User manual APPLE LOGIC PRO 7 - PLUG-IN REFERENCE MANUAL

Logic Pro 7 Plug-In Reference Apple Computer, Inc. © 2004 Apple Computer, Inc. All rights reserved. Under the copyright laws, this manual may not be copied, in whole or in part, without the written consent of Apple. Your rights to the software are governed by the accompanying software licence agreement. The Apple logo is a trademark of Apple Computer, Inc., registered in the U.S. and other countries. Use of the "keyboard" Apple logo (Option-Shift-K) for commercial purposes without the prior written consent of Apple may constitute trademark infringement and unfair competition in violation of federal and state laws. Every effort has been made to ensure that the information in this manual is accurate. Apple Computer, Inc. is not responsible for printing or clerical errors. 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Apple assumes no responsibility with regard to the performance or use of these products. 1 Contents Preface 9 10 13 13 16 19 20 21 23 23 24 29 29 32 33 37 38 38 38 39 39 42 42 43 45 45 47 49 50 52 Introducing Logic's Plug-ins About This Manual Basics Using Plug-ins The Plug-in Window Plug-in Settings Plug-in Automation Plug-ins From Other Manufacturers Instruments and Effects Effect Plug-ins Instrument Plug-ins Equalizer Channel EQ Linear Phase EQ Match EQ Fat EQ Silver EQ DJ EQ Individual EQs Dynamic Compressor Silver Compressor Expander Noise Gate Silver Gate Enveloper DeEsser Limiter Adaptive Limiter Multipressor Chapter 1 Chapter 2 Chapter 3 Chapter 4 3 Chapter 5 57 57 62 62 63 64 65 66 67 67 70 72 79 92 92 93 93 94 96 97 97 98 99 99 101 104 105 106 106 107 109 109 110 111 114 115 117 118 119 Distortion Guitar Amp Pro Distortion Overdrive Bitcrusher Clip Distortion Phase Distortion Distortion II Filter AutoFilter Fuzz-Wah EVOC 20 Filterbank EVOC 20 TO High Cut/Low Cut High Pass/Low Pass Filter Delay Sample Delay Tape Delay Stereo Delay Modulation Modulation Delay Chorus Flanger Phaser RingShifter--Ring Modulator/Frequency Shifter Tremolo Ensemble Rotor Cabinet Scanner Vibrato Spreader Reverb AVerb SilverVerb GoldVerb PlatinumVerb EnVerb Convolution Reverb: Space Designer Using Space Designer Space Designer's Parameters Chapter 6 Chapter 7 Chapter 8 Chapter 9 Chapter 10 4 Contents 137 138 Chapter 11 143 143 144 145 147 150 152 153 154 157 157 158 159 160 161 163 166 166 167 167 168 168 169 169 175 175 176 190 191 195 195 196 196 201 201 Creating Impulse Responses About Convolution Special Spectral Gate Pitch Shifter II Vocal Transformer Pitch Correction SubBass Denoiser Exciter Stereo Spread Helper Test Oscillator Tuner Gain I/O Direction Mixer Multimeter Correlation Meter Levelmeter Vocoder--Basics What Is a Vocoder? How Does a Vocoder Work? How Does a Filter Bank Work? Analyzing Speech Signals Tips for Better Speech Intelligibility The EVOC 20 PS Using the EVOC 20 PS EVOC 20 PS Parameters Block Diagram Vocoder History Synthesizer Basics Analog and Subtractive What Is Synthesis? Subtractive Synthesis EFM 1 Concept and Function Chapter 12 Chapter 13 Chapter 14 Chapter 15 Chapter 16 Chapter 17 Contents 5 202 202 204 205 Chapter 18 207 207 209 209 213 213 215 215 223 223 225 283 301 301 302 309 325 332 340 341 355 356 359 360 362 373 379 384 392 400 408 409 426 Global Parameters FM Parameters Modulator and Carrier The Output Section ES M Parameters of the ES M ES P Parameters of the ES P ES E Parameters of the ES E ES1 Parameters of the ES1 ES2 Concept and Function The ES2 Parameters Tutorials Ultrabeat The Structure of Ultrabeat Overview of Ultrabeat The Synthesizer Parameters Modulation The Step Sequencer Importing Sounds Tutorial: Creating Drum Sounds in Ultrabeat Sculpture The Synthesis Core of Sculpture Sculpture's Parameters Global Parameters String and Object Parameters Processing Post Processing Modulation Generators The Control Envelopes Morph MIDI Controller Assignments Programming: Quick Start Guide Programming: In Depth Chapter 19 Chapter 20 Chapter 21 Chapter 22 Chapter 23 Chapter 24 6 Contents Chapter 25 Chapter 26 453 455 455 456 460 477 481 482 485 485 486 502 503 505 505 506 513 516 519 520 526 531 568 570 571 573 573 575 577 599 KlopfGeist EVB3 Concepts and Function MIDI Setup The EVB3 Parameters MIDI Controller Assignments Additive Synthesis With Drawbars A Short Hammond Organ Story EVD6 The EVD6--Concept and Functions Parameters of the EVD6 Controlling the EVD6 via MIDI A Brief History of the Clavinet EVP88 The EVP88--Concept and Functions Parameters of the EVP88 The E-Piano Models Emulated EVP88 and MIDI EXS24 mkII Using Instruments File Organization Sample File Import EXS24 Key Commands A Brief History of Sampling MIDI Controller List GarageBand Instruments About GarageBand Instruments External Instrument Chapter 27 Chapter 28 Chapter 29 Chapter 30 Chapter 31 Glossary Index Contents 7 The professional Logic music and audio production software features a comprehensive collection of powerful plug-ins. These include; innovative synthesizers, high quality effect plug-ins and authentic recreations of vintage instruments. Logic also supports the use of Audio Unit plug-ins in Mac OS X and also supports TDM plug-ins for users of TDM systems. Given a fast enough computer, you could conceivably arrange and mix an entire song using several software instruments, such as Logic's ES1, ES2, EVP88, or EXS24, amongst others. These instruments have the added benefits of superior sound quality and timing as the audio signal never leaves the digital domain, and you can freely edit these software instrument parts, change the tempo and more, right up to the final mix. Don't worry if you're unfamiliar with the terminology used here--this manual will explain everything. It covers all of the general things you need to know about plug-ins and will introduce you to the individual effects and instruments and their parameters. We've included a few tutorial chapters, which will explain how to program sounds using several of Logic's instrument plug-ins. Using plug-ins is much easier if you are familiar with some of Logic's basic functions. You should be acquainted with Logic's Audio Mixer before going further. Information about it can be found in the Audio Mixer section of the Logic reference. The Bounce buttons found on the Master Audio Objects allow you to write submixes of plug-in tracks--as an audio file--to disk at any time. For details please refer to the Logic reference. Whatever you play on your instruments can be recorded by simply pressing Logic's Record button. Your performances can be freely edited in any of Logic's MIDI editors. Further details about this can be found in the Logic reference Preface 9 Introducing Logic's Plug-ins Logic's plug-ins include the following features: · Real-time processing of audio. · Support for sample rates up to 192 kHz. · Altivec optimizations for the Power Macintosh G4 and G5 processors which increase the number of software effects and instruments that can be run simultaneously. · A sophisticated, intuitive, real-time graphical editing interface for most Logic plug- ins. · A consistent window interface for Logic, Audio Unit and TDM plug-ins. · The ability to save and load individual plug-in effect and instrument settings or entire channel strip configurations, including those from Apple's GarageBand application. · Almost all plug-in parameters can be automated via Logic's total recall mix automation system. About This Manual This guide covers all areas of plug-in usage in Logic. All plug-in parameters are discussed in detail. The Basics section discusses the most essential aspects of plug-in usage, the Plug-in window interface and global plug-in commands and menus. The Instruments and Effects chapter covers the differences between effect and instrument plug-ins. Ensuing chapters discuss the parameters of individual plug-in effects and instruments. The instrument chapters include a number of tutorials that will help you to make the most of your new instrument. The Onscreen Help system--accessible from Logic's Help menu--is fundamentally the Reference Manuals in electronic form. It has the advantage of being at your fingertips when you need it, and is also searchable. Even if you're the type who just doesn't like reading manuals, we ask that you read the next section. It will provide you with essential information on the basic use of Logic's plug-ins. Please note that all topics described herein were accurate at the date of printing. For up to date information on changes or additions made after printing, please refer to the Late Breaking News on the Logic DVD, and/or to the Update Info, included with each Logic update. 