Digidesign Plug-Ins Guide Version 6.0 for TDM or LE Systems on Macintosh Version 5.3.x for TDM or LE Systems on Windows or Macintosh Version 5.1.x for MIX or LE Systems on Macintosh Digidesign 2001 Junipero Serra Boulevard Daly City, CA 94014-3886 USA tel: 650·731·6300 fax: 650·731·6399 Technical Support (USA) 650·731·6100 650·856·4275 Product Information (USA) 650·731·6102 800·333·2137 International Offices Visit the Digidesign Web site for contact information Web Site www.digidesign.
Copyright This guide is copyrighted ©2002 by Digidesign, a division of Avid Technology, Inc. (hereafter “Digidesign”), with all rights reserved. Under copyright laws, this guide 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.
contents Chapter 1. Getting Started With Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 Plug-In Formats. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2 Installing Plug-Ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Chapter 4. D-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31 Lo-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32 Sci-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34 Recti-Fi . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Chapter 9. Maxim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75 About Peak Limiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76 Maxim Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77 Using Maxim . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
vi Digidesign Plug-Ins Guide
chapter 1 Getting Started With Plug-Ins Digidesign plug-ins provide a comprehensive set of digital signal processing tools for professional audio production. This guide explains the use of each of the plugins currently available from Digidesign.
Plug-In Formats Plug-ins are special-purpose software components that add additional signal processing functionality to Pro Tools. There are three formats of plug-ins: • TDM plug-ins (real-time, DSP-based) • RTAS plug-ins (real-time, host-based) • AudioSuite plug-ins (non-real-time, file-based processing) TDM Plug-Ins (TDM Systems Only) TDM plug-ins function as track inserts, are applied to audio during playback, and process audio non-destructively in real time.
Demo Plug-Ins and Authorization Some plug-ins require an iLok USB hardware key present in order to run in demo mode. Demo authorization for such plug-ins can only be installed over the internet from Digidesign’s Web site (www.digidesign.com) while you launch Pro Tools. Make sure that your Pro Toolsequipped computer is internet-ready and that your iLok USB key is inserted in an available USB port before you launch Pro Tools to complete this automated process.
Authorizing Plug-Ins 5 Insert the License Card into the iLok. You will be able to visually verify that the License Card makes contact with the iLok’s metal card reader. iLok Digidesign plug-ins are authorized using the iLok USB hardware key from PACE Anti Piracy. One iLok hardware key is included with your HD Core system. This key can hold over 100 authorizations for all your iLok-enabled software. Once iLok is authorized for a given piece of software, use iLok to authorize that software on any computer.
Macintosh OS 9 To allocate additional memory to Pro Tools: To remove an expired plug-in: 1 Start Pro Tools so that it can calculate its basic memory allocation. 1 Open the DAE folder inside your System Folder. 2 Open the Plug-Ins folder or Plug-Ins (Unused) folder and drag the expired plug-in to the Trash. 3 Empty the Trash. Windows To remove an expired plug-in: 1 From the Start menu, choose Settings > Control Panels and double-click Add/Remove Programs.
Conventions Used in This Guide All Digidesign guides use the following conventions to indicate menu choices and key commands: : Convention Action File > Save Session Choose Save Session from the File menu Control+N Hold down the Control key and press the N key Option-click Hold down the Option key and click the mouse button The following symbols are used to highlight important information: User Tips are helpful hints for getting the most from your Pro Tools system.
chapter 2 Working with Real-Time Plug-Ins Real-time plug-ins process audio non-destructively 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 rely on the processing power of Digidesign DSP cards. TDM plug-ins run only on TDM-equipped Pro Tools systems. RTAS Plug-Ins rely on the processing power of your computer. RTAS plug-ins run on Pro Tools TDM and LE systems.
DSP Sharing Between TDM Plug-Ins Digidesign’s MultiShell II technology allows different types of TDM plug-ins to share the same DSP chip at the same time. This allows you to simultaneously use a greater variety of plug-ins by efficiently managing the DSP available on each chip in your system. In order to take advantage of this capability, plug-ins must be MultiShell II compatible. For example, D-Fi and Focusrite D3 plug-ins are fully MultiShell II compatible.
CPU Usage Limit The CPU Usage Limit controls the percentage of CPU resources allocated to Pro Tools host processing tasks. The default CPU Usage Limit setting is 40 percent. On single-processor computers, the maximum CPU Usage Limit is 85 percent. On dual-processor computers, the maximum CPU usage limit is 99 percent.
Plug-Ins as Inserts Real-time plug-ins are available as in-line inserts on audio tracks, Auxiliary Inputs, and Master Faders. 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. If you are using both TDM and RTAS plugins on the same track, RTAS plug-ins must occur first in the signal chain, followed by TDM plug-ins.
Mono Plug-Ins Are designed for use on mono tracks. Some mono plug-ins (such as Mod Delay) allow you to generate a stereo output from a mono channel. Plug-ins that occur on a track after a stereo plug-in are used in stereo as well. Multi-Mono Plug-Ins Are designed for use on stereo or greater-than-stereo multichannel tracks only when a multichannel version of the plugin is not available. Multi-mono plug-ins will analyze and process each channel independently without regard to each other.
To insert a plug-in on a track: Click the Insert Selector on the track and select the plug-in that you want to use. ■ click here Moving a plug-in Dragging an insert on top of an existing insert will replace it. To duplicate an insert: Inserting a plug-in ■ Option-drag (Macintosh) or Alt-drag (Windows) the insert to the desired location. The duplicated plug-in retains its original settings and automation. To remove an insert from a track: ■ Click the Insert Selector and choose No Insert.
Master Link button Link Enable buttons Channel Selector Effect Bypass Disables the currently displayed plug-in. This allows you to compare the track with and without the effect. Auto Allows you to enable individual plug-in parameters for automation recording. See “Automating Plug-Ins” on page 17. Safe When enabled, prevents existing plug-in automation from being overwritten. Plug-In window (multi-mono Focusrite D3 shown) Settings Menu Allows you to copy, paste, save, and import plug-in settings.
