics slider allows you to control how closely the audio events in the re-placement file track the dynamics of the source file:

◆ Setting the ratio to 1.00 matches the dy-namics of the source file.

◆ Increasing the ratio above 1.00 expands the dynamic range so that softer hits are softer, and louder hits are louder. This is useful if the source material lacks variation in its dynamic range.

◆ Decreasing the ratio below 1.00 com-presses 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/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.

Online Help

Online help


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Using SoundReplacerThe instructions that follow provide basic guidelines for using SoundReplacer effec-tively. To learn how to get the most out of SoundReplacer, also See “Getting Opti-mum Results With SoundReplacer” on page 119.

To use SoundReplacer:

1 On the source track, select the audio that you want to replace. Only audio that is se-lected will be replaced.

2 Choose SoundReplacer from the Audio-Suite menu.

3 Click the Load Sound icon to import the replacement sound for the first amplitude zone (the icon beneath the yellow slider).

4 Locate the desired audio file and click Open.

5 Adjust the amplitude zone slider.

6 Repeat steps 3–5 to load replacement sounds into amplitude zones two and three.

✽ Even if you only use a single replacement sample, for optimum results, set all three am-plitude zones. This will ensure the most accu-rate triggering. For details, See “Mapping The Same Sample Into Multiple Amplitude Zones” on page 120.


7 To align the amplitude peak in the re-placement file(s) to threshold trigger mark-ers in the source audio, enable Peak Align.

8 Click Preview to audition the replace-ment audio.

9 Adjust the threshold slider(s) to fine tune audio replacement triggering.

10 Adjust the Dynamics slider to fine tune how SoundReplacer tracks and matches changes in the source audio’s dynamics.

11 Adjust the Mix slider to get the desired balance between replacement audio and source audio.

To begin AudioSuite processing:

1 Adjust the AudioSuite file parameters. These settings will determine how the file is processed and what effect the processing will have on the original regions. Here are some guidelines:

◆ To process the selected region only in the track in which it appears, choose Playlist from the Selection Reference pop-up.

– or –

To process the selected region in the Audio Regions list only, choose Region List from this pop-up.

◆ To process and update every occurrence of the selected region throughout your ses-sion, enable the Use In Playlist button (and also choose Region List from the Selection Reference pop-up).

– or –

If you do not wish to update every occur-rence of the selected region, disable the Use In Playlist button.

◆ If you have selected multiple regions for processing and wish to create a new file that connects and consolidates all of these

regions together, choose Create Continu-ous File from the File Mode pop-up menu.

▲ SoundReplacer does not allow destructive processing, so the Overwrite Files option is not available in the File Mode Selector.

2 From the Destination Track pop-up, choose the destination for the replacement audio.

3 Click the Process button.

Getting Optimum Results With SoundReplacerGetting optimum results with Soun-dReplacer 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 that you do just that.

Using Peak AlignProper use of the Peak Align feature can sig-nificantly improve the results of sound re-placement. 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:

◆ Turn on Peak Align if you are replacing drum or percussion sounds whose peak level occurs at the initial attack.

◆ Turn off Peak Align if you are replacing sounds whose peak level occurs somewhere after the initial attack. Peak Align should

also be turned off if the sounds you are re-placing are not drum/percussion sounds.

To illustrate why Peak Align makes a differ-ence, look at the following illustrations:

Figure 1 shows a kick drum whose peak level occurs at its initial attack.

Figure 2 shows a kick drum whose peak level occurs after its initial attack.

If you turn on Peak Align and attempt to re-place the fast-peaking kick with the slow-peaking kick (or vice-versa), SoundReplacer will align their peaks—which occur at dif-ferent points in the sound. The audible re-sult would be that the replacement audio file (slow-peaking kick) would trigger too early.

Figure 1. a fast-peaking kick drum

Figure 2. a slower-peaking kick drum


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Mapping The Same Sample Into Multiple Amplitude ZonesIf you are performing drum replacement and intend to use just a single replacement sample, mapping it into multiple ampli-tude zones will ensure more accurate trig-gering. 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 rap-idly followed by the denser “attack” por-tion of the sound, where most of sound’s weight is.

With a sound like this, using a single am-plitude threshold presents a problem be-cause 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.

