Apple Logic Express - 9 Installation Manual

Logic Express 9
Effects
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Contents

An Introduction to the Logic Express Effects7Preface
About the Logic Express Effects7 About the Logic Express Documentation10 Additional Resources10
Amps and Pedals13Chapter 1
Amp Designer13 Bass Amp30 Guitar Amp Pro31 Pedalboard37
Delay Effects53Chapter 2
Echo54 Sample Delay54 Stereo Delay55 Tape Delay57
Distortion Effects59Chapter 3
Bitcrusher60 Clip Distortion61 Distortion Effect62 Distortion II63 Overdrive63 Phase Distortion64
Dynamics Processors67Chapter 4
Types of Dynamics Processors67 Compressor69 DeEsser72 Ducker74 Enveloper77 Expander79 Limiter80 Noise Gate81
3
Preset Multipressor83 Silver Compressor84 Silver Gate85
Equalizers87Chapter 5
Channel EQ88 DJ EQ91 Fat EQ92 Single-Band EQs93 Silver EQ95
Filter Effects97Chapter 6
AutoFilter97 EVOC 20 Filterbank103 EVOC 20 TrackOscillator107 Fuzz-Wah119 Spectral Gate123
Imaging Processors127Chapter 7
Direction Mixer127 Stereo Spread130
Metering Tools133Chapter 8
BPM Counter133 Correlation Meter134 Level Meter Plug-in134 Tuner135
Modulation Effects137Chapter 9
Chorus Effect138 Ensemble Effect138 Flanger Effect140 Microphaser140 Modulation Delay141 Phaser Effect143 Ringshifter144 Rotor Cabinet Effect150 Scanner Vibrato Effect152 Spreader154 Tremolo Effect155
Pitch Effects157Chapter 10
Pitch Correction Effect157 Pitch Shifter II161
4 Contents
Vocal Transformer162
Reverb Effects167Chapter 11
Plates, Digital Reverb Effects, and Convolution Reverb168 AVerb168 EnVerb169 GoldVerb172 PlatinumVerb175 SilverVerb179
Specialized Effects and Utilities181Chapter 12
Denoiser181 Enhance Timing183 Exciter184 Grooveshifter185 Speech Enhancer187 SubBass188
Utilities and Tools191Chapter 13
Gain Plug-in191 I/O Utility192 Test Oscillator194
5Contents

An Introduction to the Logic Express Effects

Logic Express has an extensive range of digital signal processing (DSP) effects and processors that are used to color or tonally shape existing audio recordings, software instruments, and external audio sources—in real time. These will cover almost every audio processing and manipulation need you will encounter in your day-to-day work.
The most common processing options include EQs, dynamic processors, modulations, distortions, reverbs, and delays.
Less common are simulations of amplifiers and speaker cabinets, which enable you to “play” your instruments or other signals through a range of vintage and modern sound reproduction systems. Guitarists will also benefit from a number of classic pedal effect emulations.
Further advanced features include precise signal meters and analyzers, a test tone generator, noise reduction, pitch correction, imaging, bass enhancement, andtime-altering processors and utilities.
Preface
As you can see, many of the included processors and utilities don’t really fall into the “effects” category, but they may prove to be invaluable in your studio.
All effects, processors, and utilities provide an intuitive interface that simplifies operation, enabling you to work quickly. Outstanding audio quality is assured when needed, or—at the other end of the spectrum—extreme processing is possible when you need to radically alter your audio. All effects and processors are highly optimized for efficient CPU usage.
This preface covers the following:
About the Logic Express Effects (p. 7)
About the Logic Express Documentation (p. 10)
Additional Resources (p. 10)

About the Logic Express Effects

Logic Express includes a comprehensive suite of effects processors and utilities that can be used to enhance your music projects. Effects are grouped in the following categories.
7
Included effectsEffect category
Amp DesignerAmp Modeling
Bass Amp
Guitar Amp Pro
Pedalboard
EchoDelay
Sample Delay
Stereo Delay
Tape Delay
BitcrusherDistortion
Clip Distortion
Distortion Effect
Distortion II
Overdrive
Phase Distortion
CompressorDynamics
DeEsser
Ducker
Enveloper
Expander
Limiter
Noise Gate
Preset Multipressor
Silver Compressor
Silver Gate
Channel EQEQ
DJ EQ
Fat EQ
Single-Band EQs
Silver EQ
AutoFilterFilter
EVOC 20 Filterbank
EVOC 20 TrackOscillator
Fuzz-Wah
Spectral Gate
8 Preface An Introduction to the Logic Express Effects
Included effectsEffect category
Direction MixerImaging
Stereo Spread
BPM CounterMetering
Correlation Meter
Level Meter Plug-in
Tuner
Chorus EffectModulation
Ensemble Effect
Flanger Effect
Microphaser
Modulation Delay
Phaser Effect
Ringshifter
Rotor Cabinet Effect
Scanner Vibrato Effect
Spreader
Tremolo Effect
Pitch Correction EffectPitch
Pitch Shifter II
Vocal Transformer
AVerbReverb
EnVerb
GoldVerb
PlatinumVerb
SilverVerb
DenoiserSpecialized
Enhance Timing
Exciter
Grooveshifter
Speech Enhancer
SubBass
Gain Plug-inUtility
I/O Utility
Test Oscillator
9Preface An Introduction to the Logic Express Effects

About the Logic Express Documentation

Logic Express comes with various documents that will help you get started as well as provide detailed information about the included applications.
Logic Express User Manual: This onscreen manual provides comprehensive instructions for using Logic Express to set up a recording system, compose music, edit audio and MIDI files, and output audio for CD productions.
Exploring Logic Express: This booklet provides a fast-paced introduction to the main features and tasks in Logic Express, encouraging hands-on exploration for new users.
Logic Express Control Surfaces Support: This onscreen manual describes the configuration and use of control surfaces with Logic Express.
Logic Express Instruments: This onscreen manual provides comprehensive instructions for using the powerful collection of instruments included with Logic Express.
Logic Express Effects: This onscreen manual provides comprehensive instructions for using the powerful collection of effects included with Logic Express.
Logic Express Working with Apogee Hardware: This onscreen manual describes the use of Apogee hardware with Logic Express.

Additional Resources

In addition to the documentation that comes with Logic Express, there are a variety of other resources you can use to find out more.
Release Notes and New Features Documents
Each application offers detailed documentation that covers new or changed features and functions. This documentation can be accessed in the following way:
• Open the application Help menu and choose Release Notes or New Features.
Logic Express Website
For general information and updates, as well as the latest news on Logic Express, go to:
http://www.apple.com/logicexpress
Apple Service and Support Websites
For software updates and answers to the most frequently asked questions for all Apple products, go to the general Apple Support webpage. You’ll also have access to product specifications, reference documentation, and technical articles about Apple products and products from other companies.
http://www.apple.com/support
For software updates, documentation, discussion forums, and answers to the most frequently asked questions for Logic Express, go to:
http://www.apple.com/support/logicexpress
10 Preface An Introduction to the Logic Express Effects
For discussion forums for all Apple products from around the world, where you can search for an answer, post your question, or answer other users’ questions, go to:
http://discussions.apple.com
11Preface An Introduction to the Logic Express Effects

Amps and Pedals

1
Logic Express features an extensive collection of guitar and bass amplifiers and classic pedal effects. You can play live—or process recorded audio and software instrument parts—through these amps and effects.
The amplifier models re-create vintage and modern tube and solid-state amps. Built-in effect units, such as reverb, tremolo, or vibrato, are also reproduced. Accompanying the amplifiers are a variety of emulated speaker cabinets, which can be used as a matching set or combined in different ways to create interesting hybrids.
Also emulated are a number of “classic” foot pedal effects—or stompboxes—that were, and remain, popular with guitarists and keyboardists. As with their real-world counterparts, you can freely chain pedals in any order to create the perfect sound.
This chapter covers the following:
Amp Designer (p. 13)
Bass Amp (p. 30)
Guitar Amp Pro (p. 31)
Pedalboard (p. 37)

Amp Designer

Amp Designer emulates the sound of over 20 famous guitar amplifiers and the speaker cabinets used with them. Each preconfigured model combines an amp, cabinet, and EQ that re-creates a well-known guitar amplifier sound. You can process guitar signals directly, which allows you to reproduce the sound of your guitar played through these amplification systems. Amp Designer can also be used for experimental sound design and processing. You are free to use it with other instruments, applying the sonic character of a guitar amp to a trumpet or vocal part, for example.
The amplifiers, cabinets, and EQs emulated by Amp Designer can be combined in a number of ways to radically or subtly alter the tone. Virtual microphones are used to pick up the signal of the emulated amplifier and cabinet. You can choose from three different microphone types, and you can reposition them.
13
Amp parameters
Amp parameters
Effects parameters
Microphone parameters
Amp Designer also emulates classic guitar amplifier effects, including spring reverb, vibrato, and tremolo.
The Amp Designer interface can be broken down into four general sections in terms of different kinds of parameters.
Model parameters: The Model pop-up menu is found at the left of the black bar at the bottom. It is used to choose a preconfigured model, consisting of an amplifier, a cabinet, an EQ type, and a microphone type. See Choosing an Amp Designer Model. The model-customizing parameters on the black bar allow you to independently choose the type of amplifier and cabinet. See Building a Customized Amp Designer Combo. The EQ type is chosen from the EQ pop-up menu above the Bass, Mids, and Treble knobs in the knobs section. See Using Amp Designer’s Equalizer.
Amp parameters: Located at each end of the knobs section, these parameters are used to set an amp’s input gain, presence, and output level. See Using Amp Designer’s Gain,
Presence, and Master Controls.
Effects parameters: Located in the center of the knobs section, these parameters allow you to control the integrated guitar effects. See Getting to Know Amp Designer’s Effects
Parameters.
Microphone parameters: Located slightly above the right end of the black bar at the bottom, these parameters are used to set the type and position of the microphone that captures the amplifier and cabinet sound. See Setting Amp Designer Microphone
Parameters.
14 Chapter 1 Amps and Pedals
To switch between full and smaller versions of the interface
Click here in full interface.
Click here in
small interface.
Click the disclosure triangle between the Cabinet and Mic pop-up menus in the full
µ
interface to switch to the smaller version. To switch back to the full interface, click the disclosure triangle beside the Output field in the small interface. You can access all the parameters, with the exception of microphone selection and positioning, in the small interface.

Choosing an Amp Designer Model

You can choose a preconfigured model—consisting of an amplifier, a cabinet, an EQ type, and a microphone type—from the Model pop-up menu at the left end of the black bar at the bottom of the Amp Designer interface. Your choices include several combinations in each of the following categories:
Tweed Combos
Classic American Combos
British Stacks
British Combos
British Alternatives
Metal Stacks
Additional Combos
15Chapter 1 Amps and Pedals

Tweed Combos

The Tweed models are based on American combos from the 1950s and early 1960s that helped define the sounds of blues, rock, and country music. They have warm, complex, clean sounds that progress smoothly through gentle distortion to raucous overdrive as you increase the gain. Even after half a century, Tweeds can still sound contemporary. Many modern boutique amplifiers are based on Tweed-style circuitry.
DescriptionModel
Small Tweed Combo
Large Tweed Combo
Mini Tweed Combo
Tip: Tweed combos respond beautifully to your playing dynamics. Adjust the knobs to create a distorted sound, then reduce the level of your guitar’s volume knob to create a cleaner tone. Turn up your guitar’s volume knob when the time comes for a scorching solo.
A 1 x 12" combo that transitions smoothly from clean to crunchy, making it a great choice for blues and rock. For extra definition, set the Treble and Presence controls to a value around 7.
This 4 x 10" combo was originally intended for bassists, but was also used by blues and rock guitarists. More open and transparent-sounding than the Small Tweed Combo, but can deliver crunchy sounds.
A small amp with a single 10" speaker, used by countless blues and rock artists. It is quite punchy-sounding, and can deliver the clean and crunch tones that the Tweed combos are known for.

Classic American Combos

The Blackface, Brownface, and Silverface models are inspired by American combos of the mid 1960s. These tend to be loud and clean with tight lows and relatively restrained distortion. They are great for clean-toned rock, vintage R & B, surf music, twangy country, jazz, or any other style where strong note definition is essential.
Large Blackface Combo
Silverface Combo
Mini Blackface Combo
Small Brownface Combo
Blues Blaster Combo
16 Chapter 1 Amps and Pedals
DescriptionModel
A 4 x 10" combo with a sweet, well-balanced tone favored by rock, surf, and R & B players. Great for lush, reverb-drenched chords or strident solos.
A 2 x 12" combo with a loud, ultra-clean tone. Its percussive, articulate attack is great for funk, R & B, and intricate chord work. It can be crunchy when overdriven, but most players favor it for clean tones.
A 1 x 10" combo that is bright and open-sounding, with a surprising amount of low-end impact. It excels at clean tones with just a hint of overdrive.
A 1 x 12" combo that is smooth and rich-sounding, but retains a nice level of detail.
A 1 x 15" combo that has a clear top end with a tight, defined low end. This model is favored by blues and rock players.
Tip: While these amps tend toward a clean and tight sound, you can use a Pedalboard distortion stompbox to attain hard-edgedcrunch sounds witha biting treble and extended low-end definition. See Distortion Pedals and Pedalboard.

British Stacks

The British Stack models are based on the 50- and 100-watt amplifier heads that have largely defined the sound of heavy rock, especially when paired with their signature 4 x 12" cabinets. At medium gain settings, these amps are great for chunky chords and riffs. Raising the gain yields lyrical solo tones and powerful rhythm guitar parts. Complex peaks and dips across the tonal spectrum keep the tones clear and appealing, even when heavy distortion is used.
DescriptionModel
Vintage British Stack
Modern British Stack
Brown Stack
British Blues Combo
Captures the sound of a late 1960s 50-watt amp famed for its powerful, smooth distortion. Notes retain clarity, even at maximum gain. After four decades this remains a definitive rock tone.
1980s and 1990s descendants of the Vintage British amplifier head, which were optimized for hard rock and metal styles of the time. The tones are deeper on the bottom, brighter on top, and more “scooped” in the middle than the Vintage British amp.
Unique tones can be coaxed from a British head by running it at lower voltages than its designers intended. The resulting “brown” sound—often more distorted and loose than the standard tone—can add interesting thickness to a guitar sound.
This 2 x 12" combo has a loud, aggressive tone that is cleaner than the British heads, yet delivers fat distortion tones at high-gain settings.
Tip: You’ll rarely go wrong combining a British head, a 4 x 12" cabinet, and a great riff at high levels. But don’t hesitate to break that mold. These heads can sound stunning through small cabinets, or at clean, low-gain settings. If the British Blues Combo is too clean for your needs, combine it with Pedalboard’s Hi Drive stompbox for an aggressive blues tone, or the Candy Fuzz stompbox for an explosive rock tone. See Distortion Pedals and Pedalboard.

British Combos

The British Combos capture the brash, treble-rich sound that will forever be associated with 1960s British rock and pop. The sonic signature of these amps is characterized by their high-end response, yet they are rarely harsh-sounding due to a sweet distortion and smooth natural compression.
DescriptionModel
British Combo
A 2 x 12" combo based on the early 1960s amps that powered the British Invasion. Perfect for chiming chords and stabbing solos.
17Chapter 1 Amps and Pedals
DescriptionModel
Small British Combo
Boutique British Combo
A 1 x 12" combo with half the power of the British Combo, this amp offers a slightly darker, less open tone.
A 2 x 12" combo that is a modern take on the original 1960s sound. The tone is thicker, with stronger lows and milder highs than the other British Combos.
Tip: Using high Treble and Presence knob settings that might become strident on other amp types can sound great with the British Combos.