LFE Enable Enables plug-in processing of the LFE (low frequency effects) channel on a multichannel track formatted for 5.1, 6.1, or 7.1 surround formats. To disable LFE processing, deselect this button. For more information on the LFE channel, refer to the Pro Tools Reference Guide. Using the Plug-In window to select a different plug-in on the same track: Click the Insert Selector and select a plug-in from the pop-up menu. ■ To bypass a plug-in: ■ Click the Plug-In window’s Bypass button.
To open a Plug-In window for a specific plug-in: In the Mix window, Shift-click the Insert button of the desired plug-in. ■ To open Plug-In windows for each channel of a multi-mono plug-in: Option-click (Macintosh) or Alt-click (Windows) the Channel Selector in the Plug-In window of the desired plug-in. Keyboard Input for Plug-In Parameters You can use your computer keyboard to edit plug-in parameters.
If necessary, you can unlink plug-in controls on specific channels of a track and edit them independently. You can also selectively link the controls of specific channels. Master Link button Link Enable buttons Channel Selector Channel Selector and Link controls A typical use for this feature is to control the dynamics of one audio signal using the dynamics of another signal (the Key Input). A kick drum track, for example, can be used to trigger gating of a bass track to tighten it up.
To use a Key Input for side-chain processing: To enable automation: 1 From the Key Input menu, choose the input or bus carrying the audio you want to use to trigger the plug-in. 1 Open the Plug-In window for the plug-in you 2 Click External Key to activate side-chain pro- to automate, and click Add, then OK. want to automate. 2 Click the Auto button, choose the parameters cessing. 3 To hear the audio source you have selected to control side-chain input, click Key Listen.
To record automation: To disable automation record safing for a plug-in: 1 In the Automation Enable window, make sure that plug-in automation is write-enabled. ■ Deselect the Safe button on the desired plugin. For more information on creating and editing automation, see the Pro Tools Reference Guide. Using the Librarian Automation Enable window The Settings Librarian makes it easy to create your own library of plug-in settings.
The Settings Menu The Settings menu allows you to save, copy, paste, and manage plug-in settings files. Set As User Default Defines the current settings as the User Default for a specific plug-in. Settings Preferences The sub-menus that appear here are for choosing preferences for saving and importing plugin settings. They include: Set Plug-In Default To Sets the default setting to either Factory Default (the standard default setting for that plug-in) or User Default (your custom setting).
If you choose Root Settings folder, Pro Tools saves to the Plug-In Settings folder within the DAE Folder unless you have specified a different location for the Root Settings folder. To select a different 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).” Managing Settings Use the Settings pop-up menu to manage settings.
Creating Subfolders for Settings To make it easier to find specific types of settings, you can subdivide settings files by creating subfolders for them. If you are using Reverb One, for example, you may want to organize settings by room size. This allows you to keep your library well organized with hierarchical menus for settings. Librarian menu with subfolders for settings To create a settings subfolder: 1 From the Settings pop-up menu, choose Save Settings.
22 Digidesign Plug-Ins Guide
chapter 3 Working with AudioSuite Plug-Ins 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 audio file or create an entirely new audio source file. In Pro Tools, AudioSuite plug-ins are accessed through the AudioSuite menu.
Plug-In Selector This menu allows you to select any AudioSuite Plug-In installed in your Plug-Ins folder. Region List Applies AudioSuite processing only to regions currently selected in the Audio Regions List. Processing will not be applied to regions located in tracks and playlists in the Edit window.
File Mode Selector Process Mode Selector This menu allows you to select whether or not the AudioSuite plug-in will process the selected audio destructively or non-destructively, and how the original files will be modified. If you have made a selection that includes multiple regions, this pop-up menu allows you to specify whether AudioSuite processing is performed on a region-by-region, or entire-selection basis.
Peak On Each Chan/Track Analyzes and processes each selected track individually. If you apply the Normalize plug-in to multiple tracks or a multichannel track in Peak On Each Chan/Track mode, each channel or track will be normalized independently, without regard to the other selected tracks. Peak On All Chans/Tracks Uses all currently selected channels and tracks for analysis.
Multichannel Processing Other AudioSuite Parameters Many Digidesign AudioSuite plug-ins allow up to 48 channels of audio to be processed simultaneously. In addition to the standard AudioSuite parameters, there are a number of special-purpose controls found on certain plug-ins: Undoing AudioSuite Processing If you have processed an audio selection nondestructively, the Undo and Redo commands allow you to undo the selected AudioSuite process. You can undo/redo an AudioSuite process during audio playback.
The AudioSuite Processing Preferences Dialog Before you begin using AudioSuite plug-ins, use the Preferences dialog (Setups > Preferences > Processing) to configure default AudioSuite parameters according to your needs. These parameters include the default dither setting, a dither on/off checkbox, and the AudioSuite buffer size parameter. Edit Settings Accesses options for the presently selected Default Dither plug-in.
TC/E Allows you to select the Time Compression Expansion plug-in and default settings used when editing with the Time Trimmer. Refer to the Pro Tools Reference Guide for more information about the Time Trimmer. Using AudioSuite Plug-Ins AudioSuite plug-ins can be applied to whole regions, partial regions, or selections that are made up of whole and partial regions across one or more tracks.
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 processed. 2 Choose the desired AudioSuite plug-in from the AudioSuite menu. 3 Click the Preview button to begin playback of the selected material. 4 Adjust the plug-in parameters to achieve the effect that you want.
chapter 4 D-Fi D-Fi consists of four separate plug-ins for TDM, RTAS, and AudioSuite. D-Fi plug-ins form a unique sound design tool kit for processing and deconstructing audio in several retro and synthesis-oriented ways.
