Apple Logic Pro X Effects User Manual
Logic Pro - X - Effects logic_pro_x_effects Free User Guide for Apple Logic Software, Manual
2013-07-16
User Manual: Apple Logic Pro X Logic Pro X Effects
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- Logic Pro X Effects
- Contents
- Chapter 1: Amps and pedals
- Chapter 2: Delay effects
- Chapter 3: Distortion effects
- Chapter 4: Dynamics processors
- Chapter 5: Equalizers
- Chapter 6: Filter effects
- Filter effects overview
- AutoFilter
- EVOC 20 Filterbank
- EVOC 20 TrackOscillator
- EVOC 20 TrackOscillator overview
- Vocoder overview
- EVOC 20 TrackOscillator interface
- EVOC 20 TrackOscillator analysis in parameters
- Use EVOC 20 TrackOscillator analysis in
- EVOC 20 TrackOscillator U/V detection parameters
- EVOC 20 TrackOscillator synthesis in parameters
- EVOC 20 TrackOscillator oscillators
- EVOC 20 TrackOscillator formant filter
- EVOC 20 TrackOscillator modulation
- EVOC 20 TrackOscillator output parameters
- Fuzz-Wah
- Spectral Gate
- Chapter 7: Imaging processors
- Chapter 8: Metering tools
- Chapter 9: MIDI plug-ins
- Chapter 10: Modulation effects
- Chapter 11: Pitch effects
- Chapter 12: Reverb effects
- Chapter 13: Space Designer convolution reverb
- Chapter 14: Specialized effects and utilities
- Chapter 15: Utilities and tools
- Appendix: Legacy effects