British Alternatives

The late 1960s amplifier heads and combos that inspired the Sunshine models are loud and aggressive, with full-bodied mid frequencies. These amps are not just for single note solos and power chords, as they can sound great with big, open chords—one reason why they were embraced by the “Brit-pop” bands of the 1990s. The Stadium amps are famed for their ability to play ultra-loud without dissolving into mushy distortion. They retain crisp treble and superb note definition, even at maximum gain settings.
DescriptionModel
Sunshine Stack
Small Sunshine Combo
Stadium Stack
Stadium Combo
A robust-sounding head paired with a 4 x 12" cabinet. It’s a great choice for powerful pop-rock chords.
A 1 x 12" combo based on a modern amp known for a “big amp” sound. It is brighter than the Sunshine Stack head, with a touch of 1960s British Combo flavor.
A classic head and 4 x 12" cabinet configuration popular with 1970s arena rock bands. Its tones are cleaner than other Amp Designer 4 x 12" stacks, while still retaining body and impact. A good choice if you need power and clarity.
A 2 x 12" combo based on a modern amp. The tone is a little smoother and rounder than that of the Stadium Stack.
Tip: The tone of the Sunshine Stack can seem dark at times, but a high Treble knob setting opens up the sound. While the Small Sunshine Combo sounds great with its default 1 x 12" cabinet, it also shines through a 4 x 12" cabinet. The Stadium amps can be slow to distort, so most famous users have paired them with aggressive fuzz pedals. Try combining it with Pedalboard’s Candy Fuzz or Fuzz Machine stompboxes. See Distortion
Pedals and Pedalboard.

Metal Stacks

The Metal Stack models are inspired by the powerful, ultra-high gain amplifier heads that put the “chunk” into modern hard rock and metal music. All are paired with 4 x 12" cabinets. Their signature tones range from heavy distortion to extremely heavy distortion. If you want powerful lows, razor-edged highs, and serious sustain, these are the models you should look to first.
18 Chapter 1 Amps and Pedals
DescriptionModel
Modern American Stack
High Octane Stack
Turbo Stack
A powerful, ultra-high gain amp that is ideal for heavy rock and metal. Use the Mids knob to set an ideal amount of scoop or boost.
Although a powerful, high-gain amp, this model offers a smooth transition between gain settings andexcellent natural compression. It is a great choice for fast soloing and for two- and three-note chords.
An aggressive-soundingamp with spiky highs and noisy harmonics, especially at high gain settings. Try the Turbo Stack when you need to slice through a mix.
Tip: Combining the Turbo Stack with distortion and fuzz pedals may actually diminish the amp’s edge. A dry sound is often the best choice for high-impact riffs.

Additional Combos

The combos and utility models in this category are versatile amps that can be used for a wide variety of musical styles.
DescriptionModel
Studio Combo
Boutique Retro Combo
Pawnshop Combo
Transparent Preamp
A 1 x 12" combo based on boutique combos of the 1980s and 1990s that use multiple gain stages to generate smooth, sustain-heavy distortion without sacrificing bold, bright, clean sounds.
A 2 x 12" combo inspired by high-end modern amps that combine the sounds of several great 1960s combos. It excels at shimmering clean tones and crunch tones, making it a good choice when you want an old-fashioned flavor, but with the crisp highs and defined lows of a modern amplifier.
A 1 x 8" combo based on the inexpensive amps sold in American department stores in the 1960s. Despite their limited features and budget workmanship, these amps are the secret behind the sound of many rock, blues, and punk players. The clean sounds are warm, and distorted sounds are thick and satisfying, despite the small speaker.
As the name suggests, a preamp stage with no coloration. You should note that the Transparent Preamp is activated in the Amp pop-up menu, not in the Model pop-up menu.
Tip: Try pairing the Studio Combo amp with one of the 4 x 12" cabinets for a heavier sound. The Boutique Retro Amp has very sensitive tone controls, providing countless tonal shadings. Even extreme settings can yield great results. Combine the Pawnshop Combo amp with Pedalboard’s Hi Drive or Candy Fuzz stompboxes to emulate hard rock tones of the late 1960s. See Distortion Pedals and Pedalboard.
19Chapter 1 Amps and Pedals

Building a Customized Amp Designer Combo

Model pop-up menu
Cabinet pop-up menu
Amp pop-up menu
Mic pop-up menu
EQ pop-up menu
You can use one of the default models or you can create your own hybrid of different amplifiers, cabinets, and so on, using the Amp, Cabinet, and Mic pop-up menus, located on the black bar at the bottom of the interface. The EQ pop-up menu is accessed by clicking the word EQ or Custom EQ toward the left of the knobs section.
Note: If you create your own hybrid amp combo, you can use the Settings menu to save it as a setting file, which also includes any parameter changes you may have made.
Building an Amp Designer model is described in the following sections:
Choosing an Amp Designer Amplifier
Choosing an Amp Designer Cabinet
Using Amp Designer’s Equalizer
Setting Amp Designer Microphone Parameters

Choosing an Amp Designer Amplifier

You can choose an amplifier model from the Amp pop-up menu on the black bar at the bottom of the Amp Designer interface. See the following sections for details on the
20 Chapter 1 Amps and Pedals
characteristics of each amplifier in these categories:
Tweed Combos
Classic American Combos
British Stacks
British Combos
British Alternatives
Metal Stacks
Additional Combos

Choosing an Amp Designer Cabinet

Cabinets have a huge impact on the character of a guitar sound (see Amp Designer
Cabinet Reference Table). While certain amplifier and cabinet pairings have been popular
for decades, departing from them is an effective way to create fresh-sounding tones. For example, most players automatically associate British heads with 4 x 12" cabinets. Amp Designer allows you to drive a small speaker with a powerful head, or to pair a tiny amp with a 4 x 12" cabinet.
There’s nothing wrong with trying random combinations. But if you consider the variables that determine a cabinet’s sound, you’ll be able to make educated guesses about non-traditional amplifier and cabinet combinations. Some factors to consider:
Combos or Stacks
Combo amps include both an amplifier and speakers in a single enclosure. These usually have an open back, so the sound resonates in multiple directions. The resulting sound is “open”—with bright, airy highs and a general feeling of spaciousness. Amplifier “stacks” consist of an amplifier head, with the speakers in a separate cabinet. These cabinets generally have a closed back, and project the sound forward in a tight, focused “beam.” They tend to sound more powerful than open-back cabinets, and typically have a tighter low-end response at the expense of some high-end transparency.
Old or New Speakers
Amp Designer models that are based on vintage cabinets capture the character of aged speakers. These may be a bit looser and duller-sounding than new speakers, but many players prefer them for their smoothness and musicality. Sounds based on new cabinets tend to have more snap and bite.
Large Speakers or Small Speakers
A larger speaker doesn’t guarantee a larger sound. In fact, the most popular bass guitar cabinet of all time uses only small 8" speakers. Don’t be surprised if you get a deeper, richer tone from a 10" speaker than from a large 4 x 12" cabinet. Try several sizes and choose the one that works best for your music.
Single Speakers or Multiple Speakers
Guitarists sometimes use cabinets with multiple speakers, and not only for the larger sound they tend to provide. Phase cancellations occur between the speakers, adding texture and interest to the tone. Much of the “classic rock” sound, for example, has to do with the tonal peaks and dips caused by this interaction between the speakers in a 4 x 12" cabinet.
21Chapter 1 Amps and Pedals

Amp Designer Cabinet Reference Table

You can choose a cabinet model from the Cabinet pop-up menu on the black bar at the bottom of Amp Designer’s interface. The table below covers the properties of each cabinet model available in Amp Designer.
DescriptionCabinet
Tweed 1 x 12
Tweed 4 x 10
Tweed 1 x 10
Blackface 4 x 10
Silverface 2 x 12
Blackface 1 x 10
Brownface 1 x 12
Brownface 1 x 15
Vintage British 4 x 12
Modern British 4 x 12
Brown 4 x 12
British Blues 2 x 12
Modern American 4 x 12
Studio 1 x 12
British 1 x 12
Boutique British 2 x 12
A 12" open-back cabinet from the 1950s with a warm and smooth tone.
A 4 x 10" open-back cabinet that was originally conceived for bassists, but guitarists love its sparkling presence. An authentic late 1950s sound.
A single 10" open-back combo amp cabinet from the 1950s with a smooth sound.
Classic open-back cabinet with four 10" speakers. Its tone is deeper and darker than the Tweed 4 x 10.
An open-back model from the 1960s that provides great low-end punch.
An open-back 1960s cabinet with glistening highs and surprising low-mid body.
A beautifully balanced 1960s open-back cabinet. It is smooth and rich-sounding, but with nice transparency.
This early 1960s open-back cabinet houses the largest speaker emulated by Amp Designer. Its highs are clear and glassy, and its lows are tight and focused.
This late 1960s closed-backcabinet is synonymous with classic rock. The tone is big and thick, yet also bright and lively, thanks to the complex phase cancellations between the four 30-watt speakers.
A closed-back 4 x 12" cabinet that is brighter, and has a better low-end than the Vintage British 4 x 12, with less mid-range emphasis.
A closed-back 4 x 12" cabinet with a great bottom end and complex mid-range.
A bright-sounding open-back cabinet with solid lows, and highs that maintain their edge even at high gain settings.
A closed-back 4 x 12" cabinet that has a full sound. The low-mids are denser than the British 4 x 12" cabinets.
A compact-sounding open-back cabinet with full mids and shimmering highs.
A mid 1960s open-back cabinet with an open, smooth tone.British 2 x 12
A small open-back cabinet with crisp highs and nice low-mid transparency.
A 2 x 12" cabinet based on the British 2 x 12. It has a richer mid-range and is more assertive in the treble range.
22 Chapter 1 Amps and Pedals
DescriptionCabinet
A 4 x 12" closed-back cabinet with a thick, rich mid-range.Sunshine 4 x 12
Sunshine 1 x 12
Stadium 4 x 12
Stadium 2 x 12
Boutique Retro 2 x 12
High Octane 4 x 12
Turbo 4 x 12
A single 12" open-back combo amp cabinet with a bright, lively sound that has sweet highs, and transparent mids.
A tight, bright, closed-back British cabinet with bold upper-mid peaks.
A nicely balanced modern British open-back cabinet. Tonally, it is a compromise between the fatness of the Blackface 4 x 10 and the brilliance of the British 2 x 12.
A 2 x 12" cabinet based on the British 2 x 12. It has a rich, open mid-range and is more assertive in the treble range.
A modern, closed-back European cabinet with strong lows and highs and scooped mids appropriate for metal and heavy rock.
A modern, closed-back European cabinet with strong lows, very strong highs, and deeply scooped mids appropriate for metal and heavy rock.
Single 8" speaker cabinet that has excellent low-end punch.Pawnshop 1 x 8
This option bypasses the speaker emulation section.Direct

Using Amp Designer’s Equalizer

Hardware amplifier tone controls vary between models and manufacturers. There’s a good chance, for example, that the treble knobs on two different models target different frequencies, or provide different levels of cut or boost. Some equalizer (EQ) sections amplify the guitar signal more than others, affecting the way the amp distorts.
Amp Designer provides multiple EQ types to mirror these variations in hardware amplifiers. No matter which EQ type you choose, you’ll see an identical set of controls: Bass, Mids, and Treble. Switching between EQ types can result in these controls behaving very differently.
Selecting an EQ type other than the one traditionally associated with a certain amplifier typically results in significant tonal changes, although these may not necessarily be for the better. As with hardware amplifiers, Amp Designer’s EQs are calibrated to perform well with particular amplifier sounds. Choosing other EQ types can sometimes produce a thin, or unpleasantly distorted tone. See Amp Designer Equalizer Type Reference Table.
23Chapter 1 Amps and Pedals
Despite these less pleasant-sounding possibilities, you should experiment with different
Bass, Mids, and Treble knobs
EQ pop-up menu
amplifier and EQ combinations because many will sound great together.
The EQ parameters include the EQ pop-up menu and the Bass, Mids, and Treble knobs. These parameters are found toward the left-end of the knobs section.
EQ pop-up menu: Click the word EQ or CUSTOM EQ above the Bass, Mids, and Treble knobs to open the EQ pop-up menu, which contains the following EQ models: British Bright, Vintage, U.S. Classic, Modern, and Boutique. Each EQ model has unique tonal qualities that affect the way the Bass, Mids, and Treble knobs respond. See Amp Designer
Equalizer Type Reference Table.
Bass, Mids, and Treble knobs: Adjust the frequency ranges of the EQ models, similar to the tone knobs on a hardware guitar amplifier. The behavior and response of these knobs changes when different EQ models are chosen.

Amp Designer Equalizer Type Reference Table

You can choose an Equalizer type by clicking the word EQ or CUSTOM EQ above the Bass, Mids, and Treble knobs in the knobs section. The table below covers the properties of each EQ type available in Amp Designer.
British Bright
24 Chapter 1 Amps and Pedals
Vintage
U.S. Classic
DescriptionEQ type
Inspired by the EQ of British combo amps of the 1960s. It is loud and aggressive, with even bolder highs than the Vintage EQ. This EQ is useful if you want more treble definition without an overly clean sound.
Emulates the EQ response of American Tweed-style amps and the vintage British stack amps that used a very similar circuit. It is loud and somewhat distortion-prone. This EQ is useful if you want to roughen the sound.
Derived from the EQ circuit of the American Blackface-style amps. The tone is of higher fidelity than the Vintage EQ, with tighter lows and crisper highs. This EQ is useful if you want to brighten your tone and reduce distortion.
DescriptionEQ type
Gain
Presence Master
Modern
Boutique
Based on a digital EQ unit popular in the 1980s and 1990s. This EQ is useful for sculpting the hyped highs, booming lows, and scooped mids associated with the era’s rock and metal music styles.
Replicates the tone section of a “retro modern” boutique amp. It excels at precise EQ adjustments, though its tone may be cleaner than desired when used with vintage amplifiers. This EQ is a good choice if you want a cleaner, brighter sound.

Using Amp Designer’s Gain, Presence, and Master Controls

The amp parameters include controls for the input gain, presence, and master output. The Gain knob is found to the left in the knobs section and the Presence and Master knobs are to the right.
Gain knob: Sets the amount of pre-amplification applied to the input signal. This control affects various amp models differently. For example, when you are using the British Amp, the maximum gain setting produces a powerful crunch sound. When you are using the Vintage British Head or Modern British Head, the same gain setting produces heavy distortion, suitable for lead solos.
Presence knob: Adjusts the high-frequency range—above the range of the Treble control. The Presence parameter affects only the output (Master) stage.
Master knob: Sets the output volume of the amplifier going to the cabinet. For tube amplifiers, increasing the Master level typically produces a somewhat compressed and saturated sound, resulting in a more distorted and powerful—that is, louder—signal. High Master settings can produce an extremely loud output that can damage your speakers or hearing, so ramp this up slowly. The final output level of Amp Designer is set with the Output slider at the lower-right edge of the interface. See Setting Amp
Designer’s Output Level.
25Chapter 1 Amps and Pedals

Getting to Know Amp Designer’s Effects Parameters

The effects parameters include Tremolo, Vibrato, and Reverb, which emulate the processors found on many amplifiers. these controls are found in the center of the knobs section.
You can use the switch toward the right to select either Tremolo (TREM), which modulates the amplitude or volume of the sound, or Vibrato (VIB), which modulates the pitch.
Reverb, which is controlled by a switch in the middle, can be added to either of these effects, or used independently.
Note: The Effects section is placed before the Presence and Master controls in the signal flow, and receives the pre-amplified, pre-Master signal.
Reverb, Tremolo, and Vibrato are described in the following sections:
Using Amp Designer’s Reverb Effect
Using Amp Designer’s Tremolo and Vibrato Effects

Using Amp Designer’s Reverb Effect

Reverb is always available in Amp Designer, even when using a model that is based on an amplifier that provides no reverb function. Reverb is controlled by an On/Off switch and a Level knob in the middle, above which is the Reverb pop-up menu. Reverb can be added to either the Tremolo or Vibrato effect, or used independently.
On/Off switch: Enables or disables the reverb effect.
Reverb pop-up menu: Click the word Reverb to choose one of the following reverb types from the pop-up menu: Vintage Spring, Simple Spring, Mellow Spring, Bright Spring, Dark Spring, Resonant Spring, Boutique Spring, Sweet Reverb, Rich Reverb, and Warm Reverb. See Amp Designer Reverb Type Reference Table for information on these reverb types.
26 Chapter 1 Amps and Pedals
Level knob: Sets the amount of reverb applied to the pre-amplified signal.