10 Preface Introducing Logic's Plug-ins Conventions of this Guide... Before moving on to the Basics section, we'd like to cover the following conventions used in this manual. Menu Functions For functions that can be reached via hierarchical menus, the different menu levels are described as follows: Menu > Menu entry > Function. Important Entries Some text will be shown as follows: Important: Information on function or parameter. These entries discuss a key concept or technical information that should, or must, be followed or taken into account. Please pay special attention to these entries. Notes Some sections provide additional information or tips that will assist your use of the effect or instrument plug-in. These are displayed as shown below: Note: Information on function or parameter. Key Commands Several plug-in functions can be activated or accessed with key commands--computer keyboard shortcuts. The key commands mentioned in this guide are based on the standard Key Command Set, assigned by the Logic Setup Assistant. Where possible, we have also included the standard Key Commands for PowerBook users. These are based on the PowerBook Key Command Set, assigned in the Logic Setup Assistant. Preface Introducing Logic's Plug-ins 11 1 Basics 1 This chapter covers all important steps required for plugin use in Logic. The steps include: · Inserting, deleting, and bypassing plug-ins. · Operating plug-ins in the Plug-in window. · Managing plug-in settings. · Automating plug-ins. Using Plug-ins Inserting and Deleting Plug-ins Plug-ins can be either; software instruments, which respond to MIDI note messages, or audio effects, which do not respond to MIDI note messages. · All plug-ins can be added via the plug-in menu of an Audio Object. · Effect plug-ins can be inserted into the Insert slots of all Audio Objects. · Software-based instruments can only be inserted into special Audio Objects, called Audio Instruments. These Audio Instrument Objects have a special Instrument slot, directly above their Output slots. 13 To add a plug-in: 1 Click-hold on an Audio Object's Insert/Instrument slot. 2 The plug-in-menu appears, showing all available plug-ins. Move the mouse through the different levels of the hierarchical menu and choose a plug-in name, then release the mouse button. The Plug-in window is launched automatically. If you do not want the Plug-in window to open automatically after insertion, uncheck the Preferences > Audio > Display > Open Plug-in window on insertion preference. You can open a closed Plug-in window by double-clicking on an assigned Insert/ Instrument slot. You can set all plug-in parameters in the Plug-in window. For further information please read "The Plug-in Window" on page 16. Closing the Plug-in window leaves the plug-in active. 14 Chapter 1 Basics To remove a plug-in: 1 Click-hold the corresponding Insert/Instrument slot. 2 The plug-in menu is opened. Select the No Plug-In menu option. Inserting Mono/Stereo Plug-ins You can insert mono and stereo effects into Logic's mono objects. If you use a stereo effect in a mono object, the plug-in menu is limited to stereo effects from this insert point onwards. Note: In general, stereo effects require twice as much processing power as their mono counterparts. In stereo objects, the plug-in menu only shows effects with stereo inputs and stereo outputs. If you hold the Option key while opening the plug-in menu on stereo objects, you can also select mono effects. Logic automatically inserts conversion modules (in the background) to handle stereo mono and mono stereo transitions. This enables you to use plug-ins in any order. Please keep the following in mind when doing so: · These conversion modules require extra processing power. · During a stereo mono conversion, all spatial information is lost. · During a mono stereo conversion, no spatial information is added--the same mono signal is sent to both outputs. Bypassing Plug-ins If you want to deactivate a plug-in, but don't want to delete it, you can bypass it. Bypassed plug-ins do not drain system resources. To bypass a plug-in: Option-click the appropriate plug-in insert/instrument slot on the desired Audio Object. The insert slot of the bypassed plug-in turns from blue to gray, indicating that the plugin is currently bypassed. You can also use bypass a plug-in from within the Plug-in window. Further information on this can be found in the following section. m Chapter 1 Basics 15 The Plug-in Window Hands-on operation of plug-ins is performed in the Plug-in window. This window allows access to all plug-in parameters. The Plug-in window can be opened by doubleclicking on the blue plug-in label on an Audio Object. Each instance of a plug-in has its own Plug-in window, allowing each to have discrete settings. Operation of Built-in Plug-ins Adjusting Parameters To toggle a Plug-in window's buttons: Click on the button. It toggles to the next/previous option, or will be enabled/disabled. To adjust a slider: Click-hold anywhere on the slider and drag up/down or left/right. To adjust rotary knobs: Click-hold on the center of the rotary knob and drag the mouse up and down. You can also move the mouse in a circular motion. Fine-tuning of values is easier when using a larger radius for this circular motion. To adjust numerical panels: Click-hold on the panel's numerical value and drag up/down. If there are up/down arrows alongside such panels, you can use them to increment/decrement the value by one step. Note: You can reset any parameter to its default value by Option-clicking on it. Note: If you hold Shift before clicking and moving a control, its value can be finetuned. m m m m Common Plug-in Window Parameters The gray area at the top of the Plug-in window is common to all Logic plug-ins. It offers a number of important functions for plug-in use. Link The button to the extreme left (with a chain on it) is called the Link button. If the Link button is switched on, a single Plug-in window will be used to display all opened plugins. Each time you launch a new plug-in, the window will update to reflect the new selection. By default, the Link button is switched off, allowing you to open several Plugin windows simultaneously. This is handy if you want to compare the settings of two plug-ins or adjust several plug-ins at the same time. 16 Chapter 1 Basics When changing the Arrange track, an open Plug-in window will update to display the corresponding slot's plug-in on the newly-selected track. As an example, if the ES1 was loaded on Audio Instrument channel 1, and an EXS24 instance was loaded on Audio Instrument channel 1, switching between these tracks would automatically update the Plug-in window to show the ES1/EXS24, respectively. Bypass The Bypass button allows a plug-in to be deactivated, but not removed from the insert/ instrument slot. You can also bypass the effect directly in the Audio Object by Optionclicking on the corresponding insert slot. Settings Menu (Arrow) Clicking the Arrow to the right of the Bypass button accesses the Settings menu. Further information on this can be found in "Plug-in Settings" on page 19. Switching the Contents of the Plug-in Window You can reassign any open Plug-in window--in two different ways--via the two pulldown menus to the right of the Settings menu (the Arrow): · Use the upper pull-down menu (Track 1 in the diagram) to switch the Plug-in window between all channels that use the same plug-in. If you have inserted the EVB3 on tracks 1 and 6, for example, you can switch between these channels and adjust the parameters of each EVB3 instance, respectively. · In the lower pull-down menu you can switch between the plug-in slots of the selected channel. As an example, if a particular channel uses an Equalizer and an EVB3 plug-in, you can switch the Plug-in window between these plug-ins. Editor--Controls View The plug-in parameters can be viewed in two forms: Controls and Editor. The Editor view shows the plug-in's graphical interface, if it offers one. The Controls view displays all plug-in functions as a set of horizontal sliders, with numerical fields to the left of each parameter. These fields are used for both the display and entry of data values. To switch the view modes: 1 Click-hold the Editor button in the gray area at the top of the Plug-in window. 2 Choose Controls from the pull-down menu. Chapter 1 Basics 17 Some Logic plug-ins may have additional parameters that don't show up on the Editor control panel. This is indicated by an additional 001/011 button next to the Link button. Activate this button to reveal sliders for the extra parameters at the bottom of the Plugin window. Plug-ins With Side Chain Input All plug-ins that support side chain inputs, feature an additional Side Chain pull-down menu in the gray area at the top of the Plug-in window. This facilitates the routing of any Audio track, Input channel or Bus Object into the plug-in via a side chain. You can also route an Instrument channel as side chain signal, if you follow these steps: 1 Create a Send, using a Bus on the Instrument channel. 