Logic Pro X Eects
For OS X
100
KApple Inc.
Copyright © 2013 Apple Inc. All rights reserved.
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software license agreement. The owner or authorized user
of a valid copy of Logic Pro software may reproduce this
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Apple
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019-2553
Contents
10 Chapter 1: Amps and pedals
10 Amps and pedals overview
10 Amp Designer
10 Amp Designer overview
13 Amp Designer models
18 Build a custom Amp Designer combo
21 Amp Designer equalizer
23 Amp Designer amplier controls
24 Amp Designer eects
26 Amp Designer microphone parameters
27 Bass Amp Designer
27 Bass Amp Designer overview
28 Bass Amp Designer models
29 Build a custom Bass Amp Designer combo
30 Bass Amp Designer signal ow
32 Use the D.I. box
33 Bass Amp Designer amplier controls
34 Bass Amp Designer eects
37 Bass Amp Designer microphone parameters
38 Pedalboard
38 Pedalboard overview
39 Use the Pedal Browser
40 Use Pedalboard’s import mode
41 Use the Pedal area
42 Use Pedalboard’s Router
44 Use Pedalboard’s Macro Controls
45 Pedalboard distortion pedals
46 Pedalboard modulation pedals
50 Pedalboard delay pedals
51 Pedalboard lter pedals
51 Pedalboard dynamics pedals
52 Pedalboard utility pedals
3
53 Chapter 2: Delay eects
53 Delay eects overview
54 Delay Designer
54 Delay Designer overview
55 Delay Designer main display
56 Use the Delay Designer Tap display
59 Create taps in Delay Designer
61 Select, move, and delete taps
63 Edit parameters in the Tap display
68 Delay Designer Tap parameter bar
69 Delay Designer sync mode
70 Delay Designer master parameters
71 Use Delay Designer in surround
72 Echo
72 Sample Delay
73 Stereo Delay
74 Tape Delay
76 Chapter 3: Distortion eects
76 Distortion eects overview
77 Bitcrusher
78 Clip Distortion
79 Distortion eect
79 Distortion II
80 Overdrive
81 Phase Distortion
82 Chapter 4: Dynamics processors
82 Dynamics processors overview
83 Adaptive Limiter
84 Compressor
84 Compressor overview
86 Use Compressor
87 DeEsser
89 Use Ducker
90 Enveloper
92 Expander
93 Limiter
94 Multipressor
94 Multipressor overview
94 Multipressor Display parameters
95 Multipressor Frequency Band parameters
96 Multipressor Output parameters
97 Use Multipressor
98 Noise Gate
98 Noise Gate overview
99 Use Noise Gate
Contents 4
100 Surround Compressor
100 Surround Compressor overview
101 Surround Compressor Link parameters
102 Surround Compressor Main parameters
103 Surround Compressor LFE parameters
104 Chapter 5: Equalizers
104 Equalizers overview
104 Channel EQ
104 Channel EQ overview
105 Channel EQ parameters
106 Channel EQ use tips
107 Channel EQ Analyzer
107 Linear Phase EQ
107 Linear Phase EQ overview
108 Linear Phase EQ parameters
109 Linear Phase EQ use tips
110 Linear Phase EQ Analyzer
110 Match EQ
110 Match EQ overview
111 Match EQ parameters
113 Use Match EQ
115 Edit the Match EQ lter curve
116 Single-Band EQ
117 Chapter 6: Filter eects
117 Filter eects overview
117 AutoFilter
117 AutoFilter overview
118 AutoFilter threshold
118 AutoFilter envelope
119 AutoFilter LFO
120 AutoFilter lter
121 AutoFilter distortion
121 AutoFilter output
12 2 EVOC 20 Filterbank
12 2 EVOC 20 Filterbank overview
12 3 EVOC 20 Filterbank Formant Filter
124 EVOC 20 Filterbank modulation
125 EVOC 20 Filterbank output parameters
12 6 EVOC 20 TrackOscillator
12 6 EVOC 20 TrackOscillator overview
12 6 Vocoder overview
12 7 EVOC 20 TrackOscillator interface
128 EVOC 20 TrackOscillator analysis in parameters
128 Use EVOC 20 TrackOscillator analysis in
129 EVOC 20 TrackOscillator U/V detection parameters
131 EVOC 20 TrackOscillator synthesis in parameters
131 EVOC 20 TrackOscillator oscillators
133 EVOC 20 TrackOscillator formant lter
Contents 5
134 EVOC 20 TrackOscillator modulation
135 EVOC 20 TrackOscillator output parameters
13 6 Fuzz-Wah
13 6 Fuzz-Wah overview
13 6 Auto Wah parameters
13 8 Fuzz-Wah Compressor parameters
13 8 Fuzz parameters
13 9 Spectral Gate
13 9 Spectral Gate overview
140 Use Spectral Gate
141 Chapter 7: Imaging processors
141 Imaging processors overview
141 Binaural Post-Processing
142 Direction Mixer
142 Direction Mixer overview
143 Stereo miking techniques
145 Stereo Spread
146 Chapter 8: Metering tools
146 Metering tools overview
146 BPM Counter
147 Correlation Meter
147 Level Meter plug-in
148 MultiMeter
148 MultiMeter overview
149 MultiMeter Analyzer parameters
150 MultiMeter Goniometer parameters
151 MultiMeter Level Meter
151 MultiMeter Correlation Meter
152 MultiMeter Peak parameters
153 Surround MultiMeter
153 Surround MultiMeter overview
153 Surround MultiMeter Analyzer mode
154 Surround MultiMeter Goniometer mode
155 Surround MultiMeter Level Meter
156 Surround MultiMeter Balance/Correlation
157 Surround MultiMeter Peak parameters
158 Use the Tuner utility
Contents 6
159 Chapter 9: MIDI plug-ins
159 Use MIDI plug-ins
160 Arpeggiator MIDI plug-in
160 Arpeggiator overview
161 Arpeggiator control parameters
162 Arpeggiator note order parameters
167 Arpeggiator pattern parameters
170 Arpeggiator options parameters
171 Arpeggiator keyboard parameters
172 Use Arpeggiator keyboard parameters
173 Assign Arpeggiator controller parameters
174 Chord Trigger MIDI plug-in
174 Chord Trigger overview
175 Use Chord Trigger
178 Modier MIDI plug-in
179 Modulator MIDI plug-in
179 Modulator MIDI plug-in overview