Amp Designer Reverb Type Reference Table

You can choose a reverb type by clicking the Reverb label in the center of the Amp section. The table below covers the properties of each reverb type available in Amp Designer.
DescriptionReverb type
Vintage Spring
Bright Spring
Resonant Spring
Boutique Spring
This bright, splashy sound has largely defined combo amp reverb since the early 1960s.
A darker, subtler spring sound.Simple Spring
An even darker, somewhat low-fidelity spring sound.Mellow Spring
Has some of the brilliance of Vintage Spring, but with less surf-style splash.
A moody-sounding spring. More restrained than Mellow Spring.Dark Spring
Another 1960s-style spring with a strong, slightly distorted mid-range emphasis.
A modernized version of the classic Vintage Spring with a richer tone in the bass and mids.
A smooth modern reverb with rich lows and restrained highs.Sweet Reverb
A bold, well-balanced modern reverb.Rich Reverb
A lush modern reverb with rich low-mids and understated highs.Warm Reverb

Using Amp Designer’s Tremolo and Vibrato Effects

Tremolo and vibrato are controlled by several switches and two knobs in the Effects section found toward the right of the knobs section. Tremolo modulates the amplitude or volume of the sound, and vibrato modulates the pitch.
On/Off switch: Enables or disables the tremolo or vibrato effect.
Trem/Vib switch: Choose either tremolo or vibrato.
Depth knob: Sets the intensity of the modulation (tremolo or vibrato).
Speed knob: Sets the speed of the modulation in Hertz. Lower settings produce a smooth, floating sound. Higher settings produce a rotor-like effect.
27Chapter 1 Amps and Pedals
Sync/Freeswitch: When the switch is set to Sync, the modulation speed is synchronized
Move your mouse above the Mic pop-up menu to display the speaker-adjustment graphic.
with the host application tempo. The Speed knob lets you select different bar, beat, and musical note values (1/8, 1/16, and so on, including triplet and dotted-note values). When the switch is set to Free, the modulation speed can be set to any available value with the Speed knob.

Setting Amp Designer Microphone Parameters

Amp Designer offers a choice between three different virtual microphones. As with every other component in the tone chain, different selections yield very different results. After choosing a cabinet, you can set the type of microphone you want to be emulated, and where the microphone is placed in relation to the cabinet. The Mic pop-up menu is available near the right end of the black bar at the bottom, and the speaker-adjustment graphic appears when you move your mouse to the area above the Mic pop-up menu.
Note: The parameters described in this section are accessible only in the full Amp Designer interface. If you are in the small interface, click the disclosure triangle to the right of the Output field at the bottom-right edge of the interface to switch back to the full interface.
Cabinet and speaker-adjustment graphic: By default, the microphone is placed in the center of the speaker cone (on-axis). This placement produces a fuller, more powerful sound, suitable for blues or jazz guitar tones. If you place the microphone on the rim of the speaker (off-axis), you obtain a brighter, thinner tone, making it suitable for cutting rock or R & B guitar parts. Moving the microphone closer to the speaker emphasizes bass response.
The microphone position is shown on the cabinet and indicated by the white dot in the speaker-adjustment graphic. Drag the whitedot to change the microphone position and distance, relative to the cabinet. Placement is limited to near-field positioning.
28 Chapter 1 Amps and Pedals
Mic pop-up menu: You can choose one of the Microphone models from the pop-up menu:
Condenser: Emulates the sound of a high-endGerman studio condenser microphone.
The sound of condenser microphones is fine, transparent, and well-balanced.
Dynamic: Emulates the sound of popular American dynamic cardioid microphones.
This microphone type sounds brighter and more cutting than the Condenser model. The mid-rangeis boosted, with lower-mid frequencies being less pronounced, making it a good choice for miking rock guitar tones. It is especially useful if you want your guitar part to cut through other tracks in a mix.
Ribbon: Emulates the sound of a ribbon microphone. A ribbon microphone is a type
of dynamic microphone that captures a sound often described as bright or brittle, yet still warm. It is useful for rock, crunch, and clean tones.
Tip: Combining multiple microphone types can produce an interesting sound. Duplicate the guitar track, and insert Amp Designer on both tracks. Select different microphones in each Amp Designer instance while retaining identical settings for all other parameters, and set track signal levels to taste.

Setting Amp Designer’s Output Level

The Output slider (or the Output field, in the small interface) is found at the lower-right corner of the Amp Designer interface. It serves as the final level control for Amp Designer and can be thought of as a “behind the speaker” volume control that sets the level of the output that is fed to the ensuing Insert slots in the channel strip, or directly to the channel strip output.
Note: This parameter is different from the Master control, which serves the dual purpose of sound design as well as controlling the level of the Amp section.
29Chapter 1 Amps and Pedals

Bass Amp

Bass Amp simulates the sound of several famous bass amplifiers. You can route bass guitar and other signals directly through the Bass Amp, reproducing the sound of your musical part played through a number of high-quality bass guitar amplification systems.
Bass Amp offers the following parameters.
Model pop-up menu: Includes the following amplifier models:
American Basic: 1970s-era American bass amp, equipped with eight 10" speakers.
Well-suited for blues and rock recordings.
American Deep: Based on the American Basic amp, but with strong lower-mid
frequency (from 500 Hz on) emphasis. Well-suited for reggae and pop recordings.
American Scoop: Based on the American Basic amp, but combines the frequency
characteristics of the American Deep and American Bright, with both low-mid (from 500 Hz) and upper-mid (from 4.5 kHz) frequencies emphasized. Well-suited for funk and fusion recordings.
American Bright: Based on the American Basic amp, this model emphasizes the
upper-mid frequencies (from 4.5 kHz upward).
New American Basic: 1980s-era American bass amp, well-suited for blues and rock
recordings.
New American Bright: Based on the New American Basic amp, this model strongly
emphasizes the frequency range above 2 kHz. Well-suited for rock and heavy metal.
30 Chapter 1 Amps and Pedals
Top Class DI Warm: Famous DI box simulation, well-suited for reggae and pop
recordings. Mid frequencies, in the range between 500 and 5000 Hz, are de-emphasized.
TopClass DI Deep: Based on the Top Class DI Warm, this model is well-suited for funk
and fusion. The mid frequency range is strongest around 700 Hz.
Top Class DI Mid: Based on the Top Class DI Warm, this model features an almost
linear frequency range, with no frequencies emphasized. It is suitable for blues, rock, and jazz recordings.
Pre Gain slider: Sets the pre-amplification level of the input signal.
Bass, Mid, and Treble sliders: Adjusts the bass, mid, and treble levels.
Mid Freq slider: Sets the center frequency of the mid band (between 200 Hz and 3000 Hz).
Output Level slider: Sets the final output level for Bass Amp.

Guitar Amp Pro

Guitar Amp Pro can simulate the sound of popular guitar amplifiers and the speakers used with them. You can process guitar signals directly, which enables you to reproduce the sound of your guitar through a number of high-quality guitar amplification systems.
Guitar Amp Pro can also be used for experimental sound design and processing. You can freely use it with other instruments, applying the sonic character of a guitar amp to a trumpet or vocal part, for example.
The amplifier, speaker, and EQ models emulated by Guitar Amp Pro can be combined in a number of ways to radically or subtly alter the tone. Virtual microphones are used to pick up the signal of the emulated amplifier and cabinet. You can choose from two different microphone types, and you can reposition them. Guitar Amp Pro also emulates classic guitar amplifier effects, including reverb, vibrato, and tremolo.
31Chapter 1 Amps and Pedals
The Guitar Amp Pro window is organized into sections according to different kinds of
Effects section
Amp section
Microphone Position section Microphone Type section
parameters.
Amp section: The model parameters at the top are used to choose the type of amp, EQ model, and speaker. See Building Your Guitar Amp Pro Model.
Farther down in the Amp section, the knobs in the V-shaped formation are used to set tone, gain, and level. See Using Guitar Amp Pro’s Gain, Tone, Presence, and Master
Controls.
Effects section: Provides parameters to control the built-in tremolo, vibrato, and reverb effects. See Using Guitar Amp Pro’s Reverb Effect and Using Guitar Amp Pro’s Tremolo
and Vibrato Effects.
Microphone Position and Type sections: These sections enable you to set the position and type of the microphone. See Setting Guitar Amp Pro Microphone Parameters.

Building Your Guitar Amp Pro Model

An amplifier “model” consists of an amplifier, speaker cabinet, EQ type, and microphone type. You can create your own hybrids of different amplifiers, cabinets, and so on—using the pop-up menus at the top center of the interface. You choose the microphone position and type in the yellow areas to the left and right.
You can use the Settings menu to save your new hybrid amp combos as setting files, which also include any parameter changes you may have made.
How to build your amplifier model is described in the following sections:
Choosing a Guitar Amp Pro Amplifier
Choosing a Guitar Amp Pro Speaker Cabinet
32 Chapter 1 Amps and Pedals
Choosing a Guitar Amp Pro Equalizer
Setting Guitar Amp Pro Microphone Parameters

Choosing a Guitar Amp Pro Amplifier

You can choose an amplifier model from the Amp pop-up menu near the top of the interface.
UK Combo 30W: Neutral-sounding amp, well-suited for clean or crunchy rhythm parts.
UK Top 50W: Quite aggressive in the high frequency range, well-suited for classical rock sounds.
US Combo 40W: Clean sounding amp model, well-suited for funk and jazz sounds.
US Hot Combo 40W: Emphasizes the high mid-frequency range, making this model ideal for solo sounds.
US Hot Top 100W: This amp produces very fat sounds, even at low Master settings, that result in broad sounds with a lot of “oomph.”
Custom 50W: With the Presence parameter set to 0, this amp model is well-suited for smooth fusion lead sounds.
British Clean (GarageBand): Simulates the classic British Class A combos used continuously since the 1960s for rock music, without any significant modification. This model is ideally suited for clean or crunchy rhythm parts.
British Gain(GarageBand): Emulates the sound of a British tube head and is synonymous with rocking, powerful rhythm parts and lead guitars with a rich sustain.
American Clean(GarageBand): Emulates the traditional full tube combos used for clean and crunchy sounds.
American Gain (GarageBand): Emulates a modern Hi-Gain head, making it suitable for distorted rhythm and lead parts.
Clean Tube Amp: Emulates a tube amp model with very low gain (distortion only when using very high input levels or Gain/Master settings).

Choosing a Guitar Amp Pro Speaker Cabinet

The speaker cabinet can have a huge bearing on the type of tones you can extract from your chosen amplifier. The speaker parameters are found near the top of the interface.
Speaker pop-up menu: You can choose one of the 15 speaker models:
UK 1 x 12 open back: Classic open enclosure with one 12" speaker, neutral,
well-balanced, multifunctional.
UK 2 x 12 open back: Classic open enclosure with two 12" speakers, neutral,
well-balanced, multifunctional.
UK 2 x 12 closed: Loads of resonance in the low frequency range, therefore well-suited
for Combos: crunchy sounds are also possible with low Bass control settings.
33Chapter 1 Amps and Pedals
UK 4 x 12 closed slanted: when used in combination with off-center miking, you will
get an interesting mid frequency range; therefore, this model works well when combined with High Gain amps.
US 1 x 10 open back: Not much resonance in the low frequency range. Suitable for
use with blues harmonicas.
US 1 x 12 open back 1: Open enclosure of an American lead combo with a single 12"
speaker.
US 1 x 12 open back 2: Open enclosure of an American clean/crunch combo with a
single 12" speaker.
US 1 x 12 open back 3: Open enclosure of another American clean/crunch combo
with a single 12" speaker.
US broad range: Simulation of a classic electric piano speaker.
Analog simulation: Internal speaker simulation of a well-known British tube
preamplifier.
UK 1 x 12 (GarageBand): A British Class A tube open back with a single 12" speaker.
UK 4 x 12 (GarageBand): Classic closed enclosure with four 12" speakers (black series),
suitable for rock.
US 1 x 12 open back (GarageBand): Open enclosure of an American lead combo with
a single 12" speaker.
US 1 x 12 bass reflex (GarageBand): Closed bass reflex cabinet with a single 12" speaker.
DI Box: This option allows you to bypass the speaker simulation section.
Amp-Speaker Linkbutton: Located between the Amp and Speaker pop-up menus, links these pop-up menus so that when you change the amp model, the speaker associated with that amp is loaded automatically.

Choosing a Guitar Amp Pro Equalizer

The EQ pop-up menu and the Amp-EQ Link button are near the top of the interface.
EQ pop-up menu: Contains the following EQ models: British1, British2, American, and Modern. Each EQ model has unique tonal qualities that affect the way the Bass, Mids, and Treble knobs in the Amp section respond.
Amp-EQ Link button: Located between the Amp and EQ pop-up menus, links these pop-up menus so that when you change the amp model, the EQ model associated with that amp is loaded automatically.
Each amp model has a speaker and EQ model associated with it. The default combinations of amp, speaker, and EQ settings recreate a well-known guitar sound. You are, of course, free to combine any speaker or EQ model with any amp by turning off the two Link buttons.
34 Chapter 1 Amps and Pedals

Using Guitar Amp Pro’s Gain, Tone, Presence, and Master Controls

The Gain, Bass, Mids, Treble, Presence, and Master knobs run from left to right in the V-shaped formation in the upper half of the interface.
Gain knob: Sets the amount of pre-amplification applied to the input signal. This control has different effects, depending on which Amp model is chosen. For example, when you are using the British Clean amp model, the maximum Gain setting produces a powerful crunch sound. If you use the British Gain or Modern Gain amps, the same Gain setting produces heavy distortion, suitable for lead solos.
Bass, Mids, and Treble knobs: Adjust the frequency range levels of the EQ models, similar to the tone knobs on a hardware guitar amplifier.
Presence knob: Adjusts the high frequency range level. The Presence parameter affects only the output (Master) stage of Guitar Amp Pro.
Master knob: Sets the output volume of the amplifier—going to the speaker. For tube amplifiers, increasing the Master level typically produces a more compressed and saturated sound, resulting in a more distorted and powerful—that is, louder—signal. High Master settings can produce an extremely loud output that can damage your speakers or hearing, so ramp this up slowly. In Guitar Amp Pro, the Master parameter modifies the sonic character, and the final output level is set using the Output parameter at the bottom of the interface. See Setting the Guitar Amp Pro Output Level.

Getting to Know Guitar Amp Pro’s Effects Section

The effects parameters include Tremolo, Vibrato, and Reverb, which emulate the processors found on many amplifiers.
You can use the pop-up menu to choose either Tremolo, which modulates the amplitude or volume of the sound, or Vibrato, which modulates the pitch.
Reverb can be added to either of these effects, or used independently.
To use or adjust an effect, you must first enable it by clicking the corresponding On button to the left. The On button is red when active.
Note: The Effects section is placed before the Presence and Master controls in the signal flow, and receives the preamplified, pre-Master signal.
Tremolo, Vibrato, and Reverb are described in the following sections:
Using Guitar Amp Pro’s Tremolo and Vibrato Effects
Using Guitar Amp Pro’s Reverb Effect
35Chapter 1 Amps and Pedals

Using Guitar Amp Pro’s Tremolo and Vibrato Effects

Tremolo and vibrato are controlled by an On button, the FX pop-up menu, the Depth and Speed knobs, and the Sync button in the Effects section. Tremolo modulates the amplitude or volume of the sound, and vibrato modulates the pitch.
FX pop-up menu: You can choose either Tremolo or Vibrato.
Depth knob: Sets the intensity of the modulation.
Speed knob: Sets the speed of the modulation in Hertz. Lower settings producea smooth and floating sound, while higher settings produce a rotor-like effect.
Sync button: When the Sync button is turned on, the modulation speed is synchronized to the project tempo. You can adjust the Speed knob to select bar, beat, and musical note values (including triplet and dotted notes). When the Sync button is turned off, the modulation speed can be set to any available value with the Speed knob.

Using Guitar Amp Pro’s Reverb Effect

Reverb is controlled by an On button, the Reverb pop-up menu, and a Level knob in the Reverb section near the bottom. Reverb can be added to either the Tremolo or Vibrato effect, or used independently.
Reverb pop-up menu: Choose one of the three types of spring reverb.
Level knob: Sets the amount of reverb applied to the pre-amplified amp signal.