2 Choose the selected Bus as a Side Chain input for the plug-in. Once the Side Chain input is selected, the plug-in processes the audio of the channel it is inserted in, using the trigger impulses provided by the Side Chain. The signal peaks of the Side Chain input, combined with the Threshold parameter of the plug-in, determine when the plug-in is triggered. Examples for Side Chaining · A sustained pad sound is sent through a noise gate, which is triggered by a drum track being used as the Side Chain input signal. This results in a rhythmic pad sound which follows the signal peaks of the drum track. · A noise gate inserted into a bass guitar channel is triggered by the kick drum track via the Side Chain. This can "tighten" the timing of the bass guitar, as it follows the kick drum signal. · Side Chains can also be used to blend a music mix with a voice-over. To achieve this, the mix needs to be routed through a compressor which, in turn, is side chained, using the voice-over track. In this type of setup, the music becomes softer when the narrator is speaking, and louder, when not. The effect is also known as ducking. Please note that in order for this to function, the automatic gain make-up or Auto Gain (if applicable to the compressor plug-in) must be disabled. 18 Chapter 1 Basics Plug-in Settings Logic's plug-ins ship with a library of ready-to-play preset sounds, known as Settings. These Settings can be found in the Logic > Plug-In Settings subfolder, following the installation procedure. Note: It is strongly recommended that you do not attempt to change the Logic > Plugin Settings folder structure. Within the Plug-in Settings folder you are, however, free to sort your settings into sub folders. This folder structure is reflected in a hierarchical menu, shown each time you load a plug-in setting. All current plug-in settings are stored with the song file, and are automatically recalled the next time you load the song. You can also recall and save individual settings via the Settings menu functions. The Settings pull-down menu can be opened by clicking on the Arrow in the gray area at the top of the Plug-in window. Functions of the Settings Menu In the gray area at the top of each Plug-in window is an Arrow button. Clicking on it opens the Settings menu, which features the following functions: Copy Setting Choose this entry to copy all parameter settings into a special Settings clipboard, which is independent from the global Logic clipboard. Paste Setting If you have opened a plug-in of the same type (two SilverVerb instances, for example), you can use this command to paste the parameter set from one to the other via the Settings clipboard. Save Setting This allows you to name and save a setting. Note: If you save a Setting with the name of #default in a plug-in's Settings folder, it will be loaded as the default plug-in Setting. Chapter 1 Basics 19 Load Setting This function can be used to load a setting. The file selector box only shows settings for compatible plug-in types. Each plug-in has its own set of parameters, and therefore its own file format. Note: Proprietary plug-in-settings created in Logic for Windows can be read by Logic for Mac OS, and vice versa. Plug-in settings files created on the Mac must be saved with a .pst file extension in order for them to work in Logic for Windows. Note: Some plug-ins allow you to load Settings files by dragging and dropping them from the Finder. This poses a problem as float windows will disappear once Logic is "in the background" and the Finder becomes the active application. To circumvent this , issue, you can hold Option when inserting a plug-in, making it a non-floating window. Next/Previous Setting These functions allow you to load the next/previous setting in the folder. You can also make use of the Next/Previous Plug-In Setting (or the Next/Previous Plug-In Setting or EXS Instrument) key commands. These are not set by default, so you will need to assign them. Once assigned, you can simply press the appropriate key command to step forwards/backwards through your plug-in settings. In Logic Pro, Previous/Next Setting can be assigned to almost any MIDI message, such as Control Change or Program Change commands. Settings of other Manufacturers Logic can read the most common settings files used by Audio Unit plug-ins. Loading and Saving Multiple Plug-ins Logic's Mixer windows allow you to save and load multiple plug-ins (inclusive of their Settings files) via the arrow pull-down menu alongside the word Inserts on channel strips. The entire channel strip can be stored and recalled for use on any suitable Audio Object, allowing common chains of effects such as Reverb, Chorus, and Delay to be loaded far more quickly than individually inserting each plug-in. Further details can be found in the Logic reference. Plug-in Automation Almost all Logic plug-ins can be fully automated, which means that you can record, edit, and play back almost any movement of any knob, switch or fader in any plug-in. For more information, please read the Automation chapter in the Logic reference. 20 Chapter 1 Basics Plug-ins From Other Manufacturers Audio Unit Support Correctly installed third-party Audio Unit plug-ins (Effects and Instruments) can be used in Logic. Clicking on an Audio object insert/instrument slot will launch the hierarchical Plug-In menu. A separate Audio Units submenu displays all installed Audio Unit plug-ins. TDM Plug-ins Users of a Digidesign TDM system can utilize TDM plug-ins in Logic. Chapter 1 Basics 21 2 Instruments and Effects 2 This chapter explains the difference between effect and instrument plug-ins. Instrument plug-ins respond to MIDI note messages, effect plug-ins do not. Therefore instrument plug-ins can only be inserted into special Audio Objects, called Audio Instruments. Effect Plug-ins Logic's effects can be installed into all insert slots of all Audio Object types (See "Inserting and Deleting Plug-ins" on page 13.). This allows processing of all audio and instrument signals. There are two ways of sending audio to effects: via an insert, or via a bus (also known as an "aux send"). Insert Effects With insert effects, all of the signal is processed. This means that 100% of the signal flows through the effect. This is suitable for equalizers or dynamic effects. This also typically applies to pan knobs and faders. If you have enough processing capacity, you can use up to 15 insert effects per audio object. An extra blank insert is created, as soon as all the currently displayed inserts are used, up to the maximum allowed. Bus Effects When you use bus effects, a controlled amount of the signal is sent to the effect. Buses are typically used for effects that you want to apply to several signals at the same time. 23 Within Logic, the effect is placed in an insert slot of a bus object. The signals of the individual tracks can each be sent to the bus, controlled by a Send knob. The audio signal is then processed with the effect, and mixed with the stereo output. The advantage of this "bussed" approach, over inserting effects on tracks, is efficiency. This method allows as many tracks as you like to be processed by one inserted plug-in, massively saving CPU power when compared to insertion of the same effect directly into multiple tracks. For computationally-intensive effects such as reverb, it's always advisable to insert them into a bus. Chorus, Flanger, and Delay effects should also always be inserted into a bus, if they are going to be used on more than one track. In some cases, it may make sense to patch an effect such as a delay, directly into the insert of an individual track. There are no restrictions in Logic as to where effects may be used. Instrument Plug-ins The Audio Instrument Object Type Unlike effect plug-ins, instrument plug-ins respond to MIDI note messages. Instrument plug-ins can only be inserted into special Audio Objects, called Audio Instruments. Audio Instruments feature a special instrument slot, directly above their Output slot. An Audio Instrument is an Audio Object with its Channel parameter switched to one of the Instruments. Any audio object can be switched to operate as an Audio Instrument, by changing this parameter (Channel) in the Object Parameter box. To create an Audio Instrument Object: 1 Open Logic's Audio Mixer, by choosing Audio > Audio Mixer. 2 In the Audio mixer window select New > Audio Object to create a new Audio Object. 24 Chapter 2 Instruments and Effects 3 Double click the newly-created Audio Object icon, so that the (grayed out) channel strip appears. 