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 re-

A kick drum with a pre-hit preceding a denser attack


placement sample too early. This results in a replacement track with faulty timing.

The best way to avoid this problem is set multiple threshold zones for the same sam-ple using a higher threshold for the louder hit. Soft hits will trigger threshold 1 and louder hits will trigger threshold 2.

To set the precise threshold for louder hits, you may need to zoom in carefully to ex-amine the waveform for trigger points (in-dicated 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 dy-namics of the source audio, you may need to use all three trigger thresholds/ampli-tude zones for optimum results.

A single low threshold will cause the second, louder kick to trigger too early, as evidenced by the trigger marker at the very start of the waveform.

Using a second, higher threshold for the louder kick will make it trigger properly, as shown by the now properly-aligned trigger marker.

✽ If only one replacement sample is loaded into SoundReplacer and it is loaded into Trig-ger threshold/amplitude zone 1 (yellow), Soun-dReplacer will allow you to 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 Replacement FilesIf you often use the same librarian settings and replacement sounds in sound replace-ment sessions, SoundReplacer provides a convenient feature that makes it easy to keep the replacement audio files and librar-ian settings linked together.

When you choose a saved setting from the Librarian menu, SoundReplacer always looks for the replacement audio files associ-ated with the setting. SoundReplacer first looks in the audio file’s original hard disk location (at the time you saved the setting.)

If it’s not there, it looks in a special folder named Audio Files within SoundReplacer’s Plug-In Settings folder. (DAE > Plug-In Set-tings > SoundReplacer > Audio Files).

If SoundReplacer finds the replacement au-dio file there, the settings file will load with the associated audio.

By making it a rule to put replacement au-dio files (or a copy of them) in this special folder, you can freely exchange Soun-dReplacer settings—and the audio files as-sociated with them—with other users.

▲ Do not create subfolders within Soun-dReplacer’s Audio Files folder. Files located within subfolders will not be recognized.

SoundReplacer Demo Session SoundReplacer includes a demo session that illustrates useful applications of this Plug-In. The librarian settings featured in the demo session are automatically in-stalled with SoundReplacer.

Before you start:

1 Move the SoundReplacer demo from the Pro Tools folder to a Pro Tools audio drive.

2 Open the Demo session.

3 Choose SoundReplacer from the Audio-Suite menu.

4 Position SoundReplacer on the bottom-left hand corner of the screen, so that you can clearly see the top audio track, as well as the Memory Locations Window.

5 Begin playback of the Demo session. A voice-over will guide you through the basic use of SoundReplacer.

Example 1This example illustrates basic sound re-placement on a kick drum track.

In the Demo session:

1 Click memory Location #5, Kick Replace 1. A kick drum source track is se-lected and centered on screen.

2 Choose Kick Double Trigger Zone from the Librarian menu.


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3 Click Update.

4 Use SoundReplacer’s Zoomer to zoom out to see several of the kick drum hits.

5 Click Preview to hear the replacement sound triggered with the source kick track.

6 Adjust the wet/dry mix with the Mix slider. Turning off the Mix button allows you to eliminate the dry sound entirely from the wet/dry mix.

7 Click Process to write the replacement file to disk on the same track.

8 Alternatively, write the replacement file to a different track by choosing a destina-tion from the Destination Track pop-up menu. An empty audio track has been pro-vided in the Demo Session for this purpose.

Example 2This example demonstrates the use of mul-tiple trigger zones using the same replace-ment sound. This allows a phase-coherent sound replacement without unwanted re-sults such as double-attack flams, comb fil-ter-like effects or mistriggered samples.

Optimum results are achieved by carefully adjusting the two threshold zones so that softer kick drum hits (the yellow zone) and louder hits (the red zone) trigger at exactly the right points.

Loading a replacement sound into the yel-low amplitude zone automatically loads it into the red and blue amplitude zones as well. This allows more subtle adjustment and control of triggering.

The example also illustrates why it is useful to turn off the Peak Align option in cases where the signal peak in a sound occurs af-ter the initial attack.



1 With SoundReplacer’s Zoomer, zoom in closely on one of the yellow trigger markers in the waveform display. These vertical lines show exactly where triggers occur.