Vari-Fi Vari-Fi provides a pitch-change effect similar to a tape deck or record turntable speeding up from or slowing down to a complete stop. Features include: Lo-Fi Lo-Fi down-processes audio by reducing its sample rate and bit resolution. It is ideal for emulating the grungy quality of 8-bit samplers. • Speed up from a complete stop to normal speed • Slow down to a complete stop from normal speed Vari-Fi is an AudioSuite-only plug-in.
Anti-Alias Filter The Anti-Alias filter works in conjunction with the Sample Rate control. As you reduce the sample rate, aliasing artifacts are produced in the audio. These produce a characteristically dirty sound. Lo-Fi’s anti-alias filter has a default setting of 100%, automatically removing all aliasing artifacts as the sample rate is lowered. This parameter is adjustable from 0% to 100%, allowing you to add precisely the amount of aliasing you want back into the mix.
Sci-Fi Sci-Fi is designed to mock-synthesize audio by adding effects such as ring modulation, resonation, and sample & hold, that are typically found on older, modular analog synthesizers. Sci-Fi is ideal for adding a synth edge to a track. Sci-Fi Input Trim Input Trim attenuates signal input level to the Sci-Fi processor.
To use keyboard note entry: 1 Option-click (Macintosh) or Windows-click (Windows) the Effect Frequency for the pop-up keyboard. extreme dynamic changes in your modulation source. This provides a smoother, more continuous modulation effect. The more slewing you add, the more gradual the changes in modulation will be. 2 Select the note on the keyboard that you want for the Effect Frequency.
If you select Envelope Follower as a modulation type, the Mod Rate slider changes to a Mod Slewing slider, which is adjustable from 0% to 100%. Output Meter The Output Meter indicates the output level of the processed signal. Note that this meter indicates the output level of the signal—not the input level. If this meter clips, the signal may have clipped on input before it reached Sci-Fi. Monitor your send or insert signal levels closely to prevent this from happening.
Alternating Rectification This alternates between rectifying the phase of the first negative waveform excursion to positive, then the next positive excursion to negative, and so on, throughout the waveform. The audible effect is a halving of the audio signal’s frequency, creating a subharmonic tone. and smooth out the sound. This Post-Filter is useful for filtering audio that contains subharmonics.
Vari-Fi Controls 2 Choose Windows > Show Memory Locations. Speed Up Speed Up applies a pitch-change effect to the selected audio, similar to a tape recorder or record turntable speeding up from a complete stop. The effect doesn’t change the duration of the audio selection. Slow Down Slow Down applies a pitch-change effect to the selected audio, similar to a tape recorder or record turntable slowing down to a complete stop. The effect doesn’t change the duration of the audio selection.
8 Listen to the effect. Note how Trigger and Hold is used to cause modulation to follow the amplitude. This provides a much more interesting type of modulation than standard envelope following. 9 Adjust the Mod Threshold to vary the modulation on 1/4 note accents. Wah Guitar Loop 1 Click memory location #3, “Wah Guitar Loop.” 2 Select Bypass to hear this loop without Sci-Fi processing. 3 Choose “Freq Mod Env. Follower Wah.” 10 Choose “Res. –16 note Trig & Hold.
5 Deselect Bypass to hear the effect. Recti-Fi Examples 6 Try each Lo-Fi setting with this loop. The examples that follow demonstrate Recti-Fi. Follow the instructions in each section below to hear useful applications for this plug-in. The loop has a hip-hop feel, and demonstrates how Lo-Fi can be used to create textures with hard percussive elements. Before you begin: Drum Kit Loop 1 Open the demo session. 1 Click memory location #2, “Drum Kit Loop.” 2 Choose Windows > Show Memory Locations.
3 Compare the sound of the processed and unprocessed audio using the Bypass button. Slam Kit Loop This setting demonstrates how to use sub-octave rectification to enhance low frequencies. 2 Choose “Trasho Kit.” 1 Click memory location #7, “Slam Kit Loop.” 3 Play the audio. Hat Loop 1 Click memory location #1, “Hat Loop.” This setting illustrates the use of Recti-Fi as a basic sound modifier for percussive sounds. 2 Choose “Noise Hat.
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chapter 5 D-Verb D-Verb is a studio-quality reverb provided in TDM, RTAS, and AudioSuite formats. The TDM version of the D-Verb plug-in is not supported at 192 kHz; use the RTAS version instead. About Reverb Digital reverberation processing can simulate the complex natural reflections and echoes that occur after a sound has been produced, imparting a sense of space and depth—the signature of an acoustic environment.
Reflected Sound In a typical concert hall, sound reaches the listener shortly after it is produced. The original direct sound is followed by reflections from the ceiling or walls. Reflections that arrive within 50 to 80 milliseconds of the direct sound are called early reflections. Subsequent reflections are called late reverberation. Early reflections provide a sense of depth and strengthen the perception of loudness and clarity.
Input Level The Input Level slider adjusts the input volume of the reverb to prevent the possibility of clipping and/or increase the level of the processed signal. Wet/Dry The Wet/Dry slider adjusts the balance between the dry signal and the effected signal, giving you control over the depth of the effect. This parameter is adjustable from 100% to 0%. Algorithm This control selects one of seven reverb algorithms: Hall, Church, Plate, Room 1, Room 2, Ambience, or Nonlinear.
fect the overall reverb density. High settings of diffusion can be used to enhance percussion. Use low or moderate settings for clearer and more natural-sounding vocals and mixes. Decay Decay controls 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.
Selecting Tracks for D-Verb AudioSuite Processing Because D-Verb adds additional 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.
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chapter 6 Focusrite D2 Focusrite D2 is a high-quality digital equalizer plug-in for Pro Tools. Developed in cooperation with Focusrite, the D2 is based on the highly-acclaimed Red Range 2™ dual EQ, designed by Rupert Neve.