4 Now, go to the Object Parameter box, and set the Channel parameter to an Instrument. The generic Audio Object will now operate as an Audio Instrument, allowing you to insert any Instrument plug-in into the instrument slot. The default song--the song that opens if you move the Autoload Song away from the Logic folder--features a number of ready-configured Instruments, that can be accessed via the Track Mixer or Audio Mixer. The output signal of a software instrument plug-in is fed into the input (the instrument slot) of the Instrument channel strip, where it can be processed via inserted plug-ins and/or sent to busses. Logic supports up to 64 discrete Audio Instruments. The number of instrument instances which can be run simultaneously is dependent on the availability of computer processing resources. Following the insertion of an instrument, the Audio Instrument Object can be used just like a MIDI track in the Arrange window. The Audio Instrument Object can also receive MIDI notes from standard MIDI instrument objects via Environment cables. This is useful for creating layered sounds with "real" MIDI instruments and virtual instruments. Please note that the Options > Preferences > MIDI > Use Unified Virtual and Classic MIDI Engine setting needs to be switched on for these features to work. When an Audio Instrument track is selected, it is ready to be played in real-time and consequently produces some system load. Normally, Logic releases system resources used by the Audio Engine when the sequencer is stopped. This is not the case, however, if an Audio Instrument track is selected in the Arrange window, and is therefore available for real-time playing. Selecting a MIDI track or a standard Audio track exits this Audio Instrument "stand by" mode, and releases reserved system resources when the sequencer is stopped. Note: Muting an Audio Instrument track in the Arrange does not reduce system load. Chapter 2 Instruments and Effects 25 Logic's Bounce function allows the entire Audio Instrument track to be recorded as an audio file. This "Bounced" audio file can then be assigned (as an audio region) to a standard Audio track, allowing you to reassign the available processing (CPU) power for further synthesizer tracks. For details, please refer to the Bounce chapter in the Logic Reference manual. You can also make use of the Freeze function to capture the output of an Audio Instrument track, again saving processing power. For details please refer to the Freeze section, in the Logic Reference manual. Accessing Multiple Outputs Logic supports the multiple outputs of the EXS24 and all Audio Unit (AU) compatible instruments. In addition to the Mono and Stereo submenus of the Audio Instrument plug-in menu, a Multi Channel submenu lists all Instruments that offer multiple outputs. A plug-in needs to be inserted from the Multi Channel submenu, in order to access its individual outputs. Note: Not all plug-ins (both Logic and third-party) are multi-output capable. If the Instrument does not appear in the Multi Channel submenu, it is not equipped with multiple output facilities. The first two outputs of a multiple output instrument are always played back as a stereo pair by the Instrument channel in which the plug-in is inserted. Additional outputs (3 and 4, 5, and 6, and so on) are accessed via the Aux Objects. 26 Chapter 2 Instruments and Effects Software Instrument Pitch The Song Settings > Tuning > Software Instrument Pitch > Tune parameter remotely controls the main tuning parameter for all software instruments (plug-in synthesizers, such as the ES1 or EXS24 sampler and others) by ±100 cents. Note: Some instruments do not recognize this remote command. No Hermode Tuning Logic allows all software instruments to be globally tuned to different tempered scales, including Hermode Tuning (see the Tuning section of the Logic reference manual for details). There may, however, be occasions where you want individual Software Instruments to be exempt from this global tuning system. When File > Song Settings > Tuning > Hermode Tuning is active, a No HMT checkbox is visible in the Object Parameter boxes of all Audio Instrument channels. Simply click in this box to prevent the selected software instrument from following the global Hermode Tuning scale. Any software instrument with an active No HMT checkbox will be played back at equal temperament. Chapter 2 Instruments and Effects 27 3 Equalizer 3 This chapter covers all Logic equalization effects. Equalizers allow you to increase or decrease the level of selected components in the overall audio spectrum. Logic's built-in equalizers include the Channel EQ, Linear Phase EQ, Match EQ, Fat EQ, Silver EQ, DJ EQ, High/Low Pass Filters, High/Low Cut EQ, Parametric EQ and High/Low Shelving EQ plug-ins. Channel EQ The extremely high-quality Channel EQ offers eight frequency bands and an integrated FFT analyzer. EQ Parameters The Band Type buttons above the display activate the Channel EQ's bands individually; inactive bands do not use any computer resources. Band 1 is a lowpass filter and band 8 is a highpass filter. Note: The Q-parameter of band 1 and band 8will have no effect when using a slope of 6 dB/Oct. 29 Bands 2 and 7 are defined as shelving equalizers. Note: When the Q parameter of band 2 and 7 is set to an extremely high value (to 100, for example), the equalizers only apply to a very narrow band, and can work in a fashion that is similar to notch filters. Bands 3to 6 are bandpass filters. You can set the band parameters either in the parameter area or directly in the central EQ display. Move the mouse horizontally over the display. When your mouse cursor is in the access area of a band, its individual curve and parameter area will be highlighted and a pivot point appears. When you click-hold the mouse button directly on the (illuminated) pivot point of a band, vertical movements (up/down) will change its Q value. Horizontal movements (left/right) change the Frequency of the band. When you click-hold the mouse button on the display background, horizontal movements will again change the Frequency of the band. Vertical mouse movements will change the Gain of band 2 to 7. The slope values of the highpass and lowpass filters (bands 1 and 8) can only be changed in the parameter area below the graphic display. Click-hold on the parameter: Moving up increases, and down decreases, the value. After boosting or cutting frequency bands, you can use the Master Gain fader to readjust the output level of the Channel EQ. Channel EQ--Analyzer The precision analyzer of the Channel EQ uses Fast Fourier Transformation (FFT) to show the energy of all frequency components of the signal. The central display of the Channel EQ fulfills multiple display functions: it shows both the curve of the FFT analyzer and the EQ curve. An identically scaled frequency axis is shown for both. This allows you to easily recognize unwanted frequencies in the analyzer curve, while using the EQ to edit them accordingly. A click on the Analyzer button activates/deactivates the FFT analyzer. The display directly under the button determines the location of the Analyzer. You can switch the Analyzer pre EQ or post EQ (default) in order to compare the original signal with your edits. Click into the display to open a pull-down menu that defines the resolution of the FFT analyzer--or more accurately, the number of frequency bands. The higher the precision of measurements, the more CPU power is needed. High resolutions are necessary whenever you need reliable results in the very low bass frequency area. The bands derived from FFT analysis are divided in accordance with the frequency linear principle--non-technically, this means that there are far more bands in the highest octave than in the lowest. 30 Chapter 3 Equalizer Use the Scales to the left and right of the EQ display, to change the vertical scale of the EQ and analyzer curves. To increase the resolution of the EQ Gain parameter (dB Warp) in the most interesting area around the zero line, click-hold in the green dB Scale on the left side of the graphic display, and move the mouse up. Moving the mouse down, will decrease the parameter value. The overall range is always ±30, but small values will be easier to recognize. As soon as the Analyzer is activated, you can change the Analyzer Top parameter, which alters the scaling of the FFT analyzer, on the right side of the graphic display. The visible area represents a dynamic range of 60 dB, but by click-holding and vertically dragging, you can adjust the maximum value between +20 dB and -40 dB. The Analyzer display is always dB-linear. Two additional Analyzer parameters are available via the 001/100 view. Analyzer Mode allows you to switch between Peak and RMS. Analyzer Decay allows you to adjust the decay rate (in dB per second) of the Analyzer curve (peak decay in Peak mode or an averaged decay in RMS mode) Note: The FFT analyzer needs additional CPU resources. In fact, resource consumption increases significantly at higher resolutions! We recommend that you disable the Analyzer or close the Channel EQ window after setting the desired EQ parameters. This will free up CPU resources for other tasks. Using the Channel EQ as the Default EQ The Channel EQ replaces the Track EQ of older Logic versions. It is inserted into the first available insert slot by double-clicking the EQ area on the upper portion of mixer channel strips. This area will change to a thumbnail view of the Channel EQ display. The thumbnails provide an overview of the EQ settings used in each individual channel. Chapter 3 Equalizer 31 Linear Phase EQ The extremely high-quality Linear Phase EQ plug-in is almost identical to the Channel EQ. With the exception of the different name and a few different colors, it uses the same familiar eight-band layout, and method of operation, as the Channel EQ. Under-the-hood, however, the Linear Phase EQ uses completely different technology which preserves the phase of the audio signal 100%--even if the wildest EQ curves are applied to the sharpest signal transients! As with all good things in life, there is a catch. The Linear Phase EQ uses more CPU power than the Channel EQ. Another factor is the inherent amount of latency introduced by this technology. Logic's plug-in delay compensation will successfully prevent the worst of these latency artefacts in mixdown situations--but don't even think about playing software instruments live when using the Linear Phase EQ. As the parameters of the Channel EQ and Linear Phase EQ are almost identical, you may freely copy settings between them. For more information on the parameters of the Linear Phase EQ, read up on the "Channel EQ" on page 29. 