2 Adjust the yellow Trigger Threshold slider and observe how it affects the place-ment of these trigger points and markers.

3 Click the Compare button to return to the original settings.

4 Click Memory Location #5.

5 From the Librarian menu, choose 808 Trigger No Peak Align. This kick drum sound has a strong “boom” component that occurs after the initial attack transient, making it louder than the actual attack it-self.

6 Click Preview, then toggle Peak Align on and off to compare the results. (Give the preview a few moments to update each time you do this.)

You should clearly hear a difference in the timing of the replacement sound. When Peak Align turned on, the replacement sound plays out of sync because the initial attack of the sound is not the actual signal peak—the subsequent “boom” is.

Example 3This example demonstrates sound replace-ment for special effects. The replacement audio file consists of a clave hit with back-wards reverb preceding it and a normal re-verb decay following it.

It illustrates how Peak Align can be used to back time a replacement sound to match the attack of the source file.

In this example, Peak Align is turned on. This forces SoundReplacer to find the peak in the replacement file (the clave hit), and align it with the kick drum hits on the source track.


1 From the Librarian menu, choose Reverse Sound Example.

2 Click Update.

3 Click Preview.

The replacement sound should accurately line up with the original hit points.

Example 4This example demonstrates two different snare sounds triggered from two amplitude zones.


1 Click Memory Location #6, Snare Re-place 1.

2 Click Update.

3 From the Librarian menu, choose Snare dbl. Trigger zone.

4 Click Preview.

The replacement sound should switch from a medium snare hit to a rim shot at the appropriate points.

Example 5This example demonstrates how the Cross-fade option can be used to smooth the transition between replacement sounds triggered in succession.

In this example, there are five hits that vary in volume from very soft to very loud. All

three amplitude zones are used, with differ-ent replacement files for each zone.


1 Zoom the waveform display out and compare the yellow trigger points with the red trigger points.

2 Click Memory Location #7, Gradual Snare. A sample snare drum source track containing several hits that vary in volume from very soft to very loud is selected and centered on-screen.

3 Click Update and zoom out to show all five drum hits in the waveform display.

4 From the Librarian menu, choose Snare Triple Trigger Zone.

5 Click Preview.

6 Preview the result.

With Crossfade enabled, SoundReplacer smoothly crossfades between the softest sound and the loudest sound. If Crossfade is disabled, SoundReplacer will hard switch on each zone.


124

appendix a

DSP Requirements for TDM Plug-Ins

The table below shows the total number of instances of each Digidesign TDM Plug-In that can be powered by a single DSP chip on the two types of Digidesign TDM-based DSP cards (the Pro Tools 24 MIX card and the DSP Farm). DSP usage differs according to card type.

Though there are a total of 6 DSP chips on a Pro Tools 24 MIX card, certain Plug-Ins, such as the Mod Delays, require a specific type of DSP chip in order to function. The table indicates the number of compatible chips per card for each Plug-In.

Table 1. TDM Plug-In DSP Usage

MIX Core/Farm (Pro Tools 24 MIX systems)DSP Farm

(Pro Tools 24 and Pro Tools III systems)

Plug-In Instances per chip on MIX card (mono/stereo)

Available Chips per MIX card

Instances per chip on DSP Farm(mono/stereo)

Available Chipsper DSP Farm

D-Fi(Lo-Fi)

4/2 6 3/1 4

D-Fi(Recti-Fi)

8/4 6 3/1 4

D-Fi(Sci-Fi)

4/2 6 3/1 4

D-Verb 2/2 3 1/1 4

DINR(BNR)

1/1 3 1/1 4

DINR(Hum Removal)

not compatible 0 4/2 4

DPP-1 1/1 3 1/1 4

Maxim 8/4 3 4/2 4

Appendix A: DSP Requirements for TDM Plug-Ins 125

126

Focusrite d2(1-2-band)

24/12 6 12/6 4

Focusrite d2(4-band)

12/6 6 6/3 4

Focusrite d2(6-band)

6/3 6 3/1 4

Focusrite d3(comp+limiter)

7/7 6 3/3 4

Focusrite d3(comp/limiter)

14/12 6 6/6 4

Bruno/Reso 1/1 3 1/1 4

Table 1. (Continued) TDM Plug-In DSP Usage

MIX Core/Farm (Pro Tools 24 MIX systems)DSP Farm

(Pro Tools 24 and Pro Tools III systems)

