User manual APPLE LOGIC EXPRESS 7 - PLUG-IN REFERENCE

Logic Express 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 7 8 11 11 14 16 17 17 19 19 20 23 23 24 24 25 27 27 30 30 32 33 34 37 37 39 39 40 41 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 Silver EQ DJ EQ Individual EQs Dynamic Compressor Silver Compressor Noise Gate Silver Gate Limiter Preset Multipressor Distortion Guitar Amp Distortion Overdrive Bitcrusher Clip Distortion Chapter 1 Chapter 2 Chapter 3 Chapter 4 Chapter 5 3 42 Chapter 6 43 43 46 48 48 49 49 50 52 53 53 54 55 55 57 57 59 59 60 61 64 67 67 68 71 71 72 74 75 75 76 76 81 81 82 82 Phase Distortion Filter AutoFilter Fuzz-Wah High Cut/Low Cut High Pass/Low Pass Filter Delay Sample Delay Tape Delay Stereo Delay Modulation Modulation Delay Chorus Flanger Phaser Tremolo Spreader Reverb AVerb SilverVerb GoldVerb PlatinumVerb Special Pitch Shifter II Denoiser Helper Tuner Gain Levelmeter Synthesizer Basics Analog and Subtractive What Is Synthesis? Subtractive Synthesis EFM 1 Concept and Function Global Parameters FM Parameters Chapter 7 Chapter 8 Chapter 9 Chapter 10 Chapter 11 Chapter 12 Chapter 13 4 Contents 84 85 Chapter 14 87 87 89 89 93 93 95 95 103 105 106 107 111 134 135 136 139 139 141 161 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 KlopfGeist EXSP24 Using Instruments File Organization Sample File Import EXSP24 Key Commands A Brief History of Sampling MIDI Controller List GarageBand Instruments About GarageBand Instruments Chapter 15 Chapter 16 Chapter 17 Chapter 18 Chapter 19 Chapter 20 Glossary Index Contents 5 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. Given a fast enough computer, you could conceivably arrange and mix an entire song using several software instruments, such as Logic's ES1, or EXSP24, 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 7 Introducing Logic's Plug-ins Logic's plug-ins include the following features: · Real-time processing of audio. · Support for sample rates up to 96 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 and Audio Unit 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. 8 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 9 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. 11 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. 12 Chapter 1 Basics You can set all plug-in parameters in the Plug-in window. For further information please read "The Plug-in Window" on page 14. Closing the Plug-in window leaves the plug-in active. 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 13 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 m 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 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. 14 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 EXSP24 instance was loaded on Audio Instrument channel 1, switching between these tracks would automatically update the Plug-in window to show the ES1/EXSP24, 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 16. Switching the Contents of the Plug-in Window You can reassign any open Plug-in window--in two different ways--via the two pull- down 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. 001/ 011 Button Some Logic plug-ins may have 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. Chapter 1 Basics 15 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. 16 Chapter 1 Basics 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. 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. 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. Chapter 1 Basics 17 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 11.). 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 4 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. 19 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. 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. 20 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 24 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 21 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 EXSP24 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. 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 EXSP24 sampler and others) by ±100 cents. Note: Some instruments do not recognize this remote command. 22 Chapter 2 Instruments and Effects 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, 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 four frequency bands. 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 4 is a highpass filter. Note: The Q-parameter of band 1 and band 4 will have no effect when using a slope of 6 dB/Oct. Bands 2 and3 are bandpass filters. 23 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 1 to 4. 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. 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. 