32 Chapter 3 Equalizer Match EQ The Match EQ plug-in allows you to "match" and transfer the frequency spectrum from , one signal to another, or to store it as a spectral template file. In this way, you can acoustically match the sound of various songs for an album, or impart the "sound" of any reference source onto your own recordings. The alignment of signals is automatic, but you can also manually draw or modify the filter curve to alter the sound as required. Note: Match EQ acoustically matches two audio signals. It does not, however, match any dynamic differences in the two signals. Description of the basic functions Match EQ is a learning equalizer that reads the frequency spectrum of any reference source, including: the input signal, an audio file, or a template. Alternatively, you can load a setting file or import the settings of another Match EQ instance via a copy and paste operation. You can analyze the average audio spectrum of the track the plug-in is assigned to or load another setting file or template. By matching both spectra, a filter curve is generated. This generated curve adapts the track signal to match the sound of the template. If required, you can modify the filter curve by boosting or cutting gain in different frequencies, or inverting it. Further to this, you can manually modify the curve by creating a virtually infinite number of peak filters, and adjust them as required. In this way, you can draw your own filter curve to optimize the sound as required. The internal analyzer allows you to visually check the frequency spectrum of the original data and the resulting curve, making manual corrections at specific points within the spectrum easier. Chapter 3 Equalizer 33 Parameters The View pull-down menu allows you to select the type of information shown on the analyzer display in the center. The following options are available: · Automatic: Depending on the selected function, the analyzer view is automatically toggled between the three following options. · Template: The analyzer display shows the average frequency curve, which is generated by analyzing the input signal or loading a template. · Current Material: The analyzer visualizes the average frequency curve, which is generated by analyzing the track signal or loading a Setting file or template. · Filter: The analyzer displays the filter curve, which is generated by matching the Template and the spectra of the Current Material. Independent of the selection, the analyzer can be activated/deactivated via the On/Off button. The Analyzer Position pull-down menu allows you to place the analyzer tap before (Pre: unchanged) or behind the filter circuit (Post: behind the Match EQ). Note: Deactivating the analyzer frees up processing power for other applications. On stereo channels, the view mode is configured via the lower View toggle menu. You can select whether the analyzer displays both audio channels via separate curves (L&R) or the summed maximum level (LR Max). You can manually modify the filter curve generated via matching the Template with the Current Material. The buttons in the Select section let you choose whether the modifications are applied only to the left, right, or both channels. You can refine this selection via the Channel Link slider. If the slider is set to 1.0, the L and R buttons for the single channels will have no effect, because both channels are represented via a common EQ curve. At the minimum value of 0.0, two separate filter curves are displayed, each of which can be selected for editing via the L and R buttons. The intermediate settings of the Channel Link slider allow you to blend these extreme values as required. As a result, any modification to either of the filter curves is transferred to both channels, dependent on the Channel Link setting. Note: In the mono version of the plug-in, the parameters in the View Mode, Select, and Channel Link sections have no effect. The Template and Current Material buttons perform the spectral analysis of the audio signals, and match the resulting curves. Clicking the Learn button in the Template section starts and stops measurement of the average frequency spectrum in the reference signal. Clicking the Learn button in the Current Material section starts and stops measurement of the average frequency spectrum in the audio material of the track. 34 Chapter 3 Equalizer Note: Audio files can also be dragged onto the Template or Current Material Learn buttons to generate template or current spectra. A progress bar displays the progress of the analysis process. If you right-click (or Control click) on either of the Learn buttons, a context menu opens. This menu allows the spectrum of the template or the track signal (Current Material) to be: · cleared · copied to the Match EQ clipboard, which is common to all Match EQ instances in the current song. · pasted from the Match EQ clipboard to the active instance. · loaded from a stored Setting file. · generated from an audio file (chosen in the File Selector). This is done by choosing the Generate Template/Current Material Spectrum from audio file option, and selecting an appropriate file in the file selector that appears. A progress bar displays the status of the analysis process. Note: When you activate the Learn button in either the Template or Current Material section, the View parameter is set to Auto, and the analyzer will display the current status of the spectral analysis, indicated by a progress bar. The Match button in the Current Material section allows you to write the differences between the learned or loaded Template and the learned or loaded spectrum of the Current Material to a filter curve. Differences in gain are automatically compensated for, with the resulting EQ curve referenced to the 0 dB line. The filter curve is updated automatically each time a new template or current material spectrum is learned or loaded, when the Match button is enabled. You can toggle between the matched (and possibly scaled and/or manually modified) filter curve and a flat response by activating/deactivating the Match button. Note: Each time a new audio spectrum is matched--either by loading/learning a new spectrum while Match is activated or by activating Match after a new spectrum has been loaded--existing manual modifications are discarded, and Apply is set to 100%. Basically, only one of the Learn buttons may be active at a time. As an example, if the Learn button in the Template section is active and you press the Learn button in the Current Material section, the analysis of the template file stops, and the current status is used as the spectral template. Analysis of the track (Current Material) will then begin. Note: If you have manually modified the filter curve, the original (or flat) curve can be restored by Option-clicking on the background of the analyzer display. A second Option-click restores the most recently modified curve. Chapter 3 Equalizer 35 The filter curve can be edited via the Smoothing slider. At a value of 0.0, the filter curve is applied to the track signal without any changes. At all other Smoothing settings, the filters are smoothed at a constant bandwidth. A value of 1.0, for example, means that all filters have a constant bandwidth of one semitone that is used to smooth the notchlike filters in the curve. A bandwidth of: four semitones (a value of 4.0--or a major third), an octave (a value of 12.0) and two octaves with the maximum setting (24.0). Note: Smoothing does not affect any manual modifications of the filter curve. The Apply slider exaggerates (101% to 200%), reduces (99% to 1%) or inverts the peaks/ dips (-1% to -100%) the effect of the filter curve on the track signal. At a value of 100%, the signal is aligned to the curve without any changes. The Phase toggle menu switches the operational principle of the filter curve. · The Linear option prevents processing from altering the signal phase. At the same time, the latency of the plug-in will increase. · The Minimal option alters the signal phase, but latency is reduced. Manual Modifications You can