179 Modulator MIDI plug-in LFO
181 Modulator MIDI plug-in envelope
183 Note Repeater MIDI plug-in
184 Randomizer MIDI plug-in
185 Scripter plug-in
185 Use the Scripter plug-in
186 Use the Script Editor
187 Scripter API overview
187 MIDI processing functions
190 JavaScript objects
193 Create Scripter controls
195 Transposer MIDI plug-in
196 Velocity Processor MIDI plug-in
196 Velocity Processor overview
197 Velocity Processor Compress/Expand mode
198 Velocity Processor Value/Range mode
198 Velocity Processor Add/Scale mode
199 Chapter 10: Modulation eects
199 Modulation eects overview
200 Chorus eect
201 Ensemble eect
202 Flanger eect
202 Microphaser
203 Modulation Delay
205 Phaser eect
206 Ringshifter
206 Ringshifter overview
206 Ringshifter interface
207 Set the Ringshifter mode
208 Ringshifter oscillator parameters
209 Ringshifter delay parameters
209 Ringshifter modulation
211 Ringshifter output parameters
Contents 7
212 Rotor Cabinet eect
212 Rotor Cabinet eect overview
213 Rotor Cabinet eect motor parameters
214 Rotor Cabinet eect microphone types
215 Rotor Cabinet eect mic processing controls
216 Scanner Vibrato eect
217 Spreader
218 Tremolo eect
219 Chapter 11: Pitch eects
219 Pitch eects overview
219 Pitch Correction eect
219 Pitch Correction eect overview
220 Pitch Correction eect parameters
221 Pitch Correction eect quantization grid
222 Exclude notes from pitch correction
223 Use Pitch Correction eect reference tuning
224 Pitch Shifter
224 Pitch Shifter overview
225 Use Pitch Shifter
226 Vocal Transformer
226 Vocal Transformer overview
226 Vocal Transformer parameters
227 Use Vocal Transformer
229 Chapter 12: Reverb eects
229 Reverb eects overview
230 EnVerb
230 EnVerb overview
231 EnVerb time parameters
232 EnVerb sound parameters
233 PlatinumVerb
233 PlatinumVerb overview
234 PlatinumVerb early reections parameters
235 PlatinumVerb reverb parameters
236 PlatinumVerb output parameters
237 SilverVerb
238 Chapter 13: Space Designer convolution reverb
238 Space Designer overview
239 Space Designer interface
240 Use impulse responses
243 Space Designer envelopes and EQ
243 Space Designer envelopes and EQ overview
244 Space Designer button bar
245 Edit Space Designer envelope parameters
246 Space Designer volume envelope
247 Space Designer density envelope
248 Use Space Designer EQ parameters
250 Space Designer lter
250 Space Designer lter parameters
251 Space Designer lter envelope
Contents 8
252 Space Designer global parameters
252 Space Designer global parameters overview
253 Use Space Designer global parameters
256 Use Space Designer output parameters
259 Chapter 14: Specialized eects and utilities
259 Specialized eects overview
259 Denoiser
259 Denoiser overview
260 Denoiser smoothing parameters
261 Exciter
262 Grooveshifter
263 Speech Enhancer
264 SubBass
264 SubBass overview
264 SubBass parameters
265 SubBass use tips
266 Chapter 15: Utilities and tools
266 Utilities and tools overview
266 Down Mixer
267 Gain plug-in
268 Use I/O utility
269 Multichannel Gain
270 Test Oscillator
271 Appendix: Legacy eects
271 Legacy eects overview
271 AVerb
272 Bass Amp
273 EQ
273 DJ EQ
274 Fat EQ
275 Single-Band EQs
276 Silver EQ
277 GoldVerb
277 GoldVerb overview
278 GoldVerb early reections parameters
279 GoldVerb reverb parameters
280 Guitar Amp Pro
280 Guitar Amp Pro overview
281 Guitar Amp Pro amplier models
281 Guitar Amp Pro cabinet models
282 Guitar Amp Pro EQ
282 Guitar Amp Pro amplier controls
283 Guitar Amp Pro eects
284 Guitar Amp Pro microphone parameters
285 Silver Compressor
286 Silver Gate
Contents 9
10
Amps and pedals overview
Logic Pro X features an extensive collection of guitar and bass ampliers and classic pedal eects.
You can play live—or process recorded audio and software instrument parts—through these
amps and eects.
The amplier models recreate vintage and modern tube and solid-state amps. Built-in eect
units, such as reverb, tremolo, or vibrato, are also reproduced. The modeled ampliers can be
paired with a number of emulated speaker cabinets. These ampliers and speaker cabinets can
be used as a matching set or combined in other ways to create interesting hybrids.
Also emulated are a number of “classic” foot pedal eects—or stompboxes—that were, and
remain, popular with guitarists and keyboardists. As with their real-world counterparts, you can
chain pedals in any order to create your sound.
Amp Designer
Amp Designer overview
Amp Designer emulates the sound of more than 20 famous guitar ampliers and the speaker
cabinets used with them. Each precongured model combines an amp, a cabinet, and EQ
that recreates a well-known guitar amplier sound. You can process guitar signals directly,
reproducing the sound of your guitar played through these amplication systems. You can also
use Amp Designer for experimental sound design and processing. You can use it with other
instruments as well, applying the sonic character of a guitar amp to a trumpet or vocal part,
for example.
The ampliers, cabinets, and EQs emulated by Amp Designer can be combined in numerous
ways to alter the tone. Virtual microphones are used to pick up the signal of the emulated
amplier and cabinet. You can choose from, and position, seven dierent microphone types.
Amp Designer also emulates classic guitar amplier eects, including spring reverb, vibrato,
and tremolo.
Amps and pedals 1