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. 24 Chapter 3 Equalizer 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. Chapter 3 Equalizer 25 4 Dynamic 4 This chapter introduces Logic's Dynamic plug-ins. This includes the Compressor, Silver Compressor, Noise Gate, Silver Gate, Limiter, and Preset Multipressor 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. 27 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. 28 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 if the 001/011 button is activated. 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 29 Silver Compressor The Silver Compressor is a simplified version of the Compressor. It is limited to Threshold, Attack, Release, and Ratio controls. 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. 30 Chapter 4 Dynamic 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. 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. Chapter 4 Dynamic 31 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. 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. 32 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 33 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). Preset Multipressor The Preset Multipressor is an easy-to-use variant of Logic Pro's Multipressor plug-in. A multi-band compressor splits the incoming signal into different frequency bands before applying compression. These frequency bands are then compressed independently. Following compression, the frequency bands are mixed back together, and sent out of the plug-in. The aim of independent compression on different frequency bands is to reach high compression levels on the bands that need it, without the pumping effect (on other bands) normally heard at high compression levels. 34 Chapter 4 Dynamic The interface of the Preset Multipressor features 12 radio buttons that allow you to choose between settings optimized for various genres; the names of the presets are pretty much self-explanatory. Make use of the different presets and use your ears to determine which one best fits your needs. Chapter 4 Dynamic 35 5 Distortion 5 This chapter introduces you to Logic's distortion effects. This includes the Distortion, Overdrive, Bitcrusher, Clip Distortion, Phase Distortion, Distortion II, and Guitar Amp effect plug-ins. Guitar Amp The Guitar Amp plug-in simulates the sound of several famous guitar amplifiers. You can process guitar signals directly within Logic, allowing you to reproduce the sound of high-quality guitar amplification systems. Guitar Amp can also be used for experimental sound design and processing. You can freely use the plug-in on other instruments, as desired--applying the sonic character of a guitar amp to a trumpet or vocal part, for example! 37 Guitar Amp offers a range of Amplifier and EQ models that can be combined in a number of ways. The EQ models are equipped with the Bass, Mid, and Treble controls typical of guitar amplifiers. Four different amplifier models can be accessed via the Model radio buttons at the top. · British Clean--simulates the classic British Class A combos which have been continuously produced since the 1960s to the present, without any significant modification. This model is ideally suited for clean or crunchy rhythm parts. · British Gain--reproduces the sound of a British tube head, and is synonymous with rocking, powerful rhythm parts and lead guitars with a rich sustain. · American Clean--emulates the traditional full tube combos used for clean and crunchy sounds. · American Gain--emulates a modern Hi-Gain head, making it suitable for distorted rhythm and lead parts. The entries in the pull-down menu at the top are Settings that refer to the simulated Amp models. Accordingly, the British Crunch, British Lead, American Clean, Crunch, and OverDrive select the appropriate Amp models, and apply new EQ and other settings. You can, however, combine any Amp model with any EQ or other parameter settings, as required. At the top of the slider section, you will find the Pre Gain control, used to set the preamplification level of the input signal. This control has different effects, dependent on the selected Amp model. As an example: A maximum Pre Gain setting produces a powerful crunch sound when used in conjunction with the British Clean Amp model, but the same Pre Gain setting results in a heavy distortion--suitable for lead sounds-- with the British or American Gain Amp models. Directly below, you will find the Low, Mid, and High controls. Use of these sliders allows you to adjust the frequency ranges of the bass, mids, and treble as desired. Presence is an additional high frequency control which exclusively affects the output stage (Master) of the Guitar Amp plug-in. The Master slider controls the output volume of the amplifier (to the "speaker"). Typically, in tube amplifiers, an increase in the Master control level produces a selfcompressed and saturated sound, along with increased level, resulting in a more distorted and powerful amp signal. In the analog domain, this results in an extreme increase in loudness. In Guitar Amp, the Master control influences the sonic character. The Output Level slider serves as a final level control for Guitar Amp's output. It can be viewed as a volume control "behind the cabinet" and is used to set the level that is fed , into ensuing plug-in slots on the channel, or into the channel output. Note: This parameter is very distinct from the Master control, which serves a dual purpose--for sound design, as well as controlling the level of the Amp section. 38 Chapter 5 Distortion Distortion This distortion effect simulates the lo-fi dirt generated by a bipolar transistor. Move the Drive slider up to increasingly saturate the transistor. Generally, the distortion created by the plug-in tends to increase the signal level, an effect that you can compensate for with the Output slider. The Tone knob filters the harmonics-laden distortion signal, delivering a somewhat less grating, softer tone. The Distortion Eye is watching--it visually represents the Drive and Tone parameter settings. Overdrive The Overdrive effect emulates the distortion of a field-effect transistor (FET). When saturated, FETs generate warmer sounding distortion than bipolar transistors. The Drive slider pushes the transistor over the edge and into overdrive. Generally, the distortion created by the plug-in tends to increase signal levels, an effect that you can compensate for with the Output slider. The Tone knob lets you filter the harmonics-laden distortion signal, which delivers an even warmer sound. The Distortion effect's Eye visually represents the settings of the Drive and Tone parameters. Chapter 5 Distortion 39 Bitcrusher Bitcrusher is the ultimate digital distortion box. You can do all kinds of wild stuff with it, such as recreate the 8-bit sound of the pioneering days of digital audio, create artificial aliasing by dividing the sample rate, or distort signals so radically that they are rendered unrecognizable. Warning: The Bitcrusher can damage your hearing (and speakers) when operated at high volumes. The Drive slider boosts the level at the input of the Bitcrusher. Please note that this tends to excite the clipping stage located at the output of the Bitcrusher as well. The Resolution knob allows you to reduce the resolution from 24 bits down to 1 bit. The number of bits is always an exponent of two. The range of available values is equivalent to the exponents of two that a given sample rate can handle. As an example, 65,536 different values are possible for 16 bits, whereas at 8 bits, you're left with just 256. The sonic image becomes ever more ragged as the values decrease because the number of sampling errors increases, thus generating more distortion. At extremely low bit resolutions, the amount of distortion can be greater than the level of the usable signal. The Downsampling slider lowers the sample rate. As an example, at a value of 2 (halved), the original 44.1 kHz signal is sampled at a rate of just 22.05 kHz. At a factor of 10, the rate is knocked all the way down to 4.41 kHz. The Clip Level slider lets you define the point below the normal threshold that you want the signal to start clipping. The Mode buttons are used to determine whether the signal peaks that exceed the clip level are Folded, Cut, or Displaced (check out the graphics on the buttons and the resulting waveform in the display). The kind of clipping that occurs in standard digital systems is usually closest to that of the center mode (Cut). Internal distortion may generate clipping similar to the types generated by the other two modes. 40 Chapter 5 Distortion Clip Distortion The Clip Distortion plug-in is a non-linear distortion effect that produces unpredictable spectra. Beyond drastic distortions, it's well suited for the simulation of warm tube overdrive sounds. The best way to learn what effect the various parameters have is to experiment with them on different signal sources. As a starting point, the following describes what each control basically does: The signal is first amplified by the Drive value, which is a simple gain control. The signal then passes through a highpass filter. The filter's cutoff frequency is determined by the Tone control. The actual non-linear distortion process is controlled by the Symmetry parameter. Once the signal has been distorted asymmetrically, the signal passes through a lowpass filter. This filter's cutoff frequency is determined by the Filter fader. The Mix parameter combines the effected signal with the dry