Setting Guitar Amp Pro Microphone Parameters

After choosing a speaker cabinet from the Speaker menu, you can set the type of microphone you want to be emulated, and where the microphone is placed in relation to the speaker. The Microphone Position parameters are available in the yellow area to the left, and the Microphone Type parameters in the yellow area to the right.
Microphone Position Parameters
Centered button: Places the microphone in the center of the speaker cone, also called on-axis. This placement produces a fuller, more powerful sound, suitable for blues or
jazz guitar tones.
Off-Center button: Places the microphone on the edge of the speaker, also referred to as off-axis. This placement produces a tone that is brighter and sharper, but also thinner—suitable for cutting rock or R & B guitar parts.
When you select either button, the graphic speaker display reflects your choice.
Microphone Type Parameters
Condenser button: Emulates the sound of a studio condenser microphone. The sound of condenser microphones is fine, transparent, and well-balanced.
36 Chapter 1 Amps and Pedals
Dynamic button: Emulates the sound of a dynamic cardioid microphone. This microphone type sounds brighter and more cutting than the Condenser model. At the same time, the lower-mid frequency range is less pronounced, making this model more suitable for miking rock guitar tones.
Tip: Combining both microphone types can sound quite interesting. Duplicate the guitar track, and insert Guitar Amp Pro as an insert effect on both tracks. Select different microphone types in each Guitar Amp Pro instance, while retaining identical settings for all other parameters, and mix the track signal levels. You can, of course, choose to vary any other parameters.

Setting the Guitar Amp Pro Output Level

The Output slider is found at the bottom, below the Effects section. It serves as the final level control for Guitar Amp Pro and can be thought of as a “behind the speaker” volume control that is used to set the level fed to the ensuing plug-in slots on the channel strip or to Output channel strips.
Note: This parameter is different from the Master control, which serves the dual purpose of sound design as well as controlling the level of the Amp section.

Pedalboard

The Pedalboard simulates the sound of a number of well-loved and famous “stompbox” pedal effects. You can process any audio signal with a combination of stompboxes.
You can add, remove, and reorder pedals. The signal flow runs from left to right in the Pedal area. The addition of two discrete busses, coupled with splitter and mixer units, enables you to experiment with sound design and precisely control the signal at any point in the signal chain.
37Chapter 1 Amps and Pedals
Macro Controls area
Pedal Browser
Pedal area
Routing area
All stompbox knobs, switches, and sliders can be automated. Eight Macro controls enable real time changes to any pedal parameter with a MIDI controller.
• The Pedal Browser shows all pedal effects and utilities. These can be dragged into the Pedal area as part of the signal chain. See Using Pedalboard’s Pedal Browser. This interface area is also used for the alternative import mode. See Using Pedalboard’s
Import Mode.
• The Pedal area is where you determine the order of effects and set effect parameters. You can add, replace, and remove stompboxes here. See Using Pedalboard’s Pedal
Area.
• The Routing area is used to control signal flow in the two effects busses (Bus A and Bus B) available in Pedalboard. See Using Pedalboard’s Routing Area.
• The Macro Controls area is used to assign eight MIDI controllers, which can be used to control any stompbox parameter in real time. See Using Pedalboard’s Macro Controls
Area.
• The effect and utility pedals are described in the following sections:
Distortion Pedals
Modulation Pedals
Delay Pedals
Filter Pedals
Dynamics Pedals
Utility Pedals
38 Chapter 1 Amps and Pedals

Using Pedalboard’s Pedal Browser

View pop-up menu
Import Mode button
Pedalboard offers dozens of pedal effects and utilities in the Pedal Browser on the right side of the interface. Each effect and utility is grouped into a category, such as distortion, modulation, and so on. For information about these types of stompboxes, see Distortion
Pedals, Modulation Pedals, Delay Pedals, Filter Pedals, Dynamics Pedals, and Utility Pedals.
To hide or show the Pedal Browser
Click the disclosure triangle in the lower-right corner of the Pedal area.
µ
To show only specific pedal groups in the Pedal Browser
Open the View pop-up menu and choose Distortion, Modulation, Delay, Filter, Dynamics,
µ
or Utility. The Pedal Browser shows only the stompboxes within the category you choose.
To show all the pedal groups, choose Show All from the View pop-up menu.
To add a stompbox to the Pedal area
Do one of the following:
Drag the effect that you want to insert from the Pedal Browser to the appropriate Pedal
µ
area position. This can be to the left, to the right, or in-between existing pedals.
Double-click an effect in the Pedal Browser to add it to the right of all existing stompboxes
µ
in the Pedal area.
39Chapter 1 Amps and Pedals
Note: Double-clicking a stompbox in the Pedal Browser when a stompbox is selected in the Pedal area will replace the selected pedal.

Using Pedalboard’s Import Mode

Pedalboard has a feature you can useto import parameter settingsfor each type ofpedal. In contrast to the plug-in window Settings menu, which you use to load a setting for the entire Pedalboard plug-in, this feature can be used to load a setting for a specific stompbox type.
To activate or deactivate import mode
Click the Import Mode button to show all pedals used in the most recent Pedalboard
µ
setting. When the Import Mode button is active,the Pedal Browser switches to analternate view mode that displays imported settings. When import mode is inactive, the normal Pedal Browser view is shown.
To import pedal settings into the Pedal Browser
1 Click the Import Mode button to activate import mode. Note thatthe View menu changes
to the Select Setting button.
Note: If this is your first attempt to import settings, a dialog opens where you can select a setting to import.
40 Chapter 1 Amps and Pedals
2 Click the Select Setting button and select a setting, then click Open. Dependent on the
chosen setting, one or more stompboxes appear in the Pedal Browser. The name of the imported setting is shown at the bottom of the Pedal Browser.
To add an imported pedal to the Pedal area
Do one of the following:
Drag the stompbox that you want to add from the Pedal Browser to the appropriate
µ
Pedal area position. This can be to the left, to the right, or in-between existing pedals.
Ensure that no pedal is selected in the Pedal area, then double-click a stompbox in the
µ
Pedal Browser to add it to the right of all existing effects in the Pedal area.
Note: The parameter settings of pedals added in import mode are also imported.
To replace a pedal setting in the Pedal area with an imported pedal setting
1 Click the pedal you want to replace in the Pedal area. It becomes highlighted with a blue
outline.
2 Click the stompbox in the Pedal Browser to replace the selected pedal (or pedal setting)
in the Pedal area. The blue outlines of the selected pedal in the Pedal area and Pedal Browser blink on and off to indicate an imported setting. The setting name area at the bottom of the Pedal Browser displays “Click selected item again to revert.”
Note: If you want to make your replacement permanent, click the background in the Pedal Browser, or click the Import Mode button.
3 To restore the selected pedal’s previous setting, click the highlighted stompbox in the
Pedal Browser. The Import Mode button and the outline of the selected pedal (in the Pedal area) become solidly highlighted, indicating that the original setting has been restored.

Using Pedalboard’s Pedal Area

Pedalboard’s stompbox effect pedals not only resemble their physical counterparts; they are also used in much the same way—without the inconvenience of patch cords, power supplies, and screws or locking mechanisms. The Pedal area layout mirrors a traditional pedalboard, with signals running from left to right.
41Chapter 1 Amps and Pedals
To add a pedal to the Pedal area
Do one of the following:
Drag the stompbox that you want to insert from the Pedal Browser to the appropriate
µ
Pedal area position. This can be to the left, to the right, or in-between existing pedals.
Ensure that no pedal is selected in the Pedal area, then double-click a stompbox in the
µ
Pedal Browser to add it to the right of all existing effects in the Pedal area.
Note: You insert Mixer and Splitter utility pedals in a different way. See Using Pedalboard’s
Routing Area.
To change an effect pedal position in the Pedal area
Drag the stompbox to a new position, either to the right or the left. Automation and bus
µ
routings, if active, are moved with the effect pedal. For information about automation and bus routings, see Using Pedalboard’s Routing Area.
Note: There are two exceptions to the bus routing rule: If the dragged pedal is the only pedal between a Splitter and Mixer utility, both utility pedals are automatically removed. If the second Bus (“B”) is not active at the destination, the pedal is inserted into Bus A.
To change a Mixer utility position in the Pedal area
Drag the Mixer utility to a new position, either to the left or the right.
µ
When moved to the left: The “downmix” of Bus A and B will occur at the earlier insertion point. Relevant effect pedals are moved to the right and are inserted into Bus A.
When moved to the right: The “downmix” of Bus A and B will occur at the later insertion point. Relevant effect pedals are moved to the left and are inserted into Bus A.
Note: A Mixer pedal cannot be moved to a position directly after (or to the left of) a corresponding split point or Splitter utility.
To change a Splitter utility position in the Pedal area
Drag the Splitter utility to a new position, either to the right or the left.
µ
When moved to the left: The split between Bus A and B will occur at the earlier insertion point. Relevant effect pedals are moved to the right and are inserted into Bus A.
When moved to the right: The split between Bus A and B will occur at the later insertion point. Relevant effect pedals are moved to the left and are inserted into Bus A.
Note: A Splitter pedal cannot be moved to a position directly preceding (or to the right of) a corresponding Mixer utility.
To replace a pedal in the Pedal area
Do one of the following:
Drag the stompbox from the Pedal Browser directly over the pedal you want to replace
µ
in the Pedal area.
42 Chapter 1 Amps and Pedals
Click to select the stompbox you want to replace in the Pedal area, then double-click the
µ
appropriate pedal in the Pedal Browser.
Note: You can only replace “effect” pedals, not the Mixer or Splitter utilities. Bus routings, if active, are not changed when an effect pedal is replaced.
To remove a pedal from the Pedal area
Do one of the following:
Drag the pedal out of the Pedal area.
µ
Click the pedal to select it and press the Delete key.
µ

Using Pedalboard’s Routing Area

Pedalboard has two discrete signal buses—Bus A and Bus B—that are found in the Routing area above the Pedal area. These busses provide a great deal of flexibility when you are setting up signal processing chains. All stompboxes that you drag into the Pedal area are inserted into Bus A, by default.
Note: The Routing area appears when you move your pointer to a position immediately above the Pedal area, and it disappears when you move the pointer away. When you create a second bus routing, the Routing area remains open even when your pointer is not over it. You can close the Routing area by clicking the small latch button at the top, and then the Routing area will open or close automatically when you move your pointer over it.
To create a second bus routing
Do one of the following:
Move your pointer immediately above the Pedal area to open the Routing area, and click
µ
the name of a stompbox in the Routing area. The pedal name moves upward, and the chosen stompbox is routed to Bus B. Two gray lines appear in the Routing area, which represent Bus A and Bus B. A Mixer utility pedal is automatically added to the end of the signal chain.
Drag a Splitter utility pedal into the Pedal area when more than one pedal is inserted.
µ
This also inserts a Mixer at the end of the signal chain if one doesn’t already exist.
To remove the second bus routing
Do one of the following:
Remove the Mixer and Splitter utility pedals from the Pedal area.
µ
43Chapter 1 Amps and Pedals
Remove all stompboxes from the Pedal area. This automatically removes an existing Mixer
µ
utility.
To remove an effect from the second bus
Click the name of the pedal (or on either of the gray lines) in the Routing area.
µ
Note: The removal of all effects from Bus B does not remove the second bus. The Mixer utility pedal remains in the Pedal area, even when a single stompbox (effect) is in the Pedal area. This allows parallel routing of wet and dry signals. Only when all pedal effects are removed from the Pedal area is the Mixer utility (and second bus) removed.
To determine the split point between busses
When more than one bus is active, a number of dots appear along the “cables” (gray
µ
lines) in the Routing area. These represent the output (the socket) of the pedal to the lower left of the dot. Click the appropriate dotto determine where the split point—where the signal is routed between busses. A cable appears between the busses when you click a dot.
Note: You can not create a split point directly before, or after, the Mixer utility.
To switch between a Splitter utility and bus split point
Double-click a bus split point dot in the Routing area to replace it with a Splitter utility.
µ
The Splitter utility is shown in the Pedal area.
Double-click the Splitter label in the Routing area to replace the Splitter utility with a bus
µ
split point dot. The Splitter utility is removed from the Pedal area.
Notes on Splitter and Mixer Utility Use
Dragging a Splitter utility into the Pedal area automatically inserts a Mixer utility to the far right of all inserted pedals.
You cannot drag a Splitter utility to the far right of all inserted pedals, to directly after an inserted Splitter utility, to directly in front of an inserted Mixer utility, or to an empty space in the Pedal area.
Dragging a Mixer utility into the Pedal area automatically creates a split point at the earliest possible (the leftmost) point within the signal chain.
You cannot drag a Mixer utility to the first slot in the Pedal area, to between an inserted Splitter and Mixer utility combo, or directly to the right of an inserted Mixer utility.

Using Pedalboard’s Macro Controls Area

Pedalboard provides eight Macro Targets—A through H—which are found in the Macro Controls area below the Pedal area. These enable you to map any parameterof an inserted stompbox as a Macro A–H target. You can save different mappings with each Pedalboard setting.
44 Chapter 1 Amps and Pedals
In Logic Express, you use a controller assignment or create a Workspace knob for “Macro A–H Value.” MIDI hardware switches, sliders, or knobs can then be used to control the mapped Pedalboard Macro A–H target parameters in real time. See the Logic Express User Manual for details.
Click the triangle at the bottom left to hide or show the Macro Controls area.
Macro A–H Target pop-up menus: Determine the parameter that you want to control with a MIDI controller.
Macro A–H Value sliders and fields: Set, and display, the current value for the parameter chosen in the corresponding Macro Target pop-up menu.
To assign a Macro A–H Target
Do one of the following:
Click any of the Macro A–H Target pop-up menus, and choose the parameter that you
µ
want to control.
Each stompbox parameter is shown in the following way: “Slot number—Pedal
Name—Parameter”. As examples: “Slot 1—Blue Echo—Time”, or “Slot 2—Roswell Ringer—Feedback”. The “slot” number refers to the pedal position, as they appear from
left to right in the Pedal area.
Choose the “-Auto assign-” item in any Macro A–H Target pop-up menu, then click the
µ
appropriate parameter in any inserted pedal.
Note: The chosen parameter is displayed in the Macro A–H Target pop-up menu.

Distortion Pedals

This section describes the distortion effects pedals.
DescriptionStompbox
Candy Fuzz
Double Dragon
A bright, “nasty” distortion effect. Drive controls the input signal gain. Level sets the effect volume.
A deluxe distortion effect. It offers independent level controls for input (Input) and output (Level). Drive controls the amount of saturation applied to the input signal. The Tone knob sets the cutoff frequency. The Squash knob sets the threshold for the internal compression circuit.Contour sets the amount ofnonlinear distortion applied to the signal. Mix sets the ratio between the source and distorted signals. The Bright/Fat switch changes between two fixed high shelving filter frequencies. Blue and red LEDs indicate each switch position, respectively.
45Chapter 1 Amps and Pedals
Fuzz Machine
Grinder
Happy Face Fuzz
Hi-Drive
Monster Fuzz
Octafuzz
Rawk! Distortion
Vintage Drive
DescriptionStompbox
An American “fuzz” distortion effect. Fuzz controls the input gain. Overall output gain is set with Level. The Tone knob increases treble, while simultaneously rolling-off low frequencies, as you move it to higher values.
Grinder is a lo-fi, dirty “metal” distortion. Grind sets the amount of drive applied to the input signal. Tone is controlled with the Filter knob, making the sound harsher and more crunchy at higher values. The Full/Scoop switch alternates between two fixed Gain/Q filter settings. At the Full position, filtering is less pronounced than at the Scoop position. Overall output level is controlled with the Level knob.
A softer, full-sounding distortion effect. Fuzz sets the amount of saturation applied to the input signal. Volumesets the output level.
An overdrive effect that can emphasize high frequency content in the signal. Level controls the effect output. The Treble/Full switch sets a fixed shelving frequency, allowing either the treble portion or the full range input signal to be processed.
A saturated, somewhat harsh distortion. Roar sets the amount of gain applied to the input signal. Growl setsthe amount of saturation. Tone sets the overall color of the distortion. Higher Tone values increase the treble content of the signal, but there is a corresponding decrease in overall volume. Texture can smooth out or roughen up the distortion. Grain sets the amount of nonlinear distortion applied to the signal. The effect output is controlled with the Level knob.
A fat fuzz effect, that can deliver a soft, saturated distortion. Fuzz controls the input gain. Level sets the ratio between the distorted and source signals. The Tone knob sets the cutoff frequency of the highpass filter.
A metal/hard rock distortion effect. Crunch sets the amount of saturation applied to the input signal. Output gain is set with Level. Tonal color is set with the Tone knob, making the sound brighter at higher values.
Overdrive effect that emulates the distortion produced by a field effect transistor (FET), which is commonly used in solid-state amplifiers. When saturated, FETs generate a warmer sounding distortion than bipolar transistors (such as those emulated by Grinder). Drivesets the saturation amount for the input signal. Tone sets the frequency for the high cut filter, resulting in a softer or harsher tone. The Fat switch, when at the top position, enhances lower frequency content in the signal. Level sets the overall output level of the effect.