Chapter 1 Amps and pedals 11
The Amp Designer interface is divided into four main parameter sections.
Model parameters
Output slider
Microphone parameters
Amp
parameters
Effects
parameters
Amp
parameters
•Model parameters: The Model pop-up menu in the black bar at the bottom is used to choose a
precongured model, consisting of an amplier, a cabinet, an EQ type, and a microphone type.
The other pop-up menus in the black bar enable you to independently choose the type of
amplier, cabinet, and microphone. See Build a custom Amp Designer combo on page 18.
•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 Amp Designer amplier controls on
page 23.
•Eects parameters: Located in the center of the knobs section, these parameters control the
integrated eects. See Amp Designer eects overview on page 24.
•Microphone parameters: Located at the right of the interface, these parameters set the type
and position of the microphone that captures the amplier and cabinet sound. See Amp
Designer microphone parameters on page 26.
•Output slider: The Output slider (or the Output eld, in the small interface) is found at the
lower-right corner of the interface. It serves as the nal level control for Amp Designer’s
output that is fed to the ensuing Insert slots in the channel strip or directly to the channel
strip output.
Note: This parameter is dierent from the Master control, which serves the dual purpose of
sound design as well as controlling the level of the Amp section.

Chapter 1 Amps and pedals 12
Switch between the full and small versions of the interface
mClick the disclosure triangle between the Cabinet and Mic pop-up menus in the full interface to
switch to the smaller version.
In the small interface you can access all parameters except microphone selection
and positioning.
mTo switch back to the full interface, click the disclosure triangle beside the Output eld in the
small interface.
Click here in
full interface.
Click here in
small interface.
Choose an Amp Designer model
You can use the Model pop-up menu to choose a precongured model, or you can build
a customized model using the Amp, Cabinet, and Mic pop-up menus. See Build a custom
Amp Designer combo. Your choices remain visible in the pop-up menus and they are also
illustrated in the visual display above them. For example, if you choose Tweed 4X10 from the
Cabinet pop-up menu, you see the Tweed cabinet with four 10" speakers on the right side of
the display.
mChoose a precongured model, consisting of an amplier, a cabinet, an EQ type, and a
microphone type, from the Model pop-up menu.

Chapter 1 Amps and pedals 13
Amp Designer models
Tweed Combos
The Tweed models are based on American combos from the 1950s and early 1960s that helped
dene 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 ampliers are
based on Tweed-style circuitry.
Model Description
Small Tweed Combo A 1 x 12" combo that transitions smoothly from clean
to crunchy, making it a great choice for blues and
rock. For extra denition, set the Treble and Presence
controls to a value around 7.
Large Tweed Combo This 4 x 10" combo was originally intended for
bassists, but it was also used by blues and rock
guitarists. It is more open and transparent-sounding
than the Small Tweed Combo, but it can deliver
crunchy sounds.
Mini Tweed Combo 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 Tweed combos are known for.
Tip: Tweed combos are responsive to 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 soloing.
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 a tight low-end and restrained distortion. They are
useful for clean-toned rock, vintage R & B, surf music, twangy country, jazz, or any other style
where strong note denition is essential.
Model Description
Large Blackface Combo A 4 x 10" combo with a sweet, well-balanced tone
favored by rock, surf, and R & B players. Great for lush,
reverb-saturated chords or strident solos.
Silverface Combo A 2 x 12" combo with a loud, clean tone. It has a
percussive, articulate attack that is suitable for funk,
R & B, and intricate chord work. It can be crunchy
when overdriven, but most players favor it for clean
tones.
Mini Blackface Combo A 1 x 10" combo that is bright and open-sounding,
with reasonable low end impact. It excels at clean
tones with a minimal overdrive.
Small Brownface Combo A 1 x 12" combo that is smooth and rich-sounding,
but retains a level of detail.
Blues Blaster Combo A 1 x 15" combo that has a clear top end with a tight,
dened low end. This model is favored by blues and
rock players.

Chapter 1 Amps and pedals 14
Tip: Although these amps tend toward a clean and tight sound, you can use a Pedalboard
distortion stompbox to attain hard-edged crunch sounds with sharp treble and extended low-
end denition. See Pedalboard distortion pedals on page 45.
British Stacks
The British Stack models are based on the 50- and 100-watt amplier heads that have largely
dened the sound of heavy rock, especially when paired with 4 x 12" cabinets. At medium gain
settings, these amps are suitable for thick chords and ris. 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.
Model Description
Vintage British Stack 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 denitive rock tone.
Modern British Stack 1980s and 1990s descendants of the Vintage British
amplier head, which were optimized for hard rock
and metal styles of the time. Tonally, it has a deeper
and brighter sound at the low and high end, with a
more “scooped” midrange than the Vintage British
amp.
Brown Stack 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.
Tip: The classic British head and 4 x 12" cabinet combo is ideal for ris at high gain levels. These
heads can also sound good through small cabinets, or at clean, low-gain settings.
British Combos
The British Combos capture the brash, treble-rich sound 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 mellow distortion and smooth compression.
Model Description
British Blues Combo This 2 x 12" combo has a loud, aggressive tone that
is cleaner than the British heads, yet delivers rich
distorted tones at high gain settings.
British Combo A 2 x 12" combo based on early 1960s amps. Perfect
for chiming chords and crisp solos.
Small British Combo A 1 x 12" combo with half the power of the British
Combo, this amp oers a darker, less open tone.
Boutique British Combo 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: You can often use higher Treble and Presence knob settings with the British Combos than
with other amp types. 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 a
heavy rock tone. See Pedalboard distortion pedals on page 45.