signal. This mixed signal then passes through yet another lowpass filter, where the cutoff frequency is controlled by the Sum Filter parameter. All filters have a slope of 6 dB/Oct. The last stage of signal processing is a tunable shelving filter. If you set its Frequency to about 12 kHz, it will behave like a normal treble control, as found in any mixer's channel strip or stereo hi-fi amplifier. Unlike such treble controls, this filter allows for boosts or cuts of up to ±30 dB (Gain parameter). This somewhat unorthodox combination of serially connected filters allows for gaps in the frequency spectra that can sound quite good with this sort of non-linear distortion. The clip circuit graphic visually depicts every parameter, with the exception of the shelving filter controls. If you activate the 001/011 button, you have access to two more parameters: Input Gain and Output Gain. These can be used to raise/lower the input and output signal levels by up to 30 dB. Chapter 5 Distortion 41 Phase Distortion The Phase Distortion plug-in is based on a modulated delay line, much like the wellknown chorus or flanger effects. As opposed to these, the delay time is not modulated by a low frequency oscillator (LFO), but rather by a lowpass-filtered version of the audio input signal itself. This is how the signal modulates its own phase position. In the signal flow of this effect, the parameters do the following: The input signal only passes the delay line and is not affected by any other process. Mix blends the effected signal with the original signal. The delay time is modulated by a Side Chain signal--namely, the input signal. The input signal passes through a resonant lowpass filter, the Cutoff frequency and Resonance of which can be set with dedicated controls. You also can listen to the filtered Side Chain (instead of the Mix signal), if you engage Monitor. The maximum delay time is set with Max Modulation. The amount of modulation itself is controlled with Intensity. In you active the 001/011 button, the Phase Reverse parameter will be shown. It is only valid for the stereophonic version of the effect. Normally, a positive input value results in a longer delay time. If you engage Phase Reverse (On), positive input values result in a reduction of the delay time on the right channel only. 42 Chapter 5 Distortion 6 Filter 6 This chapter covers Logic's filter effects. The filter effects include the AutoFilter, Fuzz-Wah, Low/High Pass Filter, and Low/High Cut plug-ins. AutoFilter The AutoFilter is an extremely versatile, resonance-capable lowpass filter, that offers a couple of truly unique features. The most important parameters are located to the right side of the Plug-in window: The Cutoff Freq. knob determines the point where the filter kicks in. Higher frequencies are attenuated, lower frequencies are allowed to pass through. 43 The Resonance knob emphasizes the frequency range surrounding the cutoff frequency. When you turn the Resonance up sufficiently, the filter itself begins oscillating (at the cutoff frequency). Self-oscillation is initiated before you max out the Resonance parameter, just like the filters on the legendary Minimoog. When working with Resonance, the manner in which the lowpass filter allows frequencies to pass changes: higher Resonance values cause the filter to cut the bottom end, making the signal sound thinner. The Fatness parameter compensates for this audio artefact. When you turn Fatness up to its maximum value, the Resonance setting has no effect on the response of the frequencies below the cutoff frequency. The Slope buttons determine the steepness of the lowpass filter: frequencies above the cutoff frequency are dampened by 6, 12, 18, or 24 dB per octave (in audio jargon, these are called filters of the 1st, 2nd, 3rd, and 4th order). Even though the 24 dB filter is largely the component of choice for synthesizer designers, be sure to experiment with the other options, as they can also deliver pretty hip results. The Distortion Input and Output parameters allow you to individually control each of the two distortion units-- one pre-input and the other post-output. Although the two distortion modules are identical, their respective positions in the signal chain--before and after the filter, respectively--enable them to generate remarkably different sounds. All other AutoFilter parameters are used to dynamically modulate the cutoff frequency. These fall into two sections: Envelope (ADSR, Envelope Generator) and LFO (Low Frequency Oscillator, Modulation Generator). The Threshold parameter applies to both sections, and analyzes the level of the input signal. If the input signal level exceeds that of the variable Threshold level, the envelope and LFO are retriggered. The Modulation slider of each