Plug-In Instances per chip on MIX card (mono/stereo)

Available Chips per MIX card

Instances per chip on DSP Farm(mono/stereo)

Available Chipsper DSP Farm


▲ This table shows theoretical maximum performance when no other Plug-Ins are sharing avail-able DSP resources. You will typically use more than one type of Plug-In simultaneously.

appendix b

DSP Delays Incurred by TDM Plug-Ins

This table shows the delays inherent in each type of Digidesign TDM Plug-In. De-lays matter only if you use a real-time TDM Plug-In on one channel of a stereo or mul-tichannel signal but not the others. This can cause the channels to be out of phase.

Table 2. Samples of delay incurred by each TDM Plug-I

Ap

You can use the DigiRack TimeAdjuster Plug-In to compensate for these delays. See the DigiRack Plug-Ins Guide for more infor-mation.

☞ See Appendix A of the Pro Tools Reference Guide for details on DSP-induced delays.

n on MIX and DSP Farm cards

Plug-In Samples of delay on MIX cards Samples of delay on DSP Farm cards

Bruno 3 3

Lo-Fi 3 3

Recti-Fi 3 3

Sci-Fi 3 3

D-Verb 3 3

DINR (BNR) 1538 1538

DINR (Hum Removal) n/a 3

DPP-1 3 3

Focusrite d2 (1-2 band) 5 3

Focusrite d2 (4 band) 5 3

Focusrite d2 (6 band) 5 3

Focusrite d3 (Comp+Lim) 5 3

pendix B: DSP Delays Incurred by TDM Plug-Ins 127

128

Focusrite d3 (Comp/Lim) 5 3

Maxim (TDM and RTAS) 1027 1027

Reso 3 3

Table 2. (Continued) Samples of delay incurred by each TDM Plug-In on MIX and DSP Farm cards

Plug-In Samples of delay on MIX cards Samples of delay on DSP Farm cards


Numerics1-2 Band EQ 514-Band EQ 516-Band EQ 51

AAcoustic Environments 45Adaptive resolution parameter 27ADSR Envelope Generator 101, 108Air conditioner rumble 65Alesis ADAT XT 20 89Algorithm parameter 42, 47Aliasing artifacts 26All (harmonics) parameter 106Alternating Rectification 30Alt-Max Rectification 31Ambient algorithm 43, 47Amplitude

envelope 101parameters 101, 107

Analyze Button 22Anechoic chamber 45Anti-Alias Filter 26Attack parameter 62, 102Attenuation parameter 88AudioSuite

buffer size 23menu 17processing 24processing preferences 22using AudioSuite Plug-Ins 23

Auto Fit button 70Auto release button 62Auto Update button 115, 117Automating Plug-Ins 12

BBend Range parameter 102, 109Bit Resolution parameter 89BNR 65Brick wall limiter 58Broadband Noise Reduction 65

parameters 67Bruno

features 93parameters 98

Bruno/Resoand MIDI 96demo session 111

Buffer Sizesetting for AudioSuite 23

Bypass Button 10

CCartesian Graph 53Ceiling parameter 88Cents 83, 110Chorus 37Church algorithm 42, 47Clip Indicator 46, 82Clipping levels 7Clock parameter 100Coarse parameter 82Compression ratio 58Compressor

controls 61in/out icon 61

Compressor+Limiter 57Compressor/Limiter 57Contour Line 66, 67

editing 73Copy Settings command 13CPU

index

Index 129

130

Load meter 8power reqirements for RTAS Plug-Ins 8

Create Continuous File command 19Create Individual Files command 19Creating a settings subfolder 16Crossfade

frequency 99parameter 83, 99, 116

Cross-Feedback parameter 41Cross-synthesis 93Customizing a Reverb 48

DD2

configurations 51filters 53input controls 52meters 53parameters 52

D3Compressor 58Limiter 58meters 60parameters 59

DAEallocating memory 6

Damping 105Damping Amount parameter 105Damping Velocity parameter 106

Decay parameter 43, 48, 102Delay

DSP-induced delay 8, 127parameter 38, 40, 82

Delete Current Settings File command 13Demo Session 32Detune Amount parameter 103, 109Detune Velocity parameter 103, 109D-Fi Demo Session 32D-fx