Chapter 1 Amps and pedals 15
British Alternatives
The late 1960s amplier heads and combos that inspired the Sunshine models are loud and
aggressive, with full mid frequencies. These amps are useful for single note solos, power chords,
and big, open chords—making them popular with the “Brit-pop” bands of the 1990s. The
Stadium amps are famed for their ability to play at extremely high levels without dissolving into
an indistinct distortion. They retain crisp treble and superb note denition, even at maximum
gain settings.
Model Description
Sunshine Stack A robust-sounding head paired with a 4 x 12" cabinet.
It is a good choice for powerful pop-rock chords. If
the tone is too dark, use a high Treble knob setting to
open up the sound.
Small Sunshine Combo A 1 x 12" combo based on a modern amp known for
a “big amp” sound. It is brighter than the Sunshine
Stack head and has tonal qualities similar to the 1960s
British Combo. This amp also sounds good with a 4 x
12" cabinet.
Stadium Stack A classic head and 4 x 12" cabinet conguration
popular with 1970s arena rock bands. Its tones are
cleaner than other Amp Designer 4 x 12" stacks, but it
retains body and impact. A good choice if you need
power and clarity.
Stadium Combo A 2 x 12" combo based on a modern amp. The tone is
smoother than the Stadium Stack.
Tip: The Stadium amps can be slow to distort, so most famous users have paired them with
aggressive fuzz pedals. Try combining them with Pedalboard’s Candy Fuzz or Fuzz Machine
stompboxes. See Pedalboard distortion pedals on page 45.
Metal Stacks
The Metal Stack models are inspired by the powerful, high gain amplier heads favored by
modern hard rock and metal musicians. All are paired with 4 x 12" cabinets. Their signature tones
range from heavy distortion to extremely heavy distortion. These models are ideal if you want
powerful lows, harsh highs, and long sustain in your guitar tones.
Model Description
Modern American Stack A powerful high-gain amp that is ideal for heavy rock
and metal. Use the Mids knob to set the right amount
of scoop or boost.
High Octane Stack Although a powerful, high-gain amp, this model oers
a smooth transition between gain settings and natural
compression. It is a good choice for fast soloing and
for two- or three-note chords.
Turbo Stack An aggressive-sounding amp with spiky highs and
noisy harmonics, especially at high gain settings. Use
the Turbo Stack when you need a guitar tone that
cuts through a mix.
Tip: Combining the Turbo Stack with distortion and fuzz pedals can diminish the amp’s edgy
tone. A dry sound is often the best choice for high-impact ris.

Chapter 1 Amps and pedals 16
Additional Combos
The combos and utility models in this category are versatile amps that you can use for a wide
variety of musical styles.
Model Description
Studio Combo A 1 x 12" combo based on boutique combos of the
1980s and 1990s. These models use multiple gain
stages to generate smooth, sustain-heavy distortion
and bold, bright, clean sounds. Can deliver a heavier
sound when paired with a 4 x 12" cabinet.
Boutique Retro Combo A 2 x 12" combo inspired by expensive modern amps
that combine the sounds of several 1960s combos. It
excels at clean and crunch tones, making it a good
choice when you want an old-fashioned avor but
with the crisp highs and dened lows of a modern
amplier. This model has very sensitive tone controls
that can deliver countless guitar tones.
Pawnshop Combo 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, despite the small
speaker.
Transparent Preamp A preamp stage with no coloration. Note that
Transparent Preamp is activated in the Amp pop-up
menu, not in the Model pop-up menu.
Tip: Combine the Pawnshop Combo amp with Pedalboard’s Hi Drive or Candy Fuzz stompboxes
to emulate hard rock tones of the late 1960s. See Pedalboard distortion pedals on page 45.
Amp Designer cabinets
This table outlines the properties of each cabinet model available in Amp Designer.
Cabinet Description
Tweed 1 x 12 A 12" open-back cabinet from the 1950s with a warm
and smooth tone.
Tweed 4 x 10 A 4 x 10" open-back cabinet from the late 1950s that
was originally conceived for bassists but that guitarists
use for its sparkling presence.
Tweed 1 x 10 A single 10" open-back combo amp cabinet from the
1950s with a smooth sound.
Blackface 4 x 10 Classic open-back cabinet with four 10" speakers. Its
tone is deeper and darker than the Tweed 4 x 10.
Silverface 2 x 12 An open-back model from the 1960s that provides
low-end punch.
Blackface 1 x 10 An open-back 1960s cabinet with glassy highs and
low/mid body.
Brownface 1 x 12 A balanced 1960s open-back cabinet that is smooth,
transparent, and rich-sounding.
Brownface 1 x 15 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.