D2 Parameters The rotary controls on the D2 Plug-in can be adjusted by dragging horizontally or vertically. Dragging upwards or to the right will increment the control. Dragging downwards or to the left will decrement the control. Adjusting rotary controls D2 Input Controls Input Level Input Level allows you to attenuate signal input level to the D2. The range of this control is from –18 dB to +12 dB. When you use the D2 plug-in in stereo, each channel has its own separate output level knob.
The Cartesian Graph The Cartesian Graph displays a visual representation 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 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. • Black = disabled. In this state the filter is not using any DSP resources and has no effect on audio. • Gray = bypassed.
Low-Mid Peak Filter High-Shelf Filter The Low-Mid Peak Filter provides three rotary controls. The upper rotary control adjusts the center frequency, variable from 33 Hz to 6.4 kHz. The lower left rotary control adjusts the filter’s amplitude gain or attenuation. Amplitude range is ±15 dB from unity (utilizing a reciprocal curve for both gain and attenuation). The lower right rotary control adjusts filter “Q” which is variable from 0.7 to 4.0.
Enabling, Disabling and Bypassing EQ Filters You can enable, disable, or bypass specific EQ filters by clicking them. To disable a filter: Control-click (Macintosh) or Start key-click (Windows) the EQ Filter icon. When disabled, the icon is black. ■ To re-enable a filter: Click the EQ filter icon. When enabled, the icon is white. Editing D2 Parameters Using a Computer Keyboard Each D2 rotary control has a corresponding parameter text field directly below or beside it.
Using D2 in Stereo Link Button Because Focusrite D2 has a single set of Filter parameter knobs, when it is used in stereo, you must select which channel, left or right, you want to edit. Selecting the Link button allows you adjust parameters for both channels simultaneously. By default, Link mode is enabled so that you can easily maintain parity between the parameter settings for both channels. Use the Left Channel, Right Channel, and Link buttons to do this.
chapter 7 Focusrite D3 Focusrite D3 is a high-quality dynamics processor plug-in for Pro Tools. Developed in cooperation with Focusrite, the D3 is based on the highly acclaimed Red Range 3™ dual mono/stereo compressor & limiter designed by Rupert Neve. 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.
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. 2 Click the Limiter icon. Side-Chain Processing Compressor Limiter Compressor and Limiter icons The D3 Compressor The D3 compressor reduces the dynamic range of audio signals that exceed a user-selectable threshold by a specific amount.
D3 Parameters To adjust rotary controls, drag horizontally or vertically. Parameter values increase as you drag upward or to the right and decrease as you drag downward or to the left. Output Level Key On/Off Input Level Input Level attenuates signal input level to the compressor or limiter. The range of this control is from –30 dB to 0 dB. When you use the D3 plug-in in stereo, each channel has its own separate Input Level knob. To adjust input levels for both channels simultaneously, Shift-drag.
A red Clip Indicator appears to the right of the output meter(s). Clicking on the Clip Indicator clears it. Option-clicking (Macintosh) or Altclicking (Windows) clears both channels when the plug-in is used in stereo. • Gray indicates bypassed. In this state the compressor is not active, but is still using available DSP resources. Meters Metering indicators: • Green indicates nominal levels. • Yellow indicates pre-clipping at –6 below full scale. • Red indicates a full scale signal (clipping).
Ratio Release Ratio sets the compression ratio. If the ratio is set to 2:1 for example, it will compress changes in signals above the threshold by one half. The range of this control is from 1.5:1 (very little compression), to 10:1 (heavy compression, bordering on limiting). Release controls how long it takes for the compressor to be fully deactivated after the input signal drops below the threshold level.
D3 Limiter Controls The Limiter icon, which represents a limiter curve, acts as a three-state switch for enabling, disabling, or bypassing the limiter. Its current state is indicated by the icon’s color: To bypass the limiter: ■ Click the icon a second time. When the limiter is bypassed, the icon is grey. • White indicates enabled. In this state, the limiter is active and using available DSP resources.
chapter 8 DINR Digidesign Intelligent Noise Reduction (DINR) consists of a pair of DSP plug-ins (TDM and AudioSuite only) that provide two types of noise removal: Broadband Noise Reduction (BNR) Provides broadband and narrowband noise reduction for suppressing such unwanted elements as tape hiss, air conditioner rumble, and microphone preamp noise. BNR is available as a real-time TDM and as an AudioSuite plug-in. The TDM version of Broadband Noise Reduction is not supported at sample rates above 96 kHz.
How Broadband Noise Reduction Works The Broadband Noise Reduction module uses a proprietary technique called Dynamic Audio Signal Modeling™ to intelligently subtract the noise from the digital audio file. Noise is removed with multiple downward expanders that linearly decrease the gain of a signal as its level falls.
DINR gives you a considerable amount of control over the above three parameters and allows you to maximize noise reduction while minimizing signal loss and artifact generation. However, as powerful as it is, DINR does have limitations. In particular, there are two instances in which DINR may not yield significant results: Cases in which the noise components of the audio are so prominent that they obscure the actual signal components of the audio. ◆ The Noise Signature The jagged line is a graph of noise.
to follow the noise signature as closely as possible will also help maximize noise reduction and minimize signal loss. See “Editing the Contour Line” on page 69. NR Amount, Response, Release, and Smoothing Noise Reduction Amount Controls how much the noise signal is reduced. It is calibrated in decibels. A setting of 0 dB specifies no noise reduction. Increasing negative amounts specify more noise reduction. The default value is 0 dB.
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. High-Shelf EQ The Gain slider controls the gain of the filter. Values range from –12 dB to +6 dB. The HighShelf EQ can be enabled and disabled by clicking the Enable button. You can also use the High-Shelf EQ to reduce the amount of high frequencies in a signal.
After you use the Fit function, BNR will automatically boost the entire Contour Line 6 dB above the noise signature so that all noise components of the audio file are below the Contour Line. You may want to adjust the Contour Line downwards as needed to modify the character of the noise reduction. Super Fit Super Fit The Super Fit button creates a noise Contour Line consisting of over five hundred breakpoints in order to follow the shape of the noise signature more precisely.