D-fx Chorus 37D-fx D-Verb 41D-fx Flanger 38D-fx Multi-Tap Delay 39D-fx Ping-Pong Delay 40

Diffusion parameter 43, 48Digital reverberation 45Distortion/Saturation parameters 27


DitherDefault for AudioSuite 22Depth for AudioSuite 23parameter 89

Down-processing audio 26Downward expanders 66DPP-1 Parameters 81Drum limiting 86DSP

requirements for Bruno/Reso 94requirements of TDM Plug-Ins 5, 7

DSP Farm Cards and Bruno/Reso 94DSP Manager 6DSP Usage window 8DSP-induced delays 9D-Verb 41, 45Dynamic Audio Signal Modeling 65Dynamic Range

of a mix 86of individual instruments 86

Dynamics parameter 117

EEarly Reflections 46Effect Amount parameter 28Effect Frequency parameter 28Enable Automation Parameters Button 10Entire Selection mode 19Envelope Follower 29, 110Envelope Generator 101EQ

bypassing 53curves 53disabling 53enabling 53filter controls 53

External Key parameter 100, 106

FFactory Default setting

creating 16Feedback parameter 38, 39, 40, 82File Mode Selector 19Fine parameter 83Fit button 67, 69Flanger 38

Focusrite D2 Plug-In 51Focusrite D3 Plug-In 57Follower parameter 110Freak Mod 28FreeMIDI software 96Frequency (filter) parameter 110

GGain Amount parameter 101, 107Gain parameter 40Gain reduction meter 60Gain Velocity parameter 101, 108Glide parameter 102

HHall algorithm 42, 47Hard switching 100, 116Harmonics 105Harmonics parameter 106Hi Frequency Cut parameter 43, 48Hi Shelf EQ parameter 69High-mid Peak Filter 54High-pass Filter 53High-Pass Filter parameter 39High-shelf EQ 66High-shelf Filter 54Histogram 85, 87Host processing 2Hum Removal 65

IImport Settings command 13Importing Plug-In settings 15Input Level parameter 37, 41, 52, 87Input Trim 28Insert/Send Selector 9Inserts 7

mono & stereo 7Inserts/Sends Editor 9

KKey Input 11, 22, 98, 100, 106

filters 11Key Input Selector 9

Key Listen parameter 100Key On/Off parameter 60Keyboard Input

for Plug-In parameters 10, 95Keyboard Shortcuts 10, 94

LLatch bar 96Late reverberation 46Learn button 69Learn First Audio Mode 69Learn Last Audio Mode 69Left Channel button 56LFO (Low Frequency Oscillator) 29LFO Rate parameter 38, 39LFO Waveform parameter 38LFO Width parameter 38, 39Librarian menu 9, 13Limit LED 63Limiter 86

controls 62in/out icon 63

Limitinga mixdown 86drums 86individual instruments 86

Linear resolution parameter 27Link button 56, 88, 90Link Mode 56Load/Unload Sound icons 115Lo-Fi 25, 26

examples 34Low Frequency Oscillator 29Low-mid Peak Filter 54Low-Pass Filter 55Low-Pass Filter parameter 38, 43, 48Low-shelf Filter 54LPF (Low Pass Filter) 110

parameters 110

MMaster Tune parameter 103, 109Maxim 85

parameters 87peak levels 85signal delay 87

Index 131

132

signal peaks 86Maximum Pitch parameter 83MIDI

and Bruno/Reso 96beat clock 100MIDI-triggered samplers 114velocity 99

MIDI Clock parameter 107Minimum Pitch parameter 83Mix parameter 31, 38, 40, 88, 101, 117Mod Amount/Mod Rate 29Mod slewing slider 29Modulation Type parameter 28Mono Inserts 7Mono voice mode 103MultiShell compatible Plug-Ins 6Multi-Tap Delay 39

NNegative Excursion 31Negative Rectification 30Noise Contour line 66Noise Generator 27Noise Reduction Amount parameter 68Noise Reduction limitations 66Noise Shaping parameter 89Noise Signature 66, 67Nonlinear algorithm 43, 47