Modulation Pedals

This section describes the modulation effects pedals.
46 Chapter 1 Amps and Pedals
Heavenly Chorus
Phase Tripper
Phaze 2
Retro Chorus
Robo Flanger
DescriptionStompbox
A rich, sweet-sounding chorus effect that can significantly thicken the sound. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Feedback sends the output of the effect back in to the input, further thickening the sound, or leading to intermodulations. Delay sets the ratio between the original and effect signals. The upper Bright switch position applies a fixed frequency internal EQ to the signal. At the bottom position, the EQ is bypassed.
A simple phasing effect. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Feedback determines the amount of the effect signal that is routed back into the input. This can change the tonal color, can make the sweeping effect more pronounced, or can do both.
A very flexible dual-phaser effect. LFO 1 and LFO2 Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. Ceiling and Floor determine the frequency range that is swept. Order switches between different algorithms, with higher (even) numbers resulting in a heavier phasing effect. Odd order numbers result in moresubtle comb-filtering effects. Feedback determines the amount of the effect signal that is routed back into the input. This can change the tonal color, can make the phasing effect more pronounced, or can do both. Tone works from the center position; turn it to the left to increase the amount of lowpass filtering, or turn it to the right to increase the amount of highpass filtering. Mix sets the level ratio between each phaser.
A subtle,vintage chorus effect. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect.
Flexible flanging effect. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Feedback determines the amount of the effect signal that is routed back into the input. This can change the tonal color, can make the flanging effect more pronounced, or can do both. The Manual knob sets a delay time between the source and effect signals. This can result in flanger-chorus effects, or in metallic-sounding modulations, particularly when used with high Feedback values.
47Chapter 1 Amps and Pedals
Roswell Ringer
Roto Phase
Spin Box
Total Tremolo
Trem-o-Tone
DescriptionStompbox
A ring modulation effect that can make incoming audio sound metallic (or unrecognizable), can deliver tremolos, brighten up signals and more. The Freq knob sets thecore filter cutoff frequency. Fine is a fine tuning knob for the filter frequency. The Lin/Exp switch determines if the frequency curve is linear (12 notes per octave) or exponential. FB (feedback) determines the amount of the effect signal that is routed back into the input. This can change the tonal color, can make the effect more pronounced, or can do both. Balance between the original and effect signals is set with the Mix knob. See Ringshifter for background information on ring modulation.
A phaser effect that adds movement to, and alters the phase of, the signal. Rate sets the modulation speed and can run freely, or be synchronized withthe host application tempo by enabling theSync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes) with the Rate knob. Intensity sets the strength of the effect. The Vintage/Modern switch activates a fixed-frequency internal EQ when switched to Vintage, and deactivates it when switched to Modern.
Emulation of a Leslie rotor speaker cabinet, commonly used with the Hammond B3 organ. Cabinet sets the type of speaker box. Fast Rate sets the maximum modulation speed (only applies when Fast button is active). Response determines the amount of time required for the rotor to reach its maximum and minimum speed. Drive increases the input gain, introducing distortion to the signal. The Bright switch activates a high shelving filter when turned on. The Slow, Brake and Fast buttons determine how the “speaker” behaves: Slow rotates the speaker slowly. Fast rotates the speaker quickly (up to the maximum speed determined by the Fast Rate knob). Brake stops the speaker rotation. See Rotor Cabinet Effect for background information on the Leslie effect.
A flexible tremolo effect (modulation of the signal level). Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Wave and Smooth work in combination to alter the waveform shape of the LFO. This enables you to create floating changes in level, or abrupt steps. Volume determines the output level of the effect. The 1/2 and 2 x Speed buttons immediately halve or double the current Rate value. Hold down the Speed Up and Slow Down buttons to gradually accelerate or reduce the current Rate value to the maximum or minimum possible values.
A tremolo effect (modulation of the signal level). Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Level sets the post-tremolo gain.
48 Chapter 1 Amps and Pedals
DescriptionStompbox
the Vibe
A vibrato/chorus effect based on the Scanner Vibrato unit found in the Hammond B3 organ. You can choose from three vibrato (V1–3) or chorus (C1–3) variations with the Type knob. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. See
Scanner Vibrato Effect for background information on this effect.

Delay Pedals

This section describes the Delay effects pedals.
DescriptionStompbox
Blue Echo
Spring Box
Tru-Tape Delay
A delay effect. Time sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). The Repeats knob determines the number of delay repeats. Mix sets the balance between the delayed and source signals. The Tone Cut switch controls afixed frequency internal filter circuit that allowsmore low (Lo) or high (Hi)frequency contentto be heard. You can also disable this filter circuit by choosing Off.
A spring reverb pedal. Time sets the length of the reverberation to short, medium, or long values. Tone controls the cutoff frequency, making the effect brighter or darker. Style switches between algorithms, each with different characteristics. You can choose from: Boutique, Simple, Vintage, Bright, and Resonant. Mix sets the ratio between the source and effect signals.
A vintage tape delay effect. The Norm/Reverse switch changes the delay playback direction. Reverse mode is indicated by a blue LED and Normal mode is indicated by a red LED. Hi Cut and Lo Cut activate a fixed frequency filter. Dirt sets the amount of input signal gain, which can introduce an overdriven, saturated quality. Flutter emulates speed fluctuations in the tape transport mechanism. Time sets the modulation speed and can run freely, or be synchronized with the host application tempoby enabling the Sync button.When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Feedback determines the amount of the effect signal that is routed back into the input. The buildup of repeating signals can be used creatively for dub-delay and other effects by adjusting Feedback in real time. Mix sets the balance between the source and effect signals.

Filter Pedals

This section describes the filter effects pedals.
49Chapter 1 Amps and Pedals
DescriptionStompbox
Auto-Funk
Classic Wah
Modern Wah
An auto-wah (filter) effect. Sensitivity sets a threshold which determines how the filter responds toincoming signal levels. Cutoff sets the center frequency for the filter. The BP/LP switch enables either a bandpass or lowpass filter circuit. Signal frequencies just above and below the cutoff point are filtered when the BP switch position is chosen. When the LP switch position is active, only signals below the cutoff point are allowed through the filter. The Hi/Lo switch chooses one of two preset (filter) resonance settings. The Up/Down switch activates a positive or negative modulation direction (the “wah” filtering occurs above or below the source signal frequency).
A funky wah effect, straight from 1970’s TV police show soundtracks. You control it by dragging the pedal.
A more aggressive wah effect. You control it by dragging the pedal. Mode enables you to choose from the following: Retro Wah, Modern Wah, Opto Wah 1, Opto Wah 2, Volume. Each has a different tonal quality. The Q knob determines the resonant characteristics. Low Q values affect a wider frequency range, resulting in softer resonances. High Q values affect a narrower frequency range, resulting in more pronounced emphasis.

Dynamics Pedals

This section describes the dynamics pedals.
DescriptionStompbox
Squash Compressor
A simple compressor. Sustainsets the threshold level. Signals above this are reduced in level. Level determines the output gain. The Attack switch can be set to Fast for signals withfast attack transients, such as drums, or to Slow for signals with slow attack phases, such as strings.

Utility Pedals

This section describes the parameters of the Mixer and Splitter pedals.
Mixer
50 Chapter 1 Amps and Pedals
DescriptionStompbox
A utility that is used to control the level relationship between Bus A and Bus B signals. It can be inserted anywhere in the signal chain, but is typically used at the end of the chain (at the extreme right of the Pedal area). See Using Pedalboard’s Routing Area for details on use. The A/Mix/B switch solos the “A” signal, mixes the “A” and “B” signals, or solos the “B” signal. The level setting of the Mix fader is relevant for all A/Mix/B switch positions.
In stereo instances, the Mixer utility also provides discrete Pan controls for each bus.
Splitter
DescriptionStompbox
A utility that can be inserted anywhere in the signal chain. Splitter can be used in two ways;
When set to Freq, it works as a frequency-dependent signal splitter that divides the incoming signal. Signals above the frequency set with the Frequency knob are sent to Bus B. Signals below this frequency are sent to Bus A.
When set to Split, the incoming signal is routed equally to both buses. The Frequency knob has no impact in this mode.
See Using Pedalboard’s Routing Area for details on use.
51Chapter 1 Amps and Pedals

Delay Effects

2
Delay effects store the input signal—and hold it for a short time—before sending it to the effect input or output.
The held, and delayed, signal is repeated after a given time period, creating a repeating echo effect. Each subsequent repeat is a little quieter than the previous one. Most delays also allow youto feed a percentage of thedelayed signal back to the input. This can result in a subtle, chorus-like effect or cascading, chaotic audio output.
The delay time can often be synchronized to the project tempo by matching the grid resolution of the project, usually in note values or milliseconds.
You can use delays to double individual sounds to resemble a group of instruments playing the same melody, to create echo effects, to place the sound in a large “space”, to generate rhythmic effects, or to enhance the stereo position of tracks in a mix.
Delay effects are generally used as channel insert or bussed effects. They are rarely used on an overall mix (in an output channel), unless you’re trying to achieve an unusual effect.
This chapter covers the following:
Echo (p. 54)
Sample Delay (p. 54)
Stereo Delay (p. 55)
Tape Delay (p. 57)
53

Echo

This simple echoeffect always synchronizes the delay time to the project tempo, allowing you to quickly create echo effects that run in time with your composition.
Time pop-upmenu: Sets the grid resolution of the delay time in musical note durations, based on the project tempo.
• “T” values represent triplets.
• “.” values represent dotted notes.
Repeat slider and field: Determines how often the delay effect is repeated.
Color slider and field: Sets the harmonic content (color) of the delay signal.
Dry and Wet sliders and fields: Control the amount of original and effect signal.

Sample Delay

Sample Delay is more a utility than an effect—you can use it to delay a channel by single sample values.
When used in conjunction with the phase inversion capabilities of the Gain effect, Sample Delay is useful for correction of timing problems that may occur with multichannel microphones. It can also be used creatively, to emulate stereo microphone channel separation.
54 Chapter 2 Delay Effects
Every sample at a frequency of 44.1 kHz is equivalent to the time taken for a sound wave to travel 7.76 millimeters. If you delay one channel of a stereo microphone by 13 samples, this will emulate an acoustic (microphone) separation of 10 centimeters.
Delay slider and field (L and R in stereo version): Determines the number of samples that the incoming signal will be delayed by.
Link L & R button (only in stereo version): Ensures that the number of samples is identical for both channels. Adjusting one channel value will adjust the other.

Stereo Delay

The Stereo Delay works much like the Tape Delay (see Tape Delay), but allows you to set the Delay, Feedback, and Mix parameters separately for the left and right channels. The Crossfeed knob for each stereo side determines the feedback intensity or the level at which each signal is routed to the opposite stereo side. You can freely use the Stereo Delay on mono tracks or busses when you want to create independent delays for the two stereo sides.
55Chapter 2 Delay Effects
Note: If you use the effect on mono channel strips, the track or bus will have two channels from the point of insertion—all Insert slots after the chosen slot will be stereo.
As the parameters for the left and right delays are identical, the descriptions below only cover the left channel—the right channel information is provided in brackets, if named differently. Parameters that are common to both channels are shown separately.
Channel Parameters
Left (Right)Input pop-up menu: Choose the input signal for the two stereo sides. Options include OFF, Left, Right, L + R, L − R.
Left (Right) Delay field: Sets the current delay time in milliseconds (this parameter is dimmed when you synchronize the delay time to the project tempo).
Groove slider and field: Determines the proximity of every second delay repeat to the absolute grid position—in other words, how close every second delay repeat is.
Note buttons: Set the grid resolution for the delay time. These are shown as note durations (these are dimmed when the delay time is not synchronized with the project tempo).
Left (Right) Feedback knob and field: Set the amount of feedback for the left and right delay signals.
Crossfeed Left toRight (Crossfeed Rightto Left)knob and field: Transfer the feedback signal of the left channel to the right channel, and vice versa.
FeedbackPhase button: Use to invert the phase of the corresponding channel’s feedback signal.
Crossfeed Phase button: Use to invert the phase of the crossfed feedback signals.
56 Chapter 2 Delay Effects
Common Parameters
Beat Sync button: Synchronizes delay repeats to the project tempo, including tempo changes.
Output Mix (Left and Right) sliders and fields: Independently control the left and right channel signals.
Low Cut and High Cut sliders and fields: Frequencies below the Low Cut value and above the High Cut value are filtered out of the source signal.

Tape Delay

Tape Delay simulates the warm sound of vintage tape echo machines, with the convenience of easy delay time synchronization to your project tempo. The effect is equipped with a highpass and lowpass filter in the feedback loop, simplifying the creation of authentic dub echo effects. Tape Delay also includes an LFO for delay time modulation, which can be used to produce pleasant or unusual chorus effects, even on long delays.
Feedback slider: Determines the amount of delayed and filtered signal that is routed back to the input of the Tape Delay. Set the Feedback slider to the lowest possible value to generate a single echo. Turn Feedback all the way up to endlessly repeat the signal. The levels of the original signal and its taps (echo repeats) tend to accumulate, and may cause distortion. You can use the internal tape saturation circuit to ensure that these overdriven signals continue to sound good.
Freeze button: Captures the current delay repeats and sustains them until the Freeze button is turned off.
Delay field: Sets the current delay time in milliseconds (this parameter is dimmed when you synchronize the delay time to the project tempo).
Sync button: Synchronizes delay repeatsto the project tempo(including tempo changes).
Tempofield: Sets the current delay time in beats per minute (this parameter is dimmed when you synchronize the delay time to the project tempo).
57Chapter 2 Delay Effects
Groove slider and field: Determines the proximity of every second delay repeat to the absolute grid position—in other words, how close every second delay repeat is. A Groove setting of 50% means that every delay has the same delay time. Settings below 50% result in every second delay being played earlier in time. Settings above 50% result in every second delay being played later in time. 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 buttons: Set the grid resolution for the delay time. These are shown as note durations.
Low Cut and High Cut sliders and fields: Frequencies below the Low Cut value and above the High Cut value are filtered out of the source signal. You can shape the sound of the echoes with the highpass and lowpass filters. The filters are located in the feedback circuit, which means that the filtering effect increases in intensity with each delay repeat. If you want an increasingly muddy and confused tone, move the High Cut slider towards the left. For ever thinner echoes, move the Low Cut slider towards the right. If you’re unable to hear the effect even though you seem to have a suitable configuration, be sure to check out both the Dry and Wet controls and the filter settings—move the High Cut slider to the far right, and the Low Cut slider to the far left.
Smooth slider and field: Evens out the LFO and flutter effect.
LFO Rate knob and field: Sets the frequency of the LFO.
LFO Depth knob andfield: Sets the amount of LFO modulation. A value of 0 turns delay modulation off.
Flutter Rate and Intensity sliders and fields: Simulate the speed irregularities of the tape transports used in analog tape delay units.
Flutter Rate: Sets the speed variation.
Flutter Intensity: Determines how pronounced the effect is.
Dry and Wet sliders and fields: Independently control the amount of original and effect signal.
Distortion Level slider and field (Extended Parameters area): Determines the level of the distorted (tape saturation) signal.
58 Chapter 2 Delay Effects

Distortion Effects

3
You can use Distortion effects to recreate the sound of analog or digital distortion and to radically transform your audio.
Distortion effects simulate the distortion created by vacuum tubes, transistors, or digital circuits. Vacuum tubes were used in audio amplifiers before the development of digital audio technology, and they are still used in musical instrument amplifiers today. When overdriven, they produce a type of distortion that many people find musically pleasing, and which has become a familiar part of the sound of rock and pop music. Analog tube distortion adds a distinctive warmth and bite to the signal.
There are also distortion effects that intentionally cause clipping and digital distortion of the signal. These can be used to modify vocal, music, and other tracks to produce an intense, unnatural effect, or to create sound effects.
Distortion effects include parameters for tone, which let you shape the way the distortion alters the signal (often as a frequency-based filter), and for gain, which let you control how much the distortion alters the output level of the signal.
Warning: When set to high output levels, distortion effects can damage your
hearing—and your speakers. When you adjust effect settings, it is recommended that you lower the output level of the track, and raise the level gradually when you are finished.
This chapter covers the following:
Bitcrusher (p. 60)
Clip Distortion (p. 61)
Distortion Effect (p. 62)
Distortion II (p. 63)
Overdrive (p. 63)
Phase Distortion (p. 64)
59

Bitcrusher

Bitcrusher is a low-resolution digital distortion effect. You can use it to emulate the sound of early digital audio devices, to create artificial aliasing by dividing the sample rate, or to distort signals until they are unrecognizable.
Drive slider and field: Sets the amount of gain in decibels applied to the input signal.
Note: Raising the Drive level tends to increase the amount of clipping at the output of the Bitcrusher as well.
Resolution slider and field: Sets the bit rate (between 1 and 24 bits). This alters the calculation precision of the process. Lowering the value increases the number of sampling errors, generating more distortion. At extremely low bit rates, the amount of distortion can be greater than the level of the usable signal.
Waveform display: Shows the impact of parameters on the distortion process.
Downsampling sliderand field: Reduces the sample rate. A value of 1 x leaves the signal unchanged, a value of 2 x halves the sample rate, and a value of 10 x reduces the sample rate to one-tenth of the original signal. (For example, if you set Downsampling to 10 x , a 44.1 kHz signal is sampled at just 4.41 kHz.)
Note: Downsampling has no impact on the playback speed or pitch of the signal.
Mode buttons: Set the distortion mode to Folded, Cut, or Displaced. Signal peaks that exceed the clip level are processed.
Note: The Clip Level parameter has a significant impact on the behavior of all three modes. This is reflected in the Waveform display, so try each mode button and adjust the Clip Level slider to get a feel for how this works.
Folded: The start and end levels of the clipped signal are unchanged, but the center
portion is effectively folded in half (halved in thelevel above the threshold), resulting in a softer distortion.
Cut: The signal is abruptly distorted when the clipping threshold is exceeded. Clipping
that occurs in most digital systems is closest to Cut mode.
60 Chapter 3 Distortion Effects
Displaced: The start, center and end levels of the signal (above the threshold) are
offset, resulting in a distortion which is less severe as signal levels cross the threshold. The center portion of the clipped signal is also softer than in Cut mode.
Clip Level slider and field: Sets the point (below the clipping threshold of the channel strip) at which the signal starts clipping.
Mix slider and field (Extended Parameters area): Sets the balance between dry (original) and wet (effect) signals.