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. Control-Option-clicking (Macintosh) or Alt-Start key-clicking (Windows) the Arrow keys on your computer keyboard performs the same function.
6 Adjust the NR amount slider to reduce the noise by the desired amount. To compare the audio with and without noise reduction, click the Bypass button. 7 To fine-tune the effects of the noise reduction, adjust the Response, Release, and Smoothing sliders to achieve optimal results. 8 To further increase noise reduction, edit the Contour Line. The quickest way to do this is to move the entire Contour Line upwards.
4 Click the Fit button to fit a Contour Line to it. Editing the Contour Line 5 In BNR’s Spectral Graph, Command-drag (Macintosh) or Control-drag (Windows) to make a selection. Select points where the highfrequency 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.
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 reduction at that range. Moving a breakpoint lower decreases noise reduction at that range. 3 To move the entire Contour Line, Commanddrag (Macintosh) or Control-drag (Windows) to select the entire range. Click the appropriate Move Breakpoint button (below the Spectral Graph) to move the selected breakpoints in 1 dB increments.
To enable either of these features, click the corresponding button. To disable them, click again. 7 To hear the noise components that are being removed, click Audition. Adjusting 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 removed with the noise, and adjust your parameters accordingly. 8 If unwanted artifacts are generated by the noise reduction process, click Post-processing.
If you have selected multiple regions for processing and want to create a new file that connects and consolidates all of these regions together, choose Create Continuous File from the File mode pop-up menu. ◆ BNR AudioSuite does not allow destructive processing, so the Overwrite Files option is not available in the File mode pop-up menu. 2 From the Destination Track pop-up, choose the destination for the replacement audio. 3 Click Process.
Release 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. Center Frequency Center Frequency Sets the fundamental frequency of the Harmonic filter. Though the Learn function automatically sets the center frequency to the fundamental frequency of the learned noise, this parameter can also be manually edited.
In order for Hum Removal to learn the audio signal from live inputs, Pro Tools must be in playback mode. Signal Present LED Lit when an audio signal is received. Using Hum Removal As 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 frequency spectrum, scanning the noise to find the most prominent frequency characteristic of the audio signal.
chapter 9 Maxim Maxim is a unique and powerful peak-limiting and sound maximizing plug-in provided in TDM, RTAS, and AudioSuite formats. Maxim is ideal for critical mastering applications, as well as standard peak-limiting tasks. Maxim offers several critical advantages over traditional hardware-based limiters.
About Peak Limiting Peak limiting is an important element of audio production. It is the process of preventing signal peaks in audio material from clipping by limiting their dynamic range to an absolute, user-selectable ceiling and not allowing them to exceed this ceiling. Limiters allow you to select a threshold in decibels. If an audio signal peak exceeds this threshold, gain reduction is applied, and the audio is attenuated by a user-selectable amount.
How Maxim Differs From Conventional Limiters Maxim is superior to conventional limiters in several ways. Unlike traditional limiters, Maxim has the ability to anticipate signal peaks and respond instantaneously with a true zero attack time. Maxim does this by buffering audio with a 1024sample delay while looking ahead and analyzing audio material on disk before applying limiting. Maxim can then instantly apply limiting before a peak builds up.
By dragging the Threshold slider downwards, you can visually adjust the level at which limiting will occur. Maxim displays the affected range in orange. dB level of waveform peaks Attenuation This meter displays the amount of gain reduction being applied over the course of playback, with the maximum peak displayed in the numeric readout at the bottom of the meter.
Mix This slider sets the ratio of dry signal to limited signal. In general, if you are applying Maxim to a main output mix, you will probably want to set this control to 100% wet. If you are applying heavy limiting to an individual track or element in a mix to modify its character, this parameter is particularly useful since it allows you add precisely the desired amount of the processed effect to the original signal.
Using Maxim Following are suggestions for using Maxim most effectively. To use Maxim: 1 Insert Maxim on the desired track. 2 Select the portion of the track containing the most prominent audio peaks. In general, a value of 0.5 dB or so is a good maximum ceiling. Don’t set the ceiling to zero, since the digital-to-analog convertors on some DATs and CD players will clip at or slightly below zero.
chapter 10 Bruno and Reso Bruno and Reso are a pair of TDM plug-ins that process audio using a sound generation technique known as cross-synthesis. • Velocity-sensitive gain and detuning Cross-synthesis generates complex sound textures by using an audio track as a tone source then applying a variety of synthesizer-type effects to that tone source.
DSP Requirements Bruno and Reso each require one full DSP chip on a Pro Tools|HD, 24 MIX, or DSP Farm card. DSP and Voice Polyphony The maximum number of Bruno/Reso voices available per DSP chip depends on the sample rate of the session and the type of DSP cards in your system. 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.
Notes played with the on-screen keyboard are triggered at a MIDI velocity of 92. To play Bruno/Reso with the on-screen keyboard: 1 Open the Plug-In window for Bruno/Reso. 2 Click Play on the Pro Tools Transport to start audio playback. 3 Click the on-screen keyboard. Bruno/Reso will only produce sound while audio plays on the source track. 2 Choose File > New Track and specify 1 new MIDI track, then click Create. Create a separate MIDI track for each Bruno/Reso plug-in you use.
Using a Key Input for Sidechain Processing Bruno and Reso feature side-chain processing capabilities. Side-chain processing allows you to trigger certain parameters from a separate reference track or external audio source. The source used for triggering is referred to as the key input. You can use this capability to control the rate at which Bruno performs sample switching or Reso toggles its harmonics back and forth using the dynamics of another signal (the key input). 4 Begin playback.