section determines the intensity of the control signal's effect on the cutoff frequency. Envelope: when the Threshold level is exceeded, the control signal is triggered at the minimum value. Following a variable interval, the length of which is determined by the Attack parameter, the signal reaches its maximum value. It drops in level during the interval defined by the Decay value, and ends up at the Sustain value. Once the signal level drops below the Threshold value, it falls all the way to its minimum value over the time determined by the Release parameter. If the input signal falls below the Threshold level before the control signal has reached the Sustain level, the Release phase is triggered. The Dynamic Modulation parameter lets you modulate the peak value of the Envelope section, by using the level of the input signal. 44 Chapter 6 Filter LFO: the wave shape used for LFO oscillation is determined by the Waveform buttons. The choices are: descending sawtooth (saw down), ascending sawtooth (saw up), triangle, pulse wave, or random (random values, Sample & Hold). Once you've selected a waveform, you can shape the curve with the Pulsewidth knob. Use the Frequency knobs to define the desired LFO frequency: Coarse sets a value between 0.1 and 10,000 Hz, Fine lets you adjust it in smaller increments. The Speed Mod. (Speed Modulation) knob is used to modulate the LFO frequency independently of the input signal level. If the input signal exceeds the Threshold level, the modulation width of the LFO increases from 0 to the value specified for Modulation. You can also define the amount of time this process takes, by entering the desired value with the Delay knob. If the Sync button is activated, the waveform is started at 0° as soon as the Threshold is exceeded. Whenever you use the AutoFilter as a stereo plug-in, you can determine the phase relationships of the LFO modulations on the two stereo sides, with the Stereo Phase knob. If you active the 001/011 button of the Autofilter plug-in, you will have access to the following five parameters: The Volume parameter can lower the Volume by as much as -50 dB, allowing you to compensate for higher levels when using Distortion, for example. If you switch Beat Sync to On, the LFO is synchronized to the sequencer's tempo. The speed values include bar values, triplet values and more. These are determined by the Rate slider directly below Beat Sync. Use Sync Phase to shift the phase relationship between the LFO and the sequencer. Dry Signal sets the level ratio/portion of the non-effected (dry) signal. Chapter 6 Filter 45 Fuzz-Wah The Fuzz-Wah effect is the standalone plug-in version of the Logic Pro 7 EVD6's Wah effect. Its parameters are outlined below. Parameters of the Fuzz-Wah FX Order This parameter allows to you select the order in which the Fuzz/Wah effects are placed. Choices are: Fuzz ­Wah or Wah­Fuzz. Wah Mode There are simulations of several classic wah effects, as well as some basic filter types available. Available models are: off, ResoLP, ResoHP, Peak, CryB, Morl1, Morl2. Wah Level Can be used to adjust the level of the wah-filtered signal, relative to the original level. Also see the Auto Gain section below. Auto Gain While sweeping through the main formants of the input signal, the output level of the Wah may vary wildly, which is not always desirable. Activating the Auto Gain parameter will automatically compensate for this side-effect. Range: on/off · · · · To hear the difference Auto Gain can make: Switch Auto Gain to on. Raise the effect level to a value just below the mixer's clipping limit. Make a sweep with a high relative Q setting. Now switch Auto Gain to off, and repeat the sweep. 46 Chapter 6 Filter Warning: Please take care while doing this, or your ears and speaker system may be damaged. Relative Q The quality of the main filter peak can be increased/decreased, relative to the model setting, thereby obtaining a sharper/softer wah sweep. When set to a value of 0, the original setting of the model is active. Range: -1.00 to +1.00 (0.00 is the default) Pedal Range Common MIDI foot pedals have a much larger mechanical range than most classic Wah pedals. The exact sweep range of the wah filter effected by the MIDI foot pedal is set with the Pedal Range parameters. The highest and lowest possible value reached by the pedal is graphically represented by a gray bracket around the Pedal Position fader (see below). The left and right limit is set by clicking and moving it with the mouse. Additionally both values can be moved simultaneously by clicking in the center of the bracket and moving it to the left or right. Pedal Position This parameter represents the current position of the Wah pedal. To control and automate the Pedal Position via an external