Chapter 1 Amps and pedals 17
Cabinet Description
Vintage British 4 x 12 This late 1960s closed-back cabinet is synonymous
with classic rock. The tone is big and thick yet
also bright and lively, due to the complex phase
cancelations between the four 30-watt speakers.
Modern British 4 x 12 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 midrange emphasis.
Brown 4 x 12 A closed-back 4 x 12" cabinet with a good low end
and complex midrange.
British Blues 2 x 12 A bright-sounding open-back cabinet with solid lows
and crisp highs, even at high gain settings.
Modern American 4 x 12 A closed-back 4 x 12" cabinet with a full sound. The
lows and mids are denser than the British 4 x 12"
cabinets.
Studio 1 x 12 A compact-sounding open-back cabinet with full
mids and glassy highs.
British 2 x 12 A mid 1960s open-back cabinet with an open, smooth
tone.
British 1 x 12 A small open-back cabinet with crisp highs and low/
mid transparency.
Boutique British 2 x 12 A 2 x 12" cabinet based on the British 2 x 12. It has a
richer midrange and is more powerful in the treble
range.
Sunshine 4 x 12 A 4 x 12" closed-back cabinet with a thick, rich
midrange.
Sunshine 1 x 12 A single 12" open-back combo amp cabinet with
a lively sound that has bright, sweet highs, and
transparent mids.
Stadium 4 x 12 A tight, bright, closed-back British cabinet with bold
upper/mid peaks.
Stadium 2 x 12 A nicely balanced modern British open-back cabinet.
Tonally, it is a compromise between the warmth of the
Blackface 4 x 10 and the brilliance of the British 2 x 12.
Boutique Retro 2 x 12 A 2 x 12" cabinet based on the British 2 x 12. It has
a rich, open midrange and is more powerful in the
treble range.
High Octane 4 x 12 A modern, closed-back European cabinet with strong
lows and highs and scooped mids appropriate for
metal and heavy rock.
Turbo 4 x 12 A modern, closed-back European cabinet with strong
lows, very strong highs, and deeply scooped mids
appropriate for metal and heavy rock.
Pawnshop 1 x 8 A single 8" speaker cabinet that has a strong low-end
punch.
Direct This option bypasses the speaker emulation section.
Tip: A creative sound design option is to choose Direct from the Cabinet pop-up menu, insert
Space Designer in the next free Insert slot, then load one of Space Designer’s “warped” speaker
impulse responses.

Chapter 1 Amps and pedals 18
Build a custom Amp Designer combo
You can use one of the default models or you can create your own hybrid of dierent ampliers,
cabinets, and so on. You create your own by using the Amp, Cabinet, and Mic pop-up menus,
located in the black bar at the bottom of the interface, as well as the EQ pop-up menu,
which you open by clicking the word EQ or Custom EQ above the knobs in the left part of the
knobs section.
Note: If you create your own hybrid amp combo, you can use the Settings pop-up menu to save
it as a setting le, which also includes any parameter changes you have made.
Amp
pop-up menu
Model
pop-up menu
Mic
pop-up menu
Cabinet
pop-up menu
EQ pop-up menu
Choose an Amp Designer amplier
mChoose an amplier from the Amp pop-up menu in the black bar at the bottom of the
Amp Designer interface. See the following sections for details on the characteristics of each
amplier in these categories:
•Tweed Combos on page 13
•Classic American Combos on page 13
•British Stacks on page 14
•British Combos on page 14
•British Alternatives on page 15
•Metal Stacks on page 15
•Additional Combos on page 16
Chapter 1 Amps and pedals 19
Choose an Amp Designer cabinet
Cabinets have a huge impact on the character of a guitar sound (see Amp Designer cabinets on
page 16).
Whereas certain amplier and cabinet pairings have been popular for decades, departing
from them can be an eective way to create fresh-sounding tones. For example, most players
automatically associate British heads with 4 x 12" cabinets. Amp Designer lets you drive a small
speaker with a powerful head, or pair a tiny amp with a 4 x 12" cabinet. You can experiment with
random amplier and cabinet combinations, but you can also make an educated guess about
nontraditional combinations by considering the variables that determine a cabinet’s “sound.”
mChoose a cabinet from the Cabinet pop-up menu in the black bar at the bottom of the
Amp Designer interface. Use the following considerations to guide your decision:
•Combos or Stacks: Combo amps include both an amplier 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 spacious sound. Amplier stacks consist of
an amplier 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 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 in history uses 8" speakers. You can often 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 typically choose cabinets with multiple speakers
for their big sound. The number of speakers is less important than it may appear. Phase
cancelations occur between the speakers, adding texture and interest to the tone. Much of
the “classic rock” sound, for example, is due to tonal peaks and dips caused by interactions
between the speakers in a 4 x 12" cabinet.
Chapter 1 Amps and pedals 20
Choose a microphone type and placement
1 Choose a microphone model from the Mic pop-up menu.
•Condenser models: Emulate the sound of high-end studio condenser microphones. The sound
of condenser microphones is ne, transparent, and well-balanced. Choose Condenser 87 or
Condenser 414.
•Dynamic models: Emulate the sound of popular dynamic cardioid microphones. Dynamic
microphones sound brighter and more cutting than Condenser models. The mid-range is
boosted, with softer lower-mid frequencies, making dynamic microphones a good choice for
rock guitar tones, especially if you want guitars to cut through other tracks in a mix. Choose
Dynamic 20, Dynamic 57, Dynamic 421, or Dynamic 609.
•Ribbon 121: 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.
2 Drag the white dot in the graphic above the Mic pop-up menu to set the microphone position
and distance relative to the cabinet.
Choose and adjust an EQ type
1 Click the word EQ or CUSTOM EQ above the Bass, Mids, and Treble knobs to open the EQ pop-up
menu, then choose an EQ model. See Amp Designer EQ types on page 22.
2 Rotate the Bass, Mids, and Treble knobs to adjust the EQ model you choose.