On-Screen Keyboard The on-screen keyboard The simplest way to play Bruno is to use its onscreen keyboard. You can click one note at a time or use keyboard latch to hold multiple notes. Notes played with the on-screen keyboard are triggered at a MIDI velocity of 92. Timbre Parameters The lower the crossfade frequency, the larger the time slice, and the slower Bruno moves between slices.
no key input is used, the dynamics of the source audio will trigger switching. If a key input is used, the dynamics of the key input signal will trigger switching. Threshold-based switching can be used at the same time as Key Input-based switching. MIDI Clock Triggers switching in sync with a MIDI beat clock signal. This creates a very regular, highly rhythmic wave sequencing effect that is ideal for sessions arranged around MIDI beat clock.
Conversely, if Gain Velocity is set to 0.0 dB, Bruno’s volume will not change no matter how hard or soft you strike a key on your MIDI controller. Gain Velocity only has an effect when you play Bruno with a velocity-sensitive MIDI controller. Mix Mix adjusts the mix of the processed audio with the original, unprocessed audio. Spread When Bruno is used in stereo, the Spread control can be used to pan multiple voices within the stereo field.
Pitch Parameters Detune Amount Detuning is a common sound-thickening technique used on synthesizers and many effects devices. Bruno’s Detune Amount parameter sets the maximum amount of pitch detuning that occurs when multiple voices are stacked together using Voice Stacking. Using a combination of voice stacking and detuning, you can create timbres that are exceptionally fat. Voices can be detuned up to 50.0 cents. (One cent is equal to 1/100th of a semitone.
Voice Parameters Voice Stack Voice Stack selects the number of voices that are used, or stacked when you play a single note. The number of voices that you choose to stack will directly affect polyphony. Selecting a larger number of stacked voices will reduce the number of notes that you can play simultaneously. Voice parameters Voice Stack These parameters control Bruno’s voice polyphony and allocation. The sample rate of your session will also affect polyphony.
If all available voices are being used, playing an additional note will replace the first note played in the chord. A maximum of 8 voices of polyphony are available per Bruno/Reso plug-in on DSP Farm cards. Online Help On-Screen Keyboard On-screen keyboard The simplest way to play Reso is to use its onscreen keyboard. You can click one note at a time or use keyboard latch to hold multiple notes. Notes played with the on-screen keyboard are triggered at a MIDI velocity of 92.
Resonance Velocity Damping Velocity Resonance Velocity increases or decreases resonance according to how hard a MIDI key is struck and how much resonance is initially specified with the Resonance Amount parameter. Damping Velocity increases or decreases damping according to how hard a MIDI key is struck and how much damping is initially specified with the Damping Amount parameter. Resonance Velocity is adjustable from a low of –10 to a high of +10.
All Adds all of the harmonic overtones generated by the resonator. In synthesizer parlance, this produces a somewhat buzzier, sawtooth wave-like timbre. ger toggling. If a key input is used, the dynamics of the key input signal will trigger toggling. Threshold-based switching can be used at the same time as Key Input-based switching. Odd Adds only the odd-numbered harmonic overtones generated by the resonator. In synthesizer parlance, this produces a somewhat more hollow, square wave-like timbre.
Gain Velocity ADSR Envelope Generator Gain Velocity sets the velocity sensitivity of the Gain Amount parameter. This gives you touchsensitive control over Reso’s volume using a MIDI keyboard. The ADSR (attack, decay, sustain, release) Envelope Generator controls Bruno’s amplitude envelope. This amplitude envelope is applied to a sound each time a note is struck. This parameter is adjustable from a low of –24 dB (maximum velocity sensitivity) to a high of 0.0 dB (no velocity sensitivity).
Pitch Parameters Detune Amount Detuning is a common sound-thickening technique used on synthesizers and many effects devices. Reso’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.
LPF/Voice Parameters Follower The Follower is an envelope follower that allows the filter cutoff frequency to dynamically follow the amplitude of the source audio signal. LPF and Voice parameters LPF (Low-Pass Filter) Reso’s Low-Pass Filter is a single resonant filter that is applied to all of Reso’s voices. Frequency The Frequency control sets the cutoff frequency of the Low-Pass Filter in Hertz. All frequencies above the selected cutoff frequency will be attenuated.
notes simultaneously. A voice stack setting of All will allow only one note at a time, but will stack all 24 voices on that note, producing an extremely fat sound. In a 44.
chapter 11 Reverb One Reverb One is a world-class reverb processing TDM plug-in. It provides a level of sonic quality and reverb-shaping control previously found only on the most advanced hardware reverberation units. A set of unique, easy-to-use audio shaping tools allows you to customize reverb character and ambience to create natural-sounding halls, vintage plates, or virtually any type of reverberant space you can imagine.
A Reverb Overview Digital reverberation processing can simulate the complex natural reflections and echoes that occur after a sound has been produced, imparting a sense of space and depth—the signature of an acoustic environment. When you use a reverberation plug-in such as Reverb One, you are artificially creating a sound space with a specific acoustic character.
Reverb One Controls Reverb One has a variety of controls for producing a wide range of reverb effects. Typing Parameter Values You can edit the numerical value of a Reverb One parameter with your computer keyboard. To type a parameter value: 1 Click on the parameter text that you want to edit. 2 Type the desired value. For parameters with values in kilohertz, typing “k” after a number value will multiply the value by 1000. (To enter a value of 8000, for example, type “8k”).
To adjust high frequency cut or damp: ■ Drag the yellow dot right or left. HF Cut/HF Damp Band Cut/Boost Dynamics Controls The Dynamics section has controls for adjusting Reverb One’s response to changes in input signal level. Dynamics can be used to modify a reverb’s decay character, making it sound more natural, or conversely, more unnatural, depending on the desired effect.
Decay Ratio Controls the ratio by which reverb time is increased when a signal is above or below the Threshold level. Dynamics behavior differs when the Decay Ratio is set above or below 1. A ratio setting of greater than 1 increases reverb time when the signal is above the threshold. A ratio setting of less than 1 increases a reverb’s time when the signal is below the threshold.
reverb Size before Time. Setting Time to its maximum value will produce infinite reverberation. The HF Damping and Reverb Color controls also affect reverb Time. Attack Attack determines the contour of the reverberation envelope. At low Attack settings, reverberation builds explosively, and decays quickly. As Attack value is increased, reverberation builds up more slowly and sustains for the length of time determined by the Spread parameter.