MIDI controller for example a MIDI pedal, your Logic Environment has to be prepared accordingly. AutoWah Depth In addition to using MIDI foot pedals (see above), the wah effect can be controlled using the Auto Wah facility. The sensitivity of the Auto Wah can be set with the Depth parameter. Range: 0.00 to 100. Chapter 6 Filter 47 AutoWah Attack/Release These parameters allow you to define how much time it takes for the Wah filter to open and close. Range (in milliseconds): 10 to 10,000 Comp Ratio The Comp Ratio of the integrated compressor can be adjusted between 1:1 (no compression) and 30:1. The Compressor is tied to the Fuzz effect, and always precedes it. As such, the FX Order parameter is very important for placement of the Compressor in the effects chain. Fuzz Gain Controls the level of Fuzz (distortion). Range: 0 dB to 20 dB. Fuzz Tone The integrated Fuzz effect can be adjusted, tonally, with this parameter. Range: 2000 Hz to 20,000 Hz High Cut/Low Cut · The Low Cut filter attenuates the frequency range below the selected frequency. · The High Cut filter attenuates the frequency range above the selected frequency. High Pass/Low Pass Filter · The High Pass Filter affects the frequency range below the set frequency. Higher frequencies pass through the filter. You can use the High Pass Filter to completely get rid of the bass range below a selectable frequency. · The Low Pass Filter affects the frequency range above the selected frequency. Lower frequencies pass through the filter. You can use the Low Pass Filter to completely get rid of the treble range above a selectable frequency. 48 Chapter 6 Filter 7 Delay 7 This chapter describes Logic's delay effects. This includes the Sample Delay, Tape Delay, and Stereo Delay plug-ins. Sample Delay This plug-in allows the simple delaying of a channel by single sample values. The stereo version of the plug-in provides separate controls for each channel. This plug-in (when used in conjunction with the phase inversion capabilities of the Gain plug-in) is particularly suited to the correction of run-time problems that may occur with multichannel microphones. Every sample (at a frequency of 44.1 kHz) is equivalent to the time taken for a sound wave to travel 7.76 millimeters. Looked at differently: If you delay one channel of a stereo microphone by 13 samples, this will emulate an acoustic (microphone) separation of 10 centimeters. 49 Tape Delay The Tape Delay simulates a vintage tape echo device, although with some very useful features that such old devices never offered. The first of these is that it's delay settings are variable in musical increments. It is equipped with a highpass and lowpass filter in the feedback circuit, as well as a circuit that simulates tape saturation effects. This plugin is ideal for the dub delays invented by Jamaican toast masters, and used in many styles of music today. Switching the Sync button on forces the plug-in to use the internal tempo of the sequencer. Tempo information is updated in the plug-in window when you open it, and every time you subsequently execute a mouse operation. The plug-in can even handle tempo changes. The Tempo parameter field serves solely to display the current bpm value--you can't use it to change the tempo of the sequencer. When you want to create dotted note values, move the Groove slider all the way to the right to 75%; for triplets, select the 33.33% setting. Note that all intermediate values are possible. You can view the current delay value in the Delay parameter field. Disengage Sync if you would like to adjust the delay time independently of the song tempo (or change the song tempo without changing the delay time). In this mode, the bpm or ms values can be altered freely by clicking in the Tempo parameter field, while dragging up or down with the mouse. Note when changing the ms values using the left portion of the Delay parameter field, the ms values will increment in large steps, while using the right portion of the field will increment the ms values in small steps. As you might expect, the Feedback slider determines feedback intensity; in other words, the amount of delayed and filtered signal that is routed back to the input of the Tape Delay. When you set it to the lowest possible value, the Tape Delay generates a single echo. If Feedback is turned all the way up, the echoes are repeated ad infinitum. Keep in mind that the levels of the original signal and its taps (echo repeats) tend to add up, and may cause distortion. This is where the internal tape saturation circuit comes to the rescue--it can be used to ensure that these overdriven signals sound good. The Freeze parameter captures the current delay repeats and sustains them until the Freeze parameter is released. 50 Chapter 7 Delay