Clip Distortion

Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra. It can simulate warm, overdriven tube sounds and can also generate drastic distortions.
Clip Distortion features an unusual combination of serially connected filters. The incoming signal is amplified by the Drive value, passes through a highpass filter, and is then subjected to nonlinear distortion. Following the distortion, the signal passes through a lowpass filter. The effect signal is then recombined with the original signal and this mixed signal is sent through a further lowpass filter. All three filters have a slope of 6 dB/octave.
This unique combinationof filters allows for gaps inthe frequency spectra that cansound quite good with this sort of nonlinear distortion.
Drive slider and field: Sets the amount of gain applied to the input signal. After being amplified by the Drive value, the signal passes through a highpass filter.
Tone slider and field: Sets the cutoff frequency (in Hertz) of the highpass filter.
Clip Circuit display: Shows the impact of all parameters, with the exception of the High Shelving filter parameters.
Symmetry sliderand field: Sets the amount of nonlinear (asymmetrical) distortion applied to the signal.
Clip Filter slider and field: Sets the cutoff frequency (in Hertz) of the first lowpass filter.
61Chapter 3 Distortion Effects
Mix slider and field: Sets the ratio between the effect (wet) signal and original (dry) signals, following the Clip Filter.
Sum LPF knob and field: Sets the cutoff frequency (in Hertz) of the lowpass filter. This processes the mixed signal.
(High Shelving) Frequency knob and field: Sets the frequency (in Hertz) of the high shelving filter. If you set the High Shelving Frequency to around 12 kHz, you can use it like the treble control on a mixer channel strip or a stereo hi-fi amplifier. Unlike these types of treble controls, however, you can boost or cut the signal by up to ±30 dB with the Gain parameter.
(High Shelving)Gain knob andfield: Sets the amount of gain applied to the output signal.
Input Gain field and slider (Extended Parameters area): Sets the amount of gain applied to the input signal.
Output Gainfield and slider(Extended Parameters area): Sets the amount of gain applied to the output signal.

Distortion Effect

The Distortion effect simulates the lo-fi, dirty distortion generated by a bipolar transistor. You can use it to simulate playing a musical instrument through a highly overdriven amplifier, or to create unique distorted sounds.
Drive slider and field: Sets the amount of saturation applied to the signal.
Display: Shows the impact of parameters on the signal.
Toneknob and field: Sets the frequency for the high cut filter. Filtering the harmonically rich distorted signal produces a softer tone.
Output sliderand field: Sets the output level. This allows you to compensatefor increases in loudness caused by adding distortion.
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Distortion II

Distortion II emulates the distortion circuit of a Hammond B3 organ. You can use it on musical instruments to recreate this classic effect, or use it creatively when designing new sounds.
PreGain knob: Sets the amount of gain applied to the input signal.
Drive knob: Sets the amount of saturation applied to the signal.
Tone knob: Sets the frequency of the highpass filter. Filtering the harmonically rich distorted signal produces a softer tone.
Type pop-up menu: Choose the type of distortion you want to apply:
Growl: Emulates a two-stage tube amplifier similar to the type found in a Leslie 122
speaker cabinet, which is often used with the Hammond B3 organ.
Bity: Emulates the sound of a bluesy (overdriven) guitar amp.
Nasty: Produces hard distortion, suitable for creating very aggressive sounds.

Overdrive

Overdrive emulates the distortion produced by a field effect transistor (FET), which is commonly used in solid-state musical instrument amplifiers and hardware effects devices. When saturated, FETs generate a warmer-sounding distortion than bipolar transistors, such as those emulated by the Distortion effect.
Drive slider and field: Sets the saturation amount for the simulated transistor.
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Display: Shows the impact of parameters on the signal.
Toneknob and field: Sets the frequency for the high cut filter. Filtering the harmonically rich distorted signal produces a softer tone.
Output sliderand field: Sets the output level. This allows you to compensatefor increases in loudness caused by using Overdrive.

Phase Distortion

The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (see Modulation Effects). Unlike these effects, however, the delay time is not modulated by a low frequency oscillator (LFO), but rather by a lowpass-filtered version of the input signal itself, using an internal sidechain. This means that the incoming signal modulates its own phase position.
The input signal only passes the delay line and is not affected by any other process. The Mix parameter blends the effect signal with the original signal.
Monitor button: Enable to hear the input signal in isolation. Disable to hear the mixed signal.
Cutoff knob and field: Sets the (center) cutoff frequency of the lowpass filter.
Resonance knob and field: Emphasizes frequencies surrounding the cutoff frequency.
Display: Shows the impact of parameters on the signal.
Mix slider and field: Adjusts the percentage of the effect signal mixed with the original signal.
Max Modulation slider and field: Sets the maximum delay time.
Intensity slider and field: Sets the amount of modulation applied to the signal.
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Phase Reverse checkbox (Extended Parametersarea): Enable to reduce the delay time on the right channel when input signals that exceed the cutoff frequency are received. Available only for stereo instances of the Phase Distortion effect.
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Dynamics Processors

4
The Dynamics processors control the perceived loudness of your audio, add focus and punch to tracks and projects, and optimize the sound for playback in different situations.
The dynamic range of an audio signal is the range between the softest and loudest parts of the signal—technically, between the lowest and highest amplitudes. Dynamics processors enable you to adjust the dynamic range of individual audio files, tracks, or an overall project. This can be to increase the perceived loudness and/or to highlight the most important sounds, while ensuring that softer sounds are not lost in the mix.
This chapter covers the following:
Types of Dynamics Processors (p. 67)
Compressor (p. 69)
DeEsser (p. 72)
Ducker (p. 74)
Enveloper (p. 77)
Expander (p. 79)
Limiter (p. 80)
Noise Gate (p. 81)
Preset Multipressor (p. 83)
Silver Compressor (p. 84)
Silver Gate (p. 85)

Types of Dynamics Processors

There are four types of dynamics processors included in Logic Express. These are each used for different audio processing tasks.
Compressors: Logic Express features a number of downward compressors. These behave like an automatic volume control, lowering the volume whenever it rises above a certain level, called the threshold. So, why would you want to reduce the dynamic level?
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By reducing the highest parts of the signal, called peaks, acompressor raises the overall level of the signal, increasing the perceived volume. This gives the signal more focus by making the louder (foreground)parts stand out, while keeping the softer background parts from becoming inaudible. Compression also tends to make sounds tighter or punchier because transients are emphasized, depending on attack and release settings, and because the maximum volume is reached more swiftly.
In addition, compression can make a project sound better when played back in different audio environments. For example, the speakers of a television set or in a car typically have a narrower dynamic range than the sound system in a cinema. Compressing the overall mix can help make the sound fuller and clearer in lower-fidelity playback situations.
Compressors are typically used on vocal tracks to make the singing prominent in an overall mix. They are also commonly used on music and sound effect tracks, but they are rarely used on ambience tracks.
Some compressors—multiband compressors—can divide the incoming signal into different frequency bands and apply different compression settings to each band. This helps to achieve the maximum level without introducing compression artifacts. Multiband compression is typically used on an overall mix.
Expanders: Expanders are similar to compressors, except that they raise, rather than lower, the signal when it exceeds the threshold. Expanders are used to add life to audio signals.
Limiters: Limiters—also called peak limiters—work in a similar way to compressors in that they reduce the audio signal when it exceeds a set threshold. The difference is that whereas a compressor gradually lowers signal levels that exceed the threshold, a limiter quickly reduces any signal that is louder than the threshold, to the threshold level. The main use of a limiter is to prevent clipping while preserving the maximum overall signal level.
Noise gates: Noise gates alter the signal in a way that is opposite to that used by compressors or limiters. Whereas a compressor lowers the level when the signal is louder than the threshold, a noise gate lowers the signal level whenever it falls below the threshold. Louder sounds pass through unchanged, but softer sounds, such as ambient noise or the decay of a sustained instrument, are cut off. Noise gates are often used to eliminate low-level noise or hum from an audio signal.
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Compressor

The Compressor is designed to emulate the sound and response of a professional-level analog (hardware) compressor. It tightens up your audio by reducing sounds that exceed a certain threshold level, smoothing out the dynamics and increasing the overall volume—the perceived loudness. Compression helps bring the key parts of a track or mix into focus, while preventing softer parts from becoming inaudible. It is probably the most versatile and widely used sound-shaping tool in mixing, next to EQ.
You can use the Compressor with individual tracks, including vocal, instrumental, and effects tracks, as well as on the overall mix. Usually you insert the Compressor directly into a channel strip.

Compressor Parameters

The Compressor offers the following parameters:
Circuit Type pop-up menu: Choose the type of circuit emulated by the Compressor. The choices are Platinum, Class(ic) A_R, Class(ic) A_U, VCA, FET, and Opto (optical).
Side Chain Detection pop-up menu: Determines if the Compressor uses the maximum level of each side-chained signal (Max) or the summed level of all side-chained signals (Sum) to exceed or fall below the threshold.
• If either of the stereo channels exceeds or falls below the Threshold, both channels
are compressed.
• If Sum is chosen, the combined level of both channels must exceed the Threshold
before compression occurs.
Gain Reduction meter: Shows the amount of compression in real time.
Attack knob and field: Determines the amount of time it takes for the compressor to react when the signal exceeds the threshold.
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Compression curve display: Shows the compression curve created by the combination of Ratio and Knee parameter values. Input (level) is shown on the x-axis and output (level) on the y-axis.
Release knob and field: Determines the amount of time it takes for the compressor to stop reducing the signal after the signal level falls below the threshold.
Auto button: When the Auto button is active, the release time dynamically adjusts to the audio material.
Ratio slider and field: Sets the compression ratio—the ratio of signal reduction when the threshold is exceeded.
Knee slider and field: Determines the strength of compression at levels close to the threshold. Lower values result in more severe/immediate compression (hard knee). Higher values result in gentler compression (soft knee).
Compressor Threshold slider and field: Sets the threshold level—signals above this threshold value are reduced in level.
Peak/RMSbuttons: Determines whether signal analysis is with thePeak or RMSmethod, when using the Platinum circuit type.
Gain slider and field: Sets the amount of gain applied to the output signal.
Auto Gain pop-up menu: Choose a value to compensate for volume reductions caused by compression. The choices are Off, 0 dB, and −12 dB.
Limiter Threshold slider and field: Sets the threshold level for the limiter.
Limiter button: Turns the integrated limiter on or off.
Output Distortion pop-up menu (Extended Parameters area): Choose whether to apply clipping above 0 dB, and the type of clipping. Choices are: Off, Soft, Hard, and Clip.
Activity pop-up menu (Extended Parameters area): Enables or disables the side chain. Choices are: Off, Listen, and On.
Mode pop-up menu (Extended Parameters area): Choose the type of filter used for the side chain. Choices are: LP (lowpass), BP (bandpass), HP (highpass), ParEQ (parametric), and HS (high shelving).
Frequency slider and field (Extended Parameters area): Sets the center frequency for the side-chain filter.
Q slider and field (Extended Parameters area): Sets the width of the frequency band affected by the side-chain filter.
Gain slider and field (Extended Parameters area): Sets the amount of gain applied to the side-chain signal.
Mix slider and field (Extended Parameters area): Determines the balance between dry (source) and wet (effect) signals.
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Using the Compressor

The following section explains how to use the main Compressor parameters.
Setting the Compressor Threshold and Ratio
The most important Compressor parameters are Threshold and Ratio. The Threshold sets the floor level in decibels. Signals that exceed this level are reduced by the amount set as the Ratio.
The Ratio is a percentage of the overall level; the more the signal exceeds the threshold, the more it is reduced. A ratio of 4:1 means that increasing the input by 4 dB results in an increase of the output by 1 dB, if above the threshold.
As an example, with the Threshold set at −20 dB and the Ratio set to 4:1, a −16 dB peak in the signal (4 dB louder than the threshold) is reduced by 3 dB, resulting in an output level of −19 dB.
Setting Suitable Compressor Envelope Times
The Attack and Release parameters shape the dynamic response of the Compressor. The Attack parameter determinesthe time it takes after the signal exceeds the threshold level before the Compressor starts reducing the signal.
Many sounds, including voices and musical instruments, rely on the initial attack phase to define the core timbre and characteristic of the sound. When compressing these types of sounds, you should set higher Attack values to ensure that the attack transients of the source signal aren’t lost or altered.
When attempting to maximize the level of an overall mix, it is best to set the Attack parameter to a lower value, because higher values often result in no, or minimal, compression.
The Release parameter determines how quickly the signal is restored to its original level after it falls below the threshold level. Set a higher Release value to smooth out dynamic differences in the signal. Set lower Release values if you want to emphasize dynamic differences.
Important: The discussion above is highly reliant on not only the type of source material,
but also the compression ratio and threshold settings.
Setting the Compressor Knee
The Knee parameter determines whether the signal is slightly, or severely, compressed as it approaches the threshold level.
Setting a Knee value close to 0 (zero) results in no compression of signal levels that fall just below the threshold, while levels at the threshold are compressed by the full Ratio amount. This is known as hard knee compression, which can cause abrupt and often unwanted transitions as the signal reaches the threshold.
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Increasing the Knee parameter value increases the amount of compression as the signal approaches the threshold, creating a smoother transition. This is called soft knee compression.
Setting Other Compressor Parameters
As the compressor reduces levels, the overall volume at its output is typically lower than the input signal. You can adjust the output level with the Gain slider.
You can also use the Auto Gain parameter to compensate for the level reduction caused by compression (choose either −12 dB or 0 dB).
When you use the Platinum circuit type, the Compressor can analyze the signal using one of two methods: Peak or root mean square (RMS). While Peak is more technically accurate, RMS provides a better indication of how people perceive the signal loudness.
Note: If you activate Auto Gain and RMS simultaneously, the signal may become over-saturated. If you hear any distortion, switch Auto Gain off and adjust the Gain slider until the distortion is inaudible.
Using a Side Chain with the Compressor
Use of a side chain with a compressor is common. This allows you to use the dynamics (level changes) of another channel strip as a control source for compression. For example, the dynamics of a drum groove can be used to rhythmically change the compression, and therefore dynamics, of a guitar part.
Important: The side-chain signal is used only as a detector or trigger in this situation. The
side-chain source is used to control the Compressor, but the audio of the side-chain signal is not actually routed through the Compressor.
To use a side chain with the Compressor
1 Insert the Compressor into a channel strip.
2 Select the channel strip that carries the desired signal (side-chain source) in the Side Chain
menu of the Compressor plug-in.
3 Choose the desired analysis method (Max or Sum) from the Side Chain Detection pop-up
menu.
4 Adjust the Compressor parameters.