Simulating Early Reflections Early reflection presets include: Different physical environments have different early reflection signatures that our ears and brain use to pinpoint location information. These reflections influence our perception of the size of a space and where an audio source sits within it. Changing early reflection characteristics changes the perceived location of the reflecting surfaces surrounding the audio source. • Room: Simulates the center of a small room without many reflections.
Spread Globally adjusts the delay characteristics of the early reflections, moving them closer together or farther apart. Use Spread to vary the size and character of an early reflection preset. Setting the Plate preset to a Spread value of 50%, for example, will change the reverb from a large, smooth plate to a small, tight plate. Frequency Sliders Sets the frequency boundaries between the low, mid, and high band ranges of the EQ.
Set the crossover to 500 Hz to boost low frequencies most effectively. Set it to 1.5 kHz to cut low frequencies most effectively. High frequency cut tings, high frequencies decay more quickly than low frequencies, simulating the effect of air absorption in a hall. The adjustable range is from 120.0 Hz to 24.0 kHz. Band cut/boost Reverb Contour The Reverb Contour graph displays the envelope of the reverb, as determined by the early reflections and reverb tail.
Input Level Meters Input meters indicate the input levels of the dry audio source signal. Output meters indicate the output levels of the processed signal. An internal clipping LED will light if the reverb is overloaded. This can occur even when the input levels are relatively low if there is excessive feedback in the delay portion of the reverb. To clear the Clip LED, click it.
chapter 12 SoundReplacer SoundReplacer is an AudioSuite plug-in designed to replace audio elements such as drums, percussion, and sound effects in Pro Tools tracks with alternate sounds. SoundReplacer can quickly and intelligently match the timing and dynamics of original performance material, making it ideal for both music and audio post production.
The SoundReplacer Solution SoundReplacer Parameters SoundReplacer solves these timing problems by matching the original timing and dynamics of the source audio while providing three separate amplitude zones per audio event. This allows you to trigger different replacement samples according to performance dynamics. Each replacement sample is assigned its own adjustable amplitude zone. Variations in amplitude within the performance determine which sample is triggered at a specific time.
If you change the audio selection on the source track, click Update to update the waveform display. If Auto Update is selected, SoundReplacer automatically updates the waveform display each time you make a new selection or begin playback. If you frequently change selections or start and stop playback, turn off Auto Update to prevent too-frequent redraws. Trigger Threshold The color of the Trigger markers correspond to the matching Threshold slider.
To audition a replacement sample before loading it into SoundReplacer, use the Import Audio command in Pro Tools. Once you have located and previewed the desired audio file, you can then load it into SoundReplacer using the Load/Unload Sound icons. SoundReplacer does not load 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.
When Auto Update is selected, SoundReplacer automatically updates the waveform display each time you make a new selection on a source track. If you frequently change selections or start and stop playback, you may want to deselect Auto Update to prevent frequent redraws. The Dynamics button provides a quick means of toggling on and off the Dynamics control. When Dynamics is toggled off, SoundReplacer will not track changes in the source audio file’s dynamics.
6 Repeat steps 3–5 to load replacement sounds into amplitude zones 2 and 3. If you use only a single replacement sample, you should still set all three amplitude zones for optimum results. This will ensure accurate triggering. For details, See “Mapping The Same Sample Into Multiple Amplitude Zones” on page 113. 7 To align the amplitude peak in the replace- ment file(s) to threshold trigger markers in the source audio, enable Peak Align. 8 Click Preview to audition the replacement au- dio.
In general: Turn on Peak Align if you are replacing drum or percussion sounds whose peak level occurs at the initial attack. ◆ Turn off Peak Align if you are replacing sounds whose peak level occurs somewhere after the initial attack. Peak Align should also be turned off if the sounds you are replacing are not drum or percussion sounds. ◆ To illustrate why Peak Align makes a difference, look at the following illustrations: Figure 1.
If you use a single amplitude threshold to trigger the replacement sample, you have to set the threshold low enough to trigger at the soft hits. The problem occurs at the loud hits: The threshold is now set so low that the pre-hit portion of the loud hits can exceed the threshold—triggering the replacement sample too early. This results in a replacement track with faulty timing.
By always putting replacement audio files in this special folder, you can freely exchange SoundReplacer settings—and the audio files associated with them—with other users. Do not create subfolders within SoundReplacer’s Audio Files folder. Files located within subfolders are not recognized. SoundReplacer Demo Session SoundReplacer includes a Demo session that illustrates useful applications of this plug-in. Before you start: 1 Open the Demo session. 2 Choose SoundReplacer from the AudioSuite menu.
In the Demo session: 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. 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.
In the Demo session: 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.
118 Digidesign Plug-Ins Guide
chapter 13 SurroundScope SurroundScope is a TDM and RTAS plug-in that assists with multichannel mixing by visually representing the phase, individual input levels, and position of multichannel signals in the surround field. SurroundScope supports all industry-standard, multichannel formats, including LCR, Quad, LCRS, and 5.0, 5.1, 6.0, 6.1, 7.0, and 7.1. SurroundScope features include: • 360° Surround Display indicates signal position within the surround field.
SurroundScope Parameters Each of SurroundScope’s interactive meters displays a different type of information about a multichannel audio signal. Following is an explanation of these meters and their behavior. Surround Display The Surround Display shows signal position within the surround field. When SurroundScope is inserted on a track, it detects the multichannel format of the track and displays each speaker channel in the signal in a circle around the Surround Display’s center.
To understand the precise location of an audio signal within the sound field, observe how the circular sound pattern moves: A perfect circle in the center of the Surround Display denotes a perfectly centered surround signal—one that is panned equally to all channels. ◆ A signal that balloons toward the outer edges of the display indicates which channels emanate the strongest signal.