Chapter 1 Amps and pedals 21
Amp Designer equalizer
Amp Designer equalizer overview
Hardware amplier tone controls vary among models and manufacturers. For example, the treble
knobs on two dierent models may target dierent frequencies or provide dierent levels of cut
or boost. Some equalizer (EQ) sections amplify the guitar signal more than others, thus aecting
the way the amp distorts.
Amp Designer provides multiple EQ types to mirror these variations in hardware ampliers. All
EQ types have identical controls—Bass, Mids, and Treble—but these controls can behave very
dierently depending on which EQ type you choose.
Selecting an EQ type other than the one traditionally associated with an amplier usually results
in signicant tonal changes. As with hardware ampliers, Amp Designer’s EQs are calibrated to
perform well with particular amplier models. Choosing other EQ types can sometimes produce
a thin or unpleasantly distorted tone.
Despite these less pleasant-sounding possibilities, it is worth experimenting with various
amplier and EQ combinations, because many will sound good together.
EQ pop-up menu
Bass, Mids,
and Treble knobs
EQ parameters
•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 aect the
way the Bass, Mids, and Treble knobs respond. See Amp Designer EQ types on page 22.
•Bass, Mids, and Treble knobs: Rotate to adjust the frequency ranges of the EQ models, similar
to the way you would adjust the tone knobs on a hardware guitar amplier. The behavior and
response of these knobs changes when dierent EQ models are chosen.

Chapter 1 Amps and pedals 22
Amp Designer EQ types
This table describes the properties of each Amp Designer EQ type.
EQ type Description
British Bright Inspired by the EQ of British combo amps of the
1960s, it is loud and aggressive, with stronger highs
than the Vintage EQ. This EQ is useful if you want
more treble denition without an overly clean sound.
Vintage Emulates the EQ response of American Tweed-style
amps and the vintage British stack amps that used a
similar circuit. It is loud and subject to distortion. This
EQ is useful if you want a rougher sound.
U.S. Classic Derived from the EQ circuit of the American Blackface
amps, it has a tone of higher delity than the Vintage
EQ, with tighter lows and crisper highs. This EQ is
useful if you want to brighten your tone and reduce
distortion.
Modern Based on a digital EQ unit popular in the 1980s and
1990s, this EQ is useful for sculpting the aggressive
highs, deep lows, and scooped mids associated with
that era’s rock and metal music styles.
Boutique Replicating the tone section of a “retro modern”
boutique amp, it excels at precise EQ adjustments,
though its tone may be too clean when used with
vintage ampliers. This EQ is a good choice if you
want a cleaner, brighter sound.

Chapter 1 Amps and pedals 23
Amp Designer amplier controls
The amp parameters include controls for the input gain, presence, and master output. The Gain
knob is located to the left in the knobs section, the Presence and Master knobs are to the right,
and the Output parameter is at the lower-right edge of the interface.
Presence
Gain Master
Amplier parameters
•Gain knob: Rotate to set the amount of pre-amplication applied to the input signal. This
control aects specic amp models in dierent ways. For example, when you use the British
Amp, the maximum gain setting produces a powerful crunch sound. When you use the
Vintage British Head or Modern British Head, the same gain setting produces heavy distortion,
suitable for lead solos.
•Presence knob: Rotate to adjust the ultra-high frequency range—above the range of the Treble
control. The Presence parameter aects only the output (Master) stage.
•Master knob: Rotate to set the output volume of the amplier signal sent to the cabinet. For
tube ampliers, increasing the Master level typically produces a compressed and saturated
sound, resulting in a more distorted and louder signal.
WARNING: Because high Master knob settings can produce an extremely loud output that
can damage your speakers or hearing, start with a low Master knob setting and then slowly
increase it.
•Output slider or eld: Drag to set the nal output level of Amp Designer.
Note: The slider is replaced with a eld in the small interface.