graphically edit the matched filter curve directly in the display. Just click at any point within the filter curve to create a new peak band. You can shift the peak frequency for this band (within the entire spectrum) by dragging the mouse horizontally. Vertically moving the mouse allows you to set the gain of this frequency band (range: -24 to +24 dB). The Q-factor of the filter is set by the vertical distance between the click point and the curve. By clicking on the curve, the maximum Q-value of 10 (for notch-like filters) is used. Clicking above or below the curve decreases the Qvalue. The further you click from the curve, the smaller the value (down to the minimum of 0.3). · The Q -factor can be changed continuously by pressing Shift and moving the mouse up/down while keeping the mouse button pressed. · If Option is hold while releasing the mouse button, the modification is cancelled. Note: The current values are shown in a window within the display while the left mouse button is held down. The colors and modes of the dB scales on the left and right of the display are automatically adapted to the active function. If the analyzer is active, the left scale displays the average spectrum in the signal, while the right scale serves as a reference for the peak values of the analyzer. Basically, the analyzer visualizes a dynamic range of 60 dB. The displayed range can, however, be shifted between the extreme values of +20 dB and -100 dB by click-dragging on the scale. 36 Chapter 3 Equalizer If the resulting filter curve is displayed, the left scale--and the right, if the analyzer is inactive--shows the dB values for the filter curve in an appropriate color. By clickdragging on one of the scales, the overall gain of the filter curve is adjusted in the range from -30 to +30 dB. Fat EQ The high-quality Fat EQ offers up to 5 fully parametric bands--buttons 1 through 5 activate these individually; inactive bands do not drain your computer's resources. The icons above the graphic display let you determine whether Band 1 acts like a highpass filter or a low shelving EQ. Similarly, Band 5 can be switched back and forth between use as a lowpass filter and a high shelving EQ. Bands 2 and 4 can be switched from their normal operating mode (as fully parametric bandpass filters) to low or high shelving EQs. The center band (number 3) always operates as a fully parametric bandpass filter. The shelving filter's slope characteristics for bands 2 and 4 are adjustable via the Q parameter. The area directly below the graphic display (depicting the frequency response curve) is used to select the frequency for the individual bands. Simply click on the number, and change the value with your mouse. You'll be able to hear an individual frequency better if you turn it up by rotating the Cut/Boost knob located below it clockwise. The same holds true for any frequency that you want to attenuate. Once you've located the frequency that you're hunting for, you can back off the Cut/Boost knob level, and set it to the desired value. Use the Q (bandwidth) parameter located in the lower display to determine the extent that the band influences neighboring frequencies. Chapter 3 Equalizer 37 At low Q values, the EQ influences a wider frequency range, and at high Q values, the effect of the EQ band is limited to a very narrow frequency range. Please bear in mind that your perception of an attenuated or boosted frequency depends heavily on the Q parameter: If you're working with a narrow frequency band, you'll generally need to cut or boost it more drastically to notice a difference. Silver EQ The Silver EQ contains one High Shelf, a Parametric and one Low Shelf filter with the corresponding parameters. More on each of these is found in the Individual EQ's section below. DJ EQ The DJ EQ combines Low and High Shelving Filters with a fixed frequency, and one Parametric EQ with its attendant parameters. More on each of these is found in the Individual EQ's section below. The special feature of the DJ EQ is that it allows the gain of the filters to be reduced down to -30 dB. Individual EQs Parametric EQ The Parametric EQ offers the following three parameters: · Hz: Center frequency · dB: Cut/Boost · Q: Quality A symmetrical frequency range on either side of the center frequency is boosted or cut. You can adjust the width of this frequency range with the Q control. High Shelving EQ/Low Shelving EQ · The Low Shelving Equalizer only affects the frequency range below the selected frequency. · The High Shelving Equalizer only affects the frequency range above the selected frequency. 38 Chapter 3 Equalizer 4 Dynamic 4 This chapter introduces Logic's Dynamic plug-ins. This includes the Compressor, Silver Compressor, Expander, Noise Gate, Silver Gate, Enveloper, DeEsser, Limiter, Multipressor, and Adaptive Limiter plug-ins. Compressor A compressor tightens up the dynamics of a signal. This means that the difference in levels between loud and soft passages is reduced. This "evening out" of the loud and soft passages means that the peak level remains pretty constant, and the overall loudness--the perceived volume--of a track is increased. Next to an EQ, a compressor is your most valuable sound-shaping tool when mixing. A compressor is a universal effect, it has a virtually unlimited range of applications. You should definitely exploit it for vocal tracks, but a compressor can also often work wonders for entire mixes. When you use a compressor, be sure to route the entire signal through it, by inserting it directly into channels. It should only be used in a bus when you want to compress a group of tracks (a drum kit, for example) simultaneously, and by the same amount. Again, these tracks (individual drums in a kit, for example) should be routed to the bus in their entirety, as opposed to using Send knobs to route just parts of each signal to the bus. You do this by selecting the appropriate bus as the output destination for the tracks that you wish to compress. 39 Logic's Compressor was designed to emulate the response of the finest analog compressors. It follows the following principle: When a signal exceeds the defined Threshold level, the compressor actually alters the response, so that it is no longer linear. What happens is that all levels that exceed the Threshold are attenuated by the value set with the Ratio slider. A ratio of 4:1 means that an incoming level that is 4 dB louder than the Threshold level is dampened, so that it comes out the other end of the compressor with a level that is just 1 dB above the Threshold level. On the flip side, if you route in a signal that is loud enough to double the output level of the compressor (+6 dB), the input signal would need to have a level 24 dB greater than the Threshold level. This tells us that a compression ratio of 4:1 is a fairly drastic manipulation of the original signal's dynamics. Given that the compressor lowers levels, the volume of its output signal is normally lower than that of the input signal. To compensate for this decrease in levels, the output of the compressor is equipped with a Gain slider. Auto Gain automatically sets the level of amplification to a value equivalent to the "sum of the threshold value minus the threshold value divided by the ratio" or put less confusingly T--(T/R). This function ensures that a normalized input signal is amplified so that the output signal is also normalized, regardless of the values set for Threshold and Ratio--provided you are dealing with relatively static signals. Use the Attack and Release knobs to shape the dynamic response of the compressor. Attack determines the amount of time it takes for the compressor to react to signals that exceed the Threshold. At higher values, the compressor does not fully dampen a signal until it runs through its Attack phase. This type of setting ensures the original attack, for example the sound of a pick or finger striking a guitar string, remains intact or clearly audible. If, on the other hand, you want to maximize the level of a master signal, set the Attack knob to low values, ensuring that the compressor responds more swiftly. Release determines the amount of time it takes for the compressor to stop dampening louder passages, once the signal level falls below the Threshold level. If the compressor generates an ugly pumping sound, adjust the Release knob accordingly. 