OOdd (harmonics) parameter 106OMS 96Online Help 85, 104, 117On-Screen Keyboard 95, 99, 104Output Level parameter 52Output Meter 46Output parameter 88Overwrite file mode command 19

PPan parameter 40Paste Settings command 13Peak Align 119Peak Align parameter 116Peak From All Tracks


mode 20Peak limiting 85, 86Peak On Each Track

mode 20Perfect attack-limiting 85Ping-Pong Delay 40Pitch

processing 81transposition 81

Pitch Parameters 102, 109Pitched Noise 65Plate algorithm 42, 47Plug-In Selector 9, 17, 18Plug-In Settings

copying 15creating user default 15importing 15pasting 15saving 14

Plug-In Settings Librarian 12, 22Plug-Ins

as Inserts 7automating 12

Poly voice mode 103Portamento 102, 109Positive Excursion 31Positive Rectification 30Post-Filter 31Preamp noise 65Pre-delay 46Pre-Delay parameter 43, 48Pre-Fader Insert Operation 7Pre-Filter 30Pro Tools LE software 2Processing

AudioSuite 20preferences 22stereo 21

QQ Parameter 110Quantization Noise 89Quantization parameter 27

RRAM and DAE memory 6

Ratio parameter 61, 83Real-Time AudioSuite Plug-Ins 2Real-Time Plug-Ins 7, 10Recti-Fi 25, 30

examples 35Rectification 30Region by Region mode 19Relative Pitch Entry parameter 82Release parameter 62, 68, 88, 102Replacing audio elements 113Reso


Resonance (Q) parameter 110Resonance Amount parameter 105Resonance Velocity parameter 105Resonant peak 110Resonator 28, 93, 105Response parameter 68Reverb 45

character 45creation 48customizing 48

Right Channel button 56Ring Mod parameter 28Ring modulation 27, 28RMS detector 58Root Plug-In Settings folder 13RTAS Plug-Ins 2

and CPU power requirements 8and signal delays 9defined 2

SSample Rate parameter 26Sample Size parameter 26Sample triggers 114Sample-and-Hold 29Saving Plug-In Settings 14

Save Plug-In Settings To command 14Save Settings As command 13Save Settings command 13

Sci-Fi 25, 27examples 32

Scroll Left/Right buttons 70Selecting Tracks for AudioSuite Processing 23

Selection Reference Selector 18Semitones 109Set As User Default command

to create default plug-in settings 15Set Plug-In Default To command 14Set Root Settings Folder command 14Settings 9, 22

creating Library subfolders 16menu 13Settings Preferences command 14

Show DSP Usage command 5Side Chain Input Pop-up Menu 22Side chain processing 9, 11, 58, 98Signal delay

and Maxim 87and RTAS Plug-Ins 9

Signal Present Indicator 81Size parameter 47Slewing 29Slow Down parameter 31Smoothing 83Smoothing parameter 68Sony PCM-9000 89SoundReplacer 113, 118


Spectral Graph 67Speed Up parameter 31Spread parameter 101Stereo Inserts 7Stereo spread 101Subharmonic synthesis 31Sum Inputs button 37, 50Super Fit button 70Sustain Level parameter 102Switch parameter 99

TTape Hiss 65TDM Plug-Ins 2

and DSP power requirements 8Threshold parameter 62, 63, 88, 100, 107Threshold slider 90Thru-zero flanging 38Timbre Parameters 99, 105Timbrometer 100

Index 133

134

TimeAdjuster Plug-In 9Time-slicing 93Toggle (harmonics) parameter 106Track Selector 9triangle wave 29Trigger and Hold 29Trigger markers 115Trigger Threshold 115

UUndo button 71Undo/Redo 21Update button 115, 117Use In Playlist Off command 18User Default

creating 15

VVari-Fi 25, 31Virtual nodes 96Voice Mode parameter 103Voice Parameters 103, 110Voice Polyphony 94Voice Stack parameter 104, 111Voice Stacking 109

WWah-wah effect 110Wave sequencing 98Waveform Display 114Wet/Dry parameter 42, 46

XX axis of histogram 87

YY axis of histogram 87

ZZero crossing 31Zoom Out/In buttons 70Zoomer 115, 116


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