DeEsser

The DeEsser is a frequency-specific compressor, designed to compress a particular frequency band within a complex audio signal. It is used to eliminate hiss (also called sibilance) from the signal.
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The advantage of using the DeEsser rather than an EQ to cut high frequencies is that it compresses the signal dynamically, rather than statically. This prevents the sound from becoming darker when no sibilance is present in the signal. The DeEsser has extremely fast attack and release times.
When using the DeEsser, you can set the frequency range being compressed (the Suppressor frequency) independently of the frequency range beinganalyzed (the Detector frequency). The two ranges can be easily compared in the DeEsser’s Detector and Suppressor frequency range displays
The Suppressor frequency range is reduced in level for as long as the Detector frequency threshold is exceeded.
The DeEsser does not use a frequency-dividing network—a crossover utilizing lowpass and highpass filters. Rather, it isolates and subtracts the frequency band, resulting in no alteration of the phase curve.
The Detector parameters are on the left side of the DeEsser window, and the Suppressor parameters are on the right. The center section includes the Detector and Suppressor displays and the Smoothing slider.
DeEsser Detector Section
Detector Frequency knob and field: Sets the frequency range for analysis.
Detector Sensitivity knob and field: Sets the degree of responsiveness to the input signal.
Monitor pop-up menu: Choose Det(ector) to monitor the isolated Detector signal, Sup(pressor) to monitor the filtered Suppressor signal, Sens(itivity) to remove the sound from the input signal in response to the Sensitivity parameter, or Off to hear the DeEsser output.
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DeEsser Suppressor Section
Suppressor Frequency knob and field: Sets the frequency band that is reduced when the Detector sensitivity threshold is exceeded.
Strength knob and field: Sets the amount of gain reduction for signals that surround the Suppressor frequency.
Activity LED: Indicates active suppression in real time.
DeEsser Center Section
Detector and Suppressor frequency displays: The upper display shows the Detector frequency range. The lower display shows the Suppressor frequency range (in Hz).
Smoothing slider: Sets the reaction speed of the gain reduction start and end phases. Smoothing controls both the attack and release times, as they are used by compressors.

Ducker

Ducking is a common technique used in radio and television broadcasting: When the DJ or announcer speaks while music is playing, the music level is automatically reduced. When the announcement has finished, the music is automatically raised to its original volume level.
Ducker provides a simple means of achieving this result with existing recordings. It does not work in real time.
Note: For technical reasons, Ducker can only be inserted in output and aux channel strips.
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Ducker Parameters

The Ducker has the following parameters:
Ducking On and Off buttons: Enable or disable ducking.
Lookahead On and Off buttons: Enable to ensure that the Ducker reads the incoming signal before processing. This results in no latency—it is primarily intended for slower computers.
Amount slider and field: Defines the amount of volume reduction of the music mix channel strip, which is, in effect, the output signal.
Threshold slider and field: Determines the lowest level that a side-chain signal must attain before it begins to reduce the music mix output level—by the amount set with the Intensity slider. If the side-chain signal level doesn’t reach the threshold, the music mix channel strip volume is not affected.
Attack slider and field: Controls how quickly the volume is reduced. If you want the music mix signal to be gently faded out, set this slider to a high value.
This value also controls whether or not the signal level is reduced before the threshold is reached. The earlier this occurs, the more latency is introduced.
Note: This only works if the ducking signal is not live—the ducking signal must be an existing recording. The host application needs to analyze the signal level before it is played back in order to predefine the point where ducking begins.
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Hold slider and field: Determines the duration for which the music mix channel strip volume is reduced. This control prevents a chattering effect that can be caused by a rapidly changing side-chain level. If the side-chain level hovers around the threshold value rather than clearly exceeding or falling short of it, set the Hold parameter to a high value to compensate for any rapid volume reductions.
Release slider and field: Controls how quickly the volume returns to the original level. Set it to a high value if you want the music mix to slowly fade up after the announcement.

Using the Ducker

The steps below show how to use the Ducker on existing recordings.
Note: For technical reasons, the Ducker plug-in can be inserted only in output and aux channel strips.
To use the Ducker plug-in
1 Insert the plug-in into an aux channel strip.
2 Assign all channel strip outputs that are supposed to “duck” (dynamically lower the volume
of the mix) to a bus—the aux channel strip chosen in step 1.
3 Choose the bus that carries the ducking (vocal) signal in the Side Chain menu of the
Ducker plug-in.
Note: Unlike all other side-chain-capable plug-ins, the Ducker side chain is mixed with the output signal after passing through the plug-in. This ensures that the ducking side-chain signal—the voice-over—is heard at the output.
4 Adjust the Ducker parameters.
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Enveloper

The Enveloper is an unusual processor that lets you shape the attack and release phases of a signal—the signal’s transients, in other words. This makes it a unique tool that can be used to achieve results that differ from other dynamic processors.
Threshold slider and field: Sets the threshold level. Signals that exceed the threshold have their attack and release phase levels altered.
(Attack) Gain slider and field: Boosts or attenuates the attack phase of the signal. When the Gain slider is set to the center position—0%—the signal is unaffected.
Lookahead slider and field: Sets the pre-read analysis time for the incoming signal. This enables the Enveloper to know in advance what signals are coming, enabling accurate and fast processing.
(Attack) Time knob and field: Determines the amount of time it takes for the signal to increase from the threshold level to the maximum Gain level.
Display: Shows the attack and release curves applied to the signal.
(Release) Time knob and field: Determines the amount of time it takes for the signal to fall from the maximum gain level to the threshold level.
(Release) Gainslider and field: Boosts or attenuates the release phase of the signal. When the Gain slider is set to the center position—0%—the signal is unaffected.
Out Level slider and field: Sets the level of the output signal.

Using the Enveloper

The most important parameters of the Enveloper are the two Gain sliders, one on each side of the central display. These govern the Attack and Release levels of each respective phase.
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Boosting the attack phase can add snap to a drum sound, or it can amplify the initial pluck or pick sound of a stringed instrument. Attenuating the attack causes percussive signals to fade in more softly. You can also mute the attack, making it virtually inaudible. A creative use for this effect is alteration of the attack transients to mask poor timing of recorded instrument parts.
Boosting the release phase also accentuates any reverb applied to the affected channel strip. Conversely, attenuating the release phase makes tracks originally drenched in reverb sound drier. This is particularly useful when working with drum loops, but it has many other applications as well. Let your imagination be your guide.
When using the Enveloper, set the Threshold to the minimum value and leave it there. Only when you seriously raise the release phase, which boosts the noise level of the original recording, should you raise the Threshold slider a little. This limits the Enveloper to affecting only the useful part of the signal.
Drastic boosting or cutting of either the release or attack phase may change the overall level of the signal. You can compensate for this by adjusting the Out Level slider.
Generally, you’ll find that Attack Time values of around 20 ms and Release Time values of 1500 ms are good to start with. Then adjust them for the type of signal that you’re processing.
The Lookahead sliderdefines how far into the future of theincoming signal the Enveloper looks, in order to anticipate future events. You generally won’t need to use this feature, except when processing signals with extremely sensitive transients. If you do raise the Lookahead value, you may need to adjust the Attack Time to compensate.
In contrast to a compressor or expander, the Enveloper operates independently of the absolute level of the input signal—but this works only if the Threshold slider is set to the lowest possible value.
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Expander

The Expander is similar in concept to a compressor, but increases, rather than reduces, the dynamic range above the threshold level. You can use the Expander to add liveliness and freshness to your audio signals.
Threshold sliderand field: Sets the threshold level. Signals above this level are expanded.
Peak/RMS buttons: Determine whether the Peak or RMS method is used to analyze the signal.
Attack knob and field: Determines the time it takes for the Expander to respond to signals that exceed the threshold level.
Expansion display: Shows the expansion curve applied to the signal.
Release knob and field: Sets the time it takes for the Expander to stop processing the signal after it falls below the threshold level.
Ratio slider and field: Sets the expansion ratio—the ratio of signal expansion when the threshold is exceeded.
Note: As the Expander is a genuine upward expander—in contrast to a downward expander, which increases the dynamic range below the Threshold—the Ratio slider features a value range of 1:1 to 0.5:1.
Knee slider and field: Determines the strength of expansion at levels close to the threshold. Lower values result in more severe or immediate expansion—hard knee. Higher values result in a gentler expansion—soft knee.
Gain slider and field: Sets the amount of output gain.
Auto Gain button: Compensates for the level increase caused by expansion. When Auto Gain is active, the signal sounds softer, even when the peak level remains the same.
Note: If you dramatically change the dynamics of a signal (with extreme Threshold and Ratio values), you may need to reduce the Gain slider level to avoid distortion. In most cases, turning on Auto Gain will adjust the signal appropriately.
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Limiter

The Limiter works much like a compressor but with one important difference: where a compressor proportionally reduces the signal when it exceeds the threshold, a limiter reduces any peak above the threshold to the threshold level, effectively limiting the signal to this level.
The Limiter is used primarily when mastering. Typically, you apply the Limiter as the very last process in the mastering signal chain, where it raises the overall volume of the signal so that it reaches, but does not exceed, 0 dB.
The Limiter is designed in such a way that if set to 0 dB Gain and 0 dB Output Level, it has no effect on a normalized signal. If the signal clips, the Limiter reduces the level before clipping can occur. The Limiter cannot, however, fix audio that is clipped during recording.
Gain reduction meter: Shows the amount of limiting in real time.
Gain slider and field: Sets the amount of gain applied to the input signal.
Lookahead slider and field: Adjusts how far ahead in milliseconds the Limiter analyzes the audio signal. This enables it to react earlier to peak volumes by adjusting the amount of reduction.
Note: Lookahead causes latency, but this has no perceptible effect when you use the Limiter as a mastering effect on prerecorded material. Set it to higher values if you want the limiting effect to occur before the maximum level is reached, thus creating a smoother transition.
Release sliderand field: Sets the amount of time, after the signal falls below the threshold level, before the Limiter stops processing.
Output Level knob and field: Sets the output level of the signal.
Softknee button: When active, the signal is limited only when it reaches the threshold. The transition to full limiting is nonlinear, producing a softer, less abrupt effect, and reducing distortion artifacts that can be produced by hard limiting.
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Noise Gate

The Noise Gate is commonly used to suppress unwanted noise that is audible when the audio signal is at a low level. You can use it to remove background noise, crosstalk from other signal sources, and low-level hum, among other uses.
The Noise Gate works by allowing signals above the threshold level to pass unimpeded, while reducing signals below the threshold level. This effectively removes lower-level parts of the signal, while allowing the desired parts of the audio to pass.

Noise Gate Parameters

The Noise Gate has the following parameters.
Threshold slider and field: Sets the threshold level. Signals that fall below the threshold will be reduced in level.
Reduction slider and field: Sets the amount of signal reduction.
Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold.
Hold knob and field: Sets the amount of time the gate is kept open after the signal falls below the threshold.
Release knob and field: Sets the amount of time it takes to reach maximum attenuation after the signal falls below the threshold.
Hysteresis slider and field: Sets the difference (in decibels) between the threshold values that open and close the gate. This prevents the gate from rapidly opening and closing when the input signal is close to the threshold.
Lookahead slider and field: Sets how far ahead the Noise Gate analyzes the incoming signal, allowing the effect to respond more quickly to peak levels.
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Monitor button: Enable to hear the side-chain signal, including the effect of the High Cut and Low Cut filters.
High Cut slider and field: Sets the upper cutoff frequency for the side-chain signal.
Low Cut slider and field: Sets the lower cutoff frequency for the side-chain signal.
Note: When no external side chain is selected, the inputsignal is used as theside chain.

Using the Noise Gate

In most situations, setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold value are completely suppressed. Setting Reduction to a higher value attenuates low-level sounds but still allows them to pass. You can also use Reduction to boost the signal by up to 20 dB, which is useful for ducking effects.
The Attack, Hold, and Release knobs modify the dynamic response of the Noise Gate. If you want the gate to open extremely quickly, for percussive signals such as drums, set the Attack knob to a lower value. For sounds with a slow attack phase, such as string pads, set Attack to a higher value. Similarly, when working with signals that fade out gradually or that have longer reverb tails, set a higher Release knob value that allows the signal to fade out naturally.
The Hold knob determines the minimum amount of time that the gate stays open. You can use the Hold knob to prevent abrupt level changes—known as chattering—caused by rapid opening or closing of the gate.
The Hysteresis slider provides another option for preventing chattering, without needing to define a minimum Hold time. Use it to set the range between the threshold values that open and close the Noise Gate. This is useful when the signal level hovers around the Threshold level, causing the Noise Gate to switch on and off repeatedly, producing the undesirable chattering effect. The Hysteresis slider essentially sets the Noise Gate to open at the Threshold level and remain open until the level drops below another, lower, level. As long as the difference between these two values is large enough to accommodate the fluctuating level of the incoming signal, the Noise Gate can function without creating chatter. This value is always negative. Generally, −6 dB is a good place to start.
In some situations, you may find that the level of the signal you want to keep and the level of the noise signal are close, making it difficult to separate them. For example, when you are recording a drum kit and using the Noise Gate to isolate the sound of the snare drum, the hi-hat may also open the gate in many cases. To remedy this, use the side-chain controls to isolate the desired trigger signal with the High Cut and Low Cut filters.
Important: The side-chain signal is used only as a detector/trigger in this situation. The
filters are used to isolate particular trigger signals in the side-chain source, but they have no influence on the actual gated signal—the audio being routed through the Noise Gate.
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To use the side-chain filters
1 Click the Monitor button to hear how the High Cut and Low Cut filters will affect the
incoming trigger signal.
2 Drag the High Cut slider to set the upper frequency. Trigger signals above this are filtered.
3 Drag the Low Cut slider to set the lower frequency. Trigger signals below this are filtered.
The filters allow only very high (loud) signal peaks to pass. In the drum kit example, you could remove the hi-hat signal, which is higher in frequency, with the High Cut filter and allow the snare signal to pass. Turn monitoring off to set a suitable Threshold level more easily.

Preset Multipressor

The Preset Multipressor is a simplified version of the Logic Pro 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 to the output of the plug-in. The aim of independent compression on different frequency bands is to strongly compress the bands that need it, without introducing the pumping effect (on other bands) that would normally be heard when using high compression levels.
The Preset Multipressor features a single pop-up menu that allows you to choose settings optimized for various genres. The names of these settings are self-explanatory. As always, use your ears to determine which one best fits your needs.
Note: The Preset Multipressor is automatically inserted in place of the Multipressor when you load Logic Pro projects (that contain Multipressor instances) into Logic Express.
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Silver Compressor

The Silver Compressor is a simplified version of the Compressor (for usage tips, see Using
the Compressor).
Gain Reduction meter: Shows the amount of compression in real time.
Threshold slider and field: Sets the threshold level. Signals that exceed the threshold are reduced in level.
Attack knob and field: Sets the amount of timeit takes for the compressor to react when the signal exceeds the threshold.
Release knob and field: Sets the amount of time it takes for the compressor to stop reducing the signal, after the signal falls below the threshold.
Ratio slider and field: Sets the ratio by which the signal is reduced, when it exceeds the threshold.
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Silver Gate

The Silver Gate is a simplified version of the Noise Gate (for usage tips, see Using the
Noise Gate).
Lookahead slider and field: Sets how far ahead the noise gate analyzes the incoming signal, allowing the Silver Gate to respond more quickly to peak levels.
Threshold slider and field: Sets the threshold level. Signals that fall below the threshold will be reduced in level.
Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold.
Hold knob and field: Sets the amount of time the gate is kept open after the signal falls below the threshold.
Release knob and field: Sets the amount of time it takes to fully close the gate after the signal falls below the threshold.
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Equalizers

5
An equalizer (commonly abbreviated as EQ) shapes the sound of incoming audio by changing the level of specific frequency bands.
Equalization is one of the most commonly used audio processes, both for music projects and in post-production work for video. You can use EQ to subtly or significantly shape the sound of an audio file, instrument, or project by adjusting specific frequencies or frequency ranges.
All EQs are specialized filters that allow certain frequencies to pass through unchanged while raising (boosting) or lowering (cutting) the level of other frequencies. Some EQs can beused in a “broad-brush” fashion, to boost or cut a large rangeof frequencies. Other EQs, particularly parametric and multiband EQs, can be used for more precise control.
The simplest types of EQs are single-band EQs, which include low cut and high cut, lowpass and highpass, shelving, and parametric EQs.
Multiband EQs (such as the Channel EQ or Fat EQ) combine several filters in one unit, enabling you to control a large part of the frequency spectrum. Multiband EQs allow you to independently set the frequency, bandwidth, and Q factor of each frequency spectrum band. This provides extensive, and precise, tone-shaping on any audio source, be it an individual audio signal or an overall mix.
Logic Express includes a variety of single band and multiband EQs.
This chapter covers the following:
Channel EQ (p. 88)
DJ EQ (p. 91)
Fat EQ (p. 92)
Single-Band EQs (p. 93)
Silver EQ (p. 95)
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Channel EQ

The Channel EQ is a highly versatile multiband EQ. It provides eight frequency bands, including lowpass and highpass filters, low and high shelving filters, and four flexible parametric bands. It also features an integrated Fast Fourier Transform (FFT) Analyzer that you can use to view the frequency curve of the audio you want to modify, allowing you to see which parts of the frequency spectrum may need adjustment.
You can use the Channel EQ to shape the sound of individual tracks or audio files, or for tone-shaping on an overall project mix. The Analyzer and graphic controls make it easy to view and change the audio signal in real time.