40 Chapter 4 Dynamic When you have configured a compressor so that it dampens the signal at or above the Threshold value by the predetermined Ratio, while the level just below the Threshold is routed through at a 1:1 Ratio, an audio engineer would term the compression as hard knee. In many cases, however, you'll come up with a better sounding track by using a more gradual transition from the 1:1 Ratio below the Threshold, to the Ratio that you entered for levels above the Threshold. In this scenario, the characteristic curve is not as radical--it rises gradually from the bottom left to the top right, as seen in the graphic display. This type of compression is called soft knee. The Knee slider lets you incrementally select anything from hard to soft knee. This wide range of options provides you with the tools you need to shape the sound as you like; whether you want to radically maximize loudness with absolutely no regard for the original dynamics (hard), or are going for the more musical compression that acoustic recordings typically require (soft). Keep in mind that Knee only controls the shape of the compression, not its intensity; use the Threshold and Ratio sliders for this purpose. Incidentally, the Gain Reduction Meter indicates the intensity of compression used to tighten up the original signal. This feature is a great help, particularly if you're not experienced with using compression. Keep an eye on it to make sure that you're not overly compressing your tracks. When the compressor has to decide whether or not the level exceeds the Threshold (or if the level is getting close to the Threshold, for soft knee compression), it can analyze either the peak or RMS level. The latter value is a better indication of how humans perceive loudness. When you use the compressor primarily as a limiter, select the Peak button. When you're compressing individual signals, use of the RMS button will often deliver better, more musical results. If you activate Auto Gain and RMS simultaneously, the signal may be saturated. If you hear any distortion, switch Auto Gain off, and enter a suitable gain level manually. The Output Clip parameter limits (clips) the output to 0 dB, via the OFF/SOFT/HARD settings. This setting is only available in the Controls view. Note: Despite all of these handy tips for tweaking sounds, you should always keep one thing in mind--there are no hard and fast rules. Use your own taste and ears. If it sounds good, it is good. Chapter 4 Dynamic 41 Silver Compressor The Silver Compressor is a simplified version of the Compressor. It is limited to Threshold, Attack, Release, and Ratio controls. Expander The Expander is similar to the Compressor, with one fundamental difference--it increases, rather than reduces, the dynamic range above the Threshold. The Ratio slider features a value range of 1:1 to 0.5:1. This means that the Expander is a genuine upward expander (as opposed to a downward expander that increases the dynamic range below the Threshold). You can use this effect to emphasize the transients of highly compressed signals. This spices up the sonic image, making it sound livelier and fresher. Please bear in mind the fact that you will perceive the signal as being softer, even when the peak level remains the same. In other words, the expander decreases loudness. If you manipulate the dynamics of a signal fairly radically (depending on the Threshold and Ratio values), you'll find that you'll need to reduce the level via the Gain slider to avoid distortion. In most cases, Auto Gain will take care of this for you. Please check the Compressor section for details on the various parameters. 42 Chapter 4 Dynamic Noise Gate Ordinarily, a noise gate suppresses unwanted noise that may become audible during a lull in the signal. You can, however, also use it as a creative sound-sculpting tool. Here's the basic principle behind a noise gate: Signals that lie above the Threshold are allowed to pass unimpeded (open gate). Anything below the defined Threshold (background noise, crosstalk from other signal sources and so on) is fully muted (a closed gate). In other words, the Threshold slider determines the lowest level that a signal must be at, in order to open the gate--it separates the wanted or useful signal, from the unwanted or noise signal. The Reduction slider allows you to control the intensity of noise suppression. As a rule, you should set it to the lowest possible value and leave it there, to ensure that the gate closes completely. If you prefer, you can select other values, thus reducing the noise signal less dramatically. As an alternative, you can actually boost the signal by up to 20 dB. The three rotary knobs (at the top) influence the dynamic response of the noise gate. If you want the gate to open extremely quickly, say for percussive signals such as drums, set the Attack knob to the lowest value by turning it as far as it will go counterclockwise. If the signal fades in a bit more softly, as is the case with string pads and the like, a noise gate that opens too quickly can wreak havoc with the signal, causing it to sound unnatural. For this type of sonic scenario, set the Attack knob so that the gate emulates the attack of the original signal. Much the same holds true for the Release phase of signals. When you're working with signals that fade out gradually or have longer reverb tails, you should turn the Release knob up, allowing the signal to fade naturally. The Hold knob determines the minimum amount of time that the gate stays open. This knob avoids the dreaded chattering effect caused by a rapidly opening and closing noise gate. The Hysteresis slider provides another option for avoiding chatter, without needing to define a minimum Hold time. Chapter 4 Dynamic 43 Let's back up a bit for a brief explanation: Noise gates often begin chattering when the level of a signal fluctuates slightly, but very rapidly, during the attack or release phase. Instead of clearly exceeding or falling short of the Threshold value, the signal level hovers around the Threshold. The Noise Gate then rapidly switches on and off to compensate, producing the undesirable chattering effect. If you were able to tell the Noise Gate to open at the determined Threshold level and remain open until the level drops below another, lower, predefined Threshold level, you'd be able to avoid chatter--as long as the sonic window formed by these two Threshold values is large enough to contain the fluctuating level of the incoming signal. This is exactly what the Hysteresis feature enables you to do--the value determined by the Hysteresis slider is actually the difference between the Threshold values that open and close the gate. This value is always negative. Generally, -6 dB is a good place to start. If you're dealing with audio material featuring extremely sensitive transients, or attack phases that are critical to the overall sound, you may find it beneficial to have the Noise Gate open up a tad before the useful signal fades in. This is what the Lookahead slider is designed for. The program analyzes the signal level ahead of time, and anticipates the point at which it can open the gate before the signal actually reaches the Threshold value. When you choose to use this feature, please make sure you set the Attack, Hold or Hysteresis controls to appropriate values. When you're working with noise gates, you'll run across scenarios where the useful signal and the noise signal have levels that are near enough to be perceived as identical. A typical example is the crosstalk of a hi-hat--its signal tends to bleed into the snare drum track when you're recording a drum kit. If you're using a noise gate to isolate the snare, you'll find that the hi-hat will also open the gate in many cases. To avoid this effect, the Noise Gate offers Side Chain filters. When you press and hold the Monitor button, you can audition the Side Chain signal. You can then set the filters to only allow frequencies that contain a particularly loud, useful signal to pass. For this example, we'll use the Noise Gate's High Cut filter--that only allows the bottom end and mids of the snare to pass, and cuts the higher frequencies of the hi-hat. When you switch Side Chain Monitoring off, it will be much easier to set a suitable Threshold level. This will be a value that is only exceeded by the level of the louder useful signal--the frequencies that make up the snare's fundamental tone, in our example. Put simply, the Noise Gate only allows the sound of the snare to pass. Should the need arise, you can follow much the same procedure to isolate a kick or snare drum within an entire mixdown. 