Chapter 1 Amps and pedals 24
Amp Designer eects
Amp Designer eects overview
The eects parameters include reverb, tremolo, and vibrato, which emulate the processors found
on many ampliers. These controls are found in the center of the knobs section.
Reverb, which is controlled by an On/O switch in the middle, can be added to either tremolo or
vibrato, or it can be used independently. See Amp Designer reverb eect on page 24.
You can select either Trem(olo), which modulates the amplitude or volume of the sound, or
Vib(rato), which modulates the pitch. See Amp Designer tremolo and vibrato on page 25.
Note: The Eects section is placed before the Presence and Master controls in the signal ow, and
receives the pre-amplied, pre-Master signal.
Amp Designer reverb eect
Reverb is always available in Amp Designer, even when you are using a model that is based on
an amplier that provides no reverb function. Reverb is controlled by an On/O switch and a
Level knob in the middle. The Reverb pop-up menu is located above these controls. You can add
Reverb to either the tremolo or vibrato eect, or you can use it independently.
Reverb parameters
•On/O switch: Turns the reverb eect on or o.
•Reverb pop-up menu: Click the word Reverb to open the pop-up menu, which includes the
following reverb types: 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 types on page 25 for information on these reverb types.
•Level knob: Rotate to set the amount of reverb applied to the pre-amplied signal.

Chapter 1 Amps and pedals 25
Amp Designer reverb types
This table indicates the properties of each Amp Designer reverb type.
Reverb type Description
Vintage Spring This bright, splashy sound has largely dened combo
amp reverb since the early 1960s.
Simple Spring A darker, subtler spring sound.
Mellow Spring An even darker, low-delity spring sound.
Bright Spring Has some of the brilliance of Vintage Spring, but with
less surf-style splash.
Dark Spring A moody-sounding spring. More restrained than
Mellow Spring.
Resonant Spring Another 1960s-style spring with a strong, slightly
distorted midrange emphasis.
Boutique Spring A modernized version of the classic Vintage Spring
with a richer tone in the bass and mids.
Sweet Reverb A smooth modern reverb with rich lows and
restrained highs.
Rich Reverb A rich and balanced modern reverb.
Warm Reverb A lush modern reverb with rich lows/mids and
understated highs.
Amp Designer tremolo and vibrato
Tremolo and vibrato are controlled by several switches and two knobs in the eects section.
Tremolo modulates the amplitude or volume of the sound, and Vibrato modulates the pitch of
the sound.
Tremolo and vibrato parameters
•On/O switch: Click to turn the tremolo or vibrato eect on or o.
•Trem(olo)/Vib(rato) switch: Click to choose either tremolo or vibrato.
•Depth knob: Rotate to set the intensity of the modulation for either tremolo or vibrato.
•Speed knob: Rotate to set the speed of the modulation in hertz. Lower settings produce a
smooth, oating sound. Higher settings produce a rotor-like eect.
•Sync/Free switch: Select Sync to synchronize the modulation speed with the host application
tempo. If you select Free, you can use the Speed knob to set the modulation speed to dierent
bar, beat, and musical note values (1/8, 1/16, and so on, including triplet and dotted-note
values).

Chapter 1 Amps and pedals 26
Amp Designer microphone parameters
Amp Designer provides seven virtual microphone types. As with other components in the tone
chain, dierent selections can yield very dierent results. After choosing a cabinet, you can set
the type of microphone to emulate and can place the microphone, relative to the cabinet.
The Mic pop-up menu is near the right end of the black bar. The speaker-adjustment graphic
appears when you move your pointer in 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
eld to switch to the full interface.
Move your mouse above the
Mic pop-up menu to display the
speaker-adjustment graphic.
Microphone parameters
•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 (o-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 is indicated by the white dot in the
speaker-adjustment graphic. Drag the white dot to change the microphone position and
distance, relative to the cabinet. Placement is limited to near-eld positioning.
•Mic pop-up menu: Choose a microphone model:
•Condenser models: Emulates the sound of high-end studio condenser microphones. The
sound of condenser microphones is ne, transparent, and well-balanced. Choose from:
Condenser 87 and Condenser 414.
•Dynamic models: Emulates the sound of popular dynamic cardioid microphones. Dynamic
microphones sound brighter and more cutting than Condenser models. The mid-range is
boosted, with softer lower-mid frequencies, making