Channel EQ Parameters

The left side of the Channel EQ window features the Gain and Analyzer controls. The central area of the window includes the graphic display and parameters for shaping each EQ band.
Channel EQ Gain and Analyzer Controls
Master Gain slider and field: Sets the overall output level of the signal. Use it after boosting or cutting individual frequency bands.
Analyzer button: Turns the Analyzer on or off.
Pre/PostEQ button: Determines whether the Analyzer shows the frequency curve before or after EQ is applied, when Analyzer mode is active.
Resolution pop-upmenu: Sets the sample resolution for the Analyzer, with the following menu items: low (1024 points), medium (2048 points), and high (4096 points).
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Channel EQ Graphic Display Section
Band On/Off buttons: Click to turn the corresponding band on or off. Each button icon indicates the filter type:
Band 1 is a highpass filter.
Band 2 is a low shelving filter.
Bands 3 through 6 are parametric bell filters.
Band 7 is a high shelving filter.
Band 8 is a lowpass filter.
Graphic display: Shows the current curve of each EQ band.
• Drag horizontally in the section of the display that encompasses each band to adjust
the frequency of the band.
• Drag vertically in the section of the display that encompasses each band to adjust
the gain of each band (except bands 1 and 8). The display reflects your changes immediately.
• Drag the pivot point in each band to adjust the Q factor. Q is shown beside the cursor
when it is moved over a pivot point.
Channel EQ Parameter Section
Frequency fields: Adjust the frequency of each band.
Gain/Slope fields: Set the amount of gainfor each band. For bands 1 and 8,this changes the slope of the filter.
Q fields: Adjust the Q factor or resonance for each band—the range of frequencies around the center frequency that are affected.
Note: The Q parameter of Band 1 and Band 8 has no effect when the slope is set to 6 dB/Oct. When the Q parameter is set to an extremely high value, such as 100, these filters affect only a very narrow frequency band and can be used as notch filters.
Link button: Activates Gain-Q coupling, which automatically adjusts the Q (bandwidth) when you raise or lower the gain on any EQ band, to preserve the perceived bandwidth of the bell curve.
Analyzer Mode buttons (Extended Parameters area): Choose Peak or RMS.
Analyzer Decay slider and field (Extended Parameters area): Adjust the decay rate (in dB per second) of the Analyzer curve (peak decay in Peak mode or an averaged decay in RMS mode).
Gain-Q Couple Strength pop-up menu (Extended Parameters area): Choose the amount of Gain-Q coupling.
• Choose “strong” to preserve most of the perceived bandwidth.
• Choose “light” or “medium” to allow some change as you raise or lower the gain.
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• The asymmetric settings feature a stronger coupling for negative gain values than
for positive values, so the perceived bandwidth is more closely preserved when you cut, rather than boost, gain.
Note: If you play back automation of the Q parameter with a different Gain-Q Couple setting, the actual Q values will be different than when the automation was recorded.

Using the Channel EQ

The way you use the Channel EQ is obviously dependent on the audio material and what you intend to do with it, but a useful workflow for many situations is as follows: Set the Channel EQ to a flat response (no frequencies boosted or cut), turn on the Analyzer and play the audio signal. Keep an eye on the graphic display to see which parts of the frequency spectrum have frequent peaks and which parts of the spectrum stay at a low level. Pay particular attention to sections where the signal distorts or clips. Use the graphic display or parameter controls to adjust the frequency bands as desired.
You can reduce or eliminate unwanted frequencies, and you can raise quieter frequencies to make them more pronounced. You can adjust the center frequencies of bands 2 through 7 to affect a specific frequency—either one you want to emphasize, such as the root note of the music, or one you want to eliminate, such as hum or other noise. While doing so, change the Q parameter(s) so that only a narrow range of frequencies are affected, or widen it to alter a broad area.
Each EQ band has a different color in the graphic display. You can graphically adjust the frequency of a band by dragging horizontally. Drag vertically to adjust the amount of gain for the band. For bands 1 and 8, the slope values can be changed only in the parameter area below the graphic display. Each band has a pivot point (a small circle on the curve) at the location of the band’s frequency; you can adjust the Q or width of the band by dragging the pivot point vertically.
You can also adjust the decibel scale of the graphic display by vertically dragging either the left or right edge of the display, where the dB scale is shown, when the Analyzer is not active. When the Analyzer is active, dragging the left edge adjusts the linear dB scale, and dragging the right edge adjusts the Analyzer dB scale.
To increase the resolution of the EQ curve display in the most interesting area around the zero line, drag the dB scale, on the left side of the graphic display, upward. Drag downward to decrease the resolution.
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Using the Channel EQ Analyzer

The Analyzer, when active, makes uses of a mathematical process called a Fast Fourier Transform (FFT) to provide a real-time curve of all frequency components in the incoming signal. This is superimposed over any EQ curves you have set. The Analyzer curve uses the same scale as the EQ curves, making it easy to recognize important frequencies in the incoming audio. This also simplifies the task of setting EQ curves to raise or lower the levels of frequencies/frequency ranges.
The bands derived from FFT analysis are divided in a logarithmic scale—there are more bands in higher octaves than in lower ones.
As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top parameter, on the right side of the graphic display. The visible area represents a dynamic range of 60 dB. Drag vertically to set the maximum value to anywhere between +20 dB and −80 dB. The Analyzer display is always dB-linear.
Note: When choosing a resolution, be aware that higher resolutions require significantly more processing power. High resolution is necessary when trying to obtain accurate analysis of very low bass frequencies, for example. It is recommended that you disable the Analyzer or close the Channel EQ window after setting the appropriate EQ parameters. This will free up CPU resources for other tasks.

DJ EQ

The DJ EQ combines high and low shelving filters, each with a fixed frequency, and one parametric EQ. You can adjust the Frequency, Gain, and Q-Factor of the latter. The DJ EQ allows the filter gain to be reduced by as much as −30 dB.
High Shelf slider and field: Sets the amount of gain for the high shelving filter.
Frequency slider and field: Sets the center frequency of the parametric EQ.
Q-Factor slider and field: Sets the range (bandwidth) of the parametric EQ.
Gain slider and field: Sets the amount of gain for the parametric EQ.
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Low Shelf slider and field: Sets the amount of gain for the low shelving filter.

Fat EQ

The Fat EQ is a versatile multiband EQ which can be used on individual sources or overall mixes. The Fat EQ provides up to five individual frequency bands, graphically displays EQ curves, and includes a set of parameters for each band.
The Fat EQ offers the following parameters.
Band Type buttons: Located above the graphic display. For bands 1–2 and 4–5, click one of the paired buttons to select the EQ type for the corresponding band.
Band 1: Click the highpass or low shelving button.
Band 2: Click the low shelving or parametric button.
Band 3: Always acts as a parametric EQ band.
Band 4: Click the parametric or high shelving button.
Band 5: Click the high shelving or lowpass button.
Graphic display: Shows the EQ curve of each frequency band.
Frequency fields: Sets the frequency for each band.
Gain knobs: Set the amount of gain for each band.
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Q fields: Sets the Q or bandwidth of each band—the range of frequencies around the center frequency that are altered. At low Q factor values, the EQ covers a wider frequency range. At high Q values, the effect of the EQ band is limited to a narrow frequency range. The Q value can significantly influence how audible your changes are—if you’re working with a narrow frequency band, you’ll generally need to cut or boost more drastically to notice the difference.
Note: For bands 1 and 5, this changes the slope of the filter.
Band On/Off buttons: Enables/disables the corresponding band.
Master Gain slider and field: Sets the overall output level of the signal. Use it after boosting or cutting individual frequency bands.

Single-Band EQs

The sections below provide descriptions for the following Logic Express single-band EQ effects:
Low Cut and High Cut Filter
High Pass and Low Pass Filter
High Shelving and Low Shelving EQ
Parametric EQ
You can find these effects by opening the plug-in menu and choosing EQ > Single Band.

Low Cut and High Cut Filter

The Low Cut Filter attenuates the frequency range that falls below the selected frequency. The High Cut Filter attenuates the frequency range above the selected frequency. Use the Frequency slider and field to set the cutoff frequency.

High Pass and 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 eliminate the bass below a selectable frequency.
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In contrast, the Low Pass Filter affects the frequency range above the selected frequency.
Frequency slider and field: Sets the cutoff frequency.
Order sliderand field: Sets the filter order. The more orders used, the stronger the filtering effect.
Smoothing slider and field: Adjusts the amount of smoothing, in milliseconds.

High Shelving and Low Shelving EQ

The Low Shelving EQ affects only the frequency range that falls below the selected frequency. The High Shelving EQ affects only the frequency range above the selected frequency.
Gain slider and field: Sets the amount of cut or boost.
Frequency slider and field: Sets the cutoff frequency.
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Parametric EQ

The Parametric EQ is a simple filter with a variable center frequency. It can be used to boost or cut any frequency band in the audio spectrum, either with a wide frequency range, or as a notch filter with a very narrow range. A symmetrical frequency range on either side of the center frequency is boosted or cut.
Gain slider and field: Sets the amount of cut or boost.
Frequency slider and field: Sets the cutoff frequency.
Q-Factor slider and field: Adjusts the Q (bandwidth).

Silver EQ

The Silver EQ includes three bands—a high shelving EQ, a parametric EQ, and a low shelving EQ. You can adjust the cutoff frequencies for the high shelving and low shelving EQs. You can adjust the center frequency, gain, and Q for the parametric EQ.
High Shelf slider and field: Sets the level of the high shelving EQ.
High Frequency slider and field: Sets the cutoff frequency for the high shelving EQ.
Frequency slider and field: Sets the center frequency of the parametric EQ.
Q-Factor slider and field: Adjusts the range (bandwidth) of the parametric EQ.
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Gain slider and field: Sets the amount of cut or boost for the parametric EQ.
Low Shelf slider and field: Sets the level of the low shelving EQ.
Low Frequency slider and field: Sets the cutoff frequency for the low shelving EQ.
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Filter Effects

6
Filters are used to emphasize or suppress frequencies in an audio signal, resulting in a change to the tonal color of the audio.
Logic Express contains a variety of advanced filter-based effects that you can use to creatively modify your audio. These effects are most often used to radically alter the frequency spectrum of a sound or mix.
Note: Equalizers (EQs) are special types of filters. Typically, they are not used as “effects” per-se, but as tools to refine the frequency spectrum of a sound or mix. See Equalizers.
This chapter covers the following:
AutoFilter (p. 97)
EVOC 20 Filterbank (p. 103)
EVOC 20 TrackOscillator (p. 107)
Fuzz-Wah (p. 119)
Spectral Gate (p. 123)

AutoFilter

The AutoFilter is a versatile filter effect with several unique features. You can use it to create classic, analog-style synthesizer effects, or as a tool for creative sound design.
The effect works by analyzing incoming signal levels through use of a threshold parameter. Any signal level that exceeds the threshold is used as a trigger for a synthesizer-style ADSR envelope or an LFO (low frequency oscillator). These control sources are used to dynamically modulate the filter cutoff.
The AutoFilter allows you to choose between different filter types and slopes, control the amount of resonance, add distortion for more aggressive sounds, and mix the original, dry signal with the processed signal.
97

Getting to Know the AutoFilter Interface

Filter parametersThreshold parameter Envelope parameters
Distortion parametersLFO parameters
Output parameters
The main areas of the AutoFilter window are the Threshold, Envelope, LFO, Filter, Distortion, and Output parameter sections.
Threshold parameter: Sets an input level that—if exceeded—triggers the envelope or LFO, which are used to dynamically modulate the filter cutoff frequency. See AutoFilter
Threshold Parameter.
Envelope parameters: Define how the filter cutoff frequency is modulated over time. See AutoFilter Envelope Parameters.
LFO parameters: Define how the filter cutoff frequency is modulated by the LFO. See
AutoFilter LFO Parameters.
Filter parameters: Control the tonal color of the filtered sound. See AutoFilter Filter
Parameters.
Distortion parameters: Distort the signal both before and after the filter. See AutoFilter
Distortion Parameters.
Output parameters: Set the level of both the dry and effect signal. See AutoFilter Output
Parameters.
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AutoFilter Threshold Parameter

The Threshold parameter analyzes the level of the input signal. If the input signal level exceeds the set threshold level, the envelope and LFO are retriggered—this applies only if the Retrigger button is active.
The envelope and LFO can be used to modulate the filter cutoff frequency.

AutoFilter Envelope Parameters

The envelope is used to shape the filter cutoff over time. When the input signal exceeds the set threshold level, the envelope is triggered.
Attack knob and field: Sets the attack time for the envelope.
Decay knob and field: Sets the decay time for the envelope.
Sustain knob and field: Sets the sustain time for the envelope. If the input signal falls below the threshold level before the envelope sustain phase, the release phase is triggered.
Release knob and field: Sets the release time for the envelope (this is triggered as soon as the input signal falls below the threshold).
Dynamic knob and field: Determines the input signal modulation amount. You can modulate the peak value of the envelope section by varying this control.
Cutoff Mod. slider and field: Determines the impact of the envelope on the cutoff frequency.
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AutoFilter LFO Parameters

The LFO is used as a modulation source for filter cutoff.
Coarse Rate knob, Fine Rate slider and field: Used to set the speed of LFO modulation. Drag the Coarse Rate knob to set the LFO frequency in Hertz. Drag the Fine Rate slider (the semicircular slider above the Coarse Rate knob) to fine-tune the frequency.
Note: The labels shown for the Rate knob, slider, and field change when you activate Beat Sync. Only the Rate knob (and field) is available.
Beat Sync button: Activate to synchronize the LFO to the host application tempo. You can choose from bar values, triplet values, and more. These are determined by the Rate knob or field.
Phase knob and field: Shifts the phase relationship between the LFO rate and the host application tempo—when Beat Sync is active. This parameter is grayed out when Beat Sync is disabled.
Decay/Delay knob and field: Sets the amount of time it takes for the LFO to go from 0 to its maximum value.
Rate Mod. knoband field: Sets therate of modulation for the LFO frequency, independent of the input signal level. Typically, when the input signal exceeds the threshold, the modulation width of the LFO increases from 0 to the Rate Mod. value. This parameter allows you to override this behavior.
Stereo Phase knob and field: In stereo instances of the AutoFilter, sets the phase relationship of the LFO modulations between the two channels.
Cutoff Mod. slider and field: Determines the impact of the LFO on the cutoff frequency.
Retrigger button: When the Retrigger button is active, the waveform starts at 0 each time the threshold is exceeded.
Waveform buttons: Click one of the following buttons to set the shape of the LFO waveform: descending sawtooth, ascending sawtooth, triangle, pulse wave, or random.
Pulse Width slider and field: Shapes the curve of the selected waveform.
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