To choose the channels you want to compare on the Phase meter: 1 With SurroundScope inserted on a track, click in the Surround display on the letter corresponding to the desired speaker channel. 2 Click the letter corresponding to the second desired speaker channel. Input Level Meters The Input Level meters provide decibel level readings for each channel of a multichannel signal, including readings for an LFE channel if present.
appendix a DSP Requirements for TDM Plug-Ins The tables on the following pages show the total number of instances of each Digidesign TDM plug-in that can be powered by a single DSP chip on Pro Tools|HD-series and MIX-series cards. DSP usage differs according to card type. There are a total of 9 DSP chips on a Pro Tools|HD card. 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 to function.
Table 1. Maximum instances of real-time TDM plug-ins per DSP chip for a Pro Tools|HD card at different sample rates (mono and stereo). Sample Rate: Plug-In 44.1/48 kHz 88.2/96 kHz 174.
Table 2. Maximum instances of real-time TDM plug-ins per DSP chip for a Pro Tools|HD card at 48 kHz* LCR Quad & LCRS 5.0 5.1 & 6.0 6.1 & 7.0 7.
≈ ≈ Table 3. Maximum instances of real-time TDM plug-ins per DSP chip for a Pro Tools|24 MIX card Table 4.
appendix b DSP Delays Incurred by TDM Plug-Ins Table 5 on page 128 shows the delays inherent in each type of Digidesign TDM plug-in. Delays matter only if you use a real-time TDM plug-in on one channel of a stereo or multichannel signal but not the others. This can cause the channels to be out of phase. You can use the DigiRack TimeAdjuster plug-in to compensate for these delays. See the DigiRack Plug-Ins Guide for more information. See the Pro Tools Reference Guide for details on DSP-induced delays.
≈ Table 5.
index Numerics 100% Wet parameter 100 1-2 Band EQ 49 4-Band EQ 49 6-Band EQ 49 A acoustic environments 43, 98 adaptive quantization 33 ADSR Envelope Generator 87, 93 Algorithm parameter 45 aliasing artifacts 33 All (harmonics) parameter 92 Alternating Rectification 37 Alt-Max Rectification 37 Amplitude envelope 87 parameters 86, 92 Analyze button 27 anechoic chamber 43, 98 Anti-Alias Filter parameter 33 Attack parameter 59, 87, 102 Attenuation parameter 78 AudioSuite AudioSuite Buffer 28 buffer size 28 me
C D3 Cartesian Graph 51 Ceiling parameter 78 Chan/Track Process Mode 25 Channel selector 13 Church algorithm 45 Clip Indicator D-Verb 44 clipping indicator (Reverb One) 106 Clock parameter 86 closing multiple plug-in windows 15 Compare button 13 Compressor controls 58 in/out icon 58 Contour Line 62, 63 editing 69 Convert Plug-In pop-up 13 Copy Settings command 19 copying a plug-in setting 20 CPU meter 8 CPU power requirements for RTAS plug-ins 8 CPU Usage Limit parameter 8 Create Continuous File command 2
E H early reflections 44, 98 Early Reflect On 104 Early Reflections parameter 102 ER (early reflection) button 105 ER Settings parameter 103 presets 103 simulating 103 Edit Settings parameter 28 Effect Amount parameter 34 Effect Frequency parameter 34 effects recording with 10 Entire Selection mode 25 Envelope Follower 35, 95 Envelope Generator 87 EQ bypassing 51 curves 51 disabling 51 enabling 51 filter controls 51 External Key parameter 85, 92 Hall algorithm 45 Hardware Buffer Size 8 harmonic overtones
LFE Enable button 14 LFO (Low Frequency Oscillator) 35 Librarian menu 13 license card 4 Limit LED of D3 60 Limiter D3 controls 60 D3 Limiter in/out icon 60 Limiting a mixdown 76 drums 76 individual instruments 76 Linear Quantization parameter 33 Link button 54, 79, 80 Link Enable buttons 13 linking plug-in controls 15 Lissajous Meter 119 Load/Unload Sound icons 109 Lock Settings File command 19 locked regions and AudioSuite processing 29 Lo-Fi 31, 32 demo session 39 Low-Pass Filter 52 parameters 46 Low-Pass
Pitch Parameters of Bruno 88 Pitch Parameters of Reso 94 pitched noise removal 61 Playlist command 24 plug-in parameters editing 15 linking 15 Plug-In Selector 13, 24 plug-in settings librarian 18 Plug-In window closing multiple windows 15 opening multiple windows 14 plug-in-induced delays 9 plug-ins as Inserts 10 automating 17 creating a default setting 20 inserting 11 inserting during playback 11 inserting on-the-fly 11 mono 11 moving and duplicating 12 multichannel 11 multi-mono 11 Plug-In Selector 23 re
Reverb One adjusting parameters 99 Chorus controls 101 DSP requirements 97 HF Cut parameter 104 HF Damp parameter 105 Online Help 105 reverberation explained 98 Right Channel button 54 Ring Mod parameter 34 ring modulation 34 RMS detector 56 Root Settings folder 19 specifying 20 RTAS plug-ins 2 improving performance 8 S Sample Rate parameter 32 Sample Size parameter 33 Sample+Hold parameter 35 saving plug-in settings 20 Save Plug-In Settings To command 19 Save Settings As command 19 Save Settings command 1
Time parameter 101 Time Trimmer 29 TimeAdjuster plug-in using for delay compensation 9 time-slicing 81 Toggle (harmonics) parameter 92 Track Selector 13 triangle wave 35 Trigger and Hold 35 Trigger markers 108 Trigger Threshold 109 typing parameter values 99 U Undo button for DINR 67 unlinking plug-in controls 16 Update button 109, 110 Use in Playlist button 24 User Default setting, creating 20 V Vari-Fi 32, 37 Voice Mode parameter 89 Voice parameters 89, 95 voice polyphony of Bruno/Reso 82 voice stacking