44 Chapter 4 Dynamic Silver Gate The Silver Gate is a cut-down version of the Noise Gate. It is limited to Threshold, Lookahead, Attack, Hold, and Release controls. Enveloper The Enveloper is an unusual tool that lets you shape transients--the attack and release phases of signals. No other type of dynamic effect (such as a compressor or expander) can achieve similar results--and these results can be quite impressive indeed. The most important Enveloper controls are the two Gain sliders that govern Attack (left) and Release (right). In the center position, the signal remains unprocessed. If you turn the gain up, the attack or release phase is emphasized. If you turn it down, the corresponding phase is attenuated. As an example, boosting the attack lends a drum sound more snap, or amplifies the sound of a guitar string being plucked or picked. When you cut the attack, percussive signals fade in more softly. You can also mute the attack, making it virtually inaudible. This facilitates a range of interesting manipulations. Another handy application for this plug-in is maintaining friendships--it allows you to mask the poor timing of accompanying instruments, rather than tell your pals that they "groove like a bunch of accountants at the office Christmas party" . Chapter 4 Dynamic 45 Emphasizing the release also boosts the amount of any reverb on the affected track. Conversely, when you tone down the release phase, tracks originally drenched in reverb end up sounding drier. This effect is particularly useful when you're working with drumloops, but there are, of course, many other applications. Let your imagination be your guide. When using the Enveloper, you should set the Threshold to the minimum value and leave it there. Only when you seriously crank the release phase, thus boosting the noise level of the original recording, should you turn the Threshold slider up a little. This limits the Enveloper to only influencing the useful signal. Drastic boosting or cutting of the release or attack phase may change the overall level of the signal. The Out Level slider allows you to compensate for this effect. The Time parameters for the attack and release phase (2 knobs below the graphic display) enable you to access the time-based intervals that the plug-in interprets as the attack and release phases. Generally, you'll find values of around 20 ms (attack) and 1500 ms (release) are fine to start with. Adjust them accordingly for the type of signal that you're processing. Similar to its Noise Gate counterpart, the Lookahead slider allows you to define values that tell the Enveloper to anticipate what the signal will do in the very near future. Normally, you won't need to use this feature, except possibly for signals with extremely sensitive transients. If you do decide to use Lookahead, you may need to adjust the attack time accordingly. To give you a better insight into the true nature of the Enveloper, here's a quick look at how it works: It is equipped with two internal envelope followers. One follows the amplitude of the input signal directly, whereas the other follows all changes generated by the variable delays (individually adjustable for attack and release). The difference between the two envelope followers is used to boost or cut the original signal by way of the corresponding Gain sliders (also individually adjustable for attack and release). In contrast to a compressor or expander, the Enveloper operates independently of the absolute level of the input signal--provided the Threshold slider is set to the lowest possible value. 46 Chapter 4 Dynamic DeEsser A DeEsser is a signal processor used for the rejection of hissing, or sibilant noises. This is why it is called a "DeEsser" and occasionally an "S"-Suppressor. You can, of course, reject , sizzling frequencies with an equalizer, but a DeEsser only rejects this high frequency band for as long as a threshold level is being exceeded in a specific frequency band. This "dynamic" ability is why the sound doesn't get darker when no "sizzling" consonants are present in the signal. A DeEsser is a frequency-specific compressor, designed to only compress a particular frequency band within a complex full band signal. It features extremely fast attack and release times. In the Logic DeEsser, the dynamic rejection does not necessarily need to take place in the same frequency range that's being analyzed. Rather, the DeEsser performs a gain reduction in the frequency band displayed in the lower window for as long as the level exceeds a threshold (which falls within the frequency range) displayed in the upper window. Note: Please don't confuse a DeEsser with an effect known as a "Vocal Stressor" The . latter reduces the gain of the entire range when the level exceeds a threshold defined in a given frequency range. This type of processing can be achieved with any compressor with a high pass filter or EQ inserted in its side chain. The Logic DeEsser does not make use of a frequency dividing network (a crossover, utilizing low and high pass filters). Rather, it is based on a subtraction of the isolated frequency band, leaving the phase-curve untouched. The DeEsser is especially important in FM radio applications, because sharp S-type consonants can cause harsh intermodulation distortion noises. The need for this depends very much on the language spoken: English has fewer of these consonants than German or Spanish, for example. Chapter 4 Dynamic 47 Parameters of the DeEsser Detector Frequency This parameter defines the frequency band the DeEsser acts upon. It's not necessarily the same band that will be reduced. Detector Sensitivity This parameter defines the threshold level that needs to be exceeded (around the Detector Frequency), in order to reduce the level around the Suppressor Frequency. Monitor Activation of this switch allows you to monitor the Side Chain signal used by the DeEsser. If you want to reduce sizzling noises, listen to the input signal, and set the Detector Frequency in a way that makes the sizzling frequency range more prominent. (If you find that you like this filtered sound, combine a highpass and lowpass filter--in order to construct a bandpass--as this approach uses less processing power). Suppressor Frequency This parameter defines the frequency band that is reduced when the Detector Frequency Sensitivity threshold is exceeded. Strength Strength sets the amount of gain reduction around the Suppressor Frequency. Smoothing Smoothing controls the reaction speed of the gain reduction start and end phases. It's a combination of attack and release time parameters, as known from compressors. 48 Chapter 4 Dynamic Limiter The Limiter is also a standard effect for processing a summed stereo signal. It is normally used for mastering. You could say that a limiter is a compressor with an infinite compression ratio. The Limiter restricts dynamics to an absolute level. Any input level that exceeds the Limiter's threshold (Gain) will be output at this "limited" level, no matter how much higher the original signal level may have been. The fact that there is a level that the signal cannot exceed is the distinguishing characteristic of a limiter, when compared to a compressor. Parameters of the Limiter Gain Most analog limiters would have a "Threshold" control (like that of the Multipressor), rather than a "Gain" control. This sets the level at which the Limiter will begin to work. As the Limiter is digital, and is normally is applied as the very last mastering tool, we can presuppose that: · the input signal sometimes reaches 0 dB, but does not exceed this value, and · that the Limiter is being used to raise the signal's overall volume. This is the reason why you find a Gain control here--to set the desired level of gain for the signal. The Limiter is designed in such a way that if set to 0 dB Gain and 0 dB Output Level, it doesn't work at all--on normalized regions. If there should be a region that clips (red gain line), the Limiter--using its basic settings--reduces the level before clipping can occur. (This does not apply to data that was clipped during recording). Lookahead Lookahead determines how far the processor looks into the future, in order to react earlier (thus better) to peak volumes. Unlike stand-alone processors, this function does not apply a general signal delay, as the Limiter is not limited to seeing the signal in real-time. Set Lookahead to higher values, if you want the limiting effect to take place before the maximum level is reached. Chapter 4 Dynamic 49 Release Here, you can set the time required by the Limiter (after limiting) to release the effect. Output Level This simple volume control sets the desired maximum level of the Limiter's output signal. Softknee Activate the Softknee button to produce a softer transition from no limiting to full limiting. If switched off, the signal will be limited (following a linear curve) absolutely and exactly when a level of 0 dB is reached. If switched on, the transition to full limiting is non-linear, meaning softer. The limiting of the signal will start before a level of 0 dB is reached. This will avoid distortion artefacts occurring when strong limiting is used without softknee. Graphic Display The graphic display shows the reduction of the level (starting from 0 dB downwards). Adaptive Limiter With its Compressor (see "Compressor" on page 39), Multipressor (see "Multipressor" on page 52) and Limiter (see "Limiter" on page 49) Logic features several extremely versatile options for increasing perceived volumes. A further tool which can be used to increase the perceived level of signals is the Adaptive Limiter plug-in. In the world of analog processors, it could be more closely compared to a clipper, rounding and smoothing only harsh level peaks, rather than to a VCA-type limiter. It allows you to achieve maximum gain, without having to fear exceeding 0 dBFS. The Adaptive Limiter may slightly color the sound. An effect most similar to an amplifier when driven hard. 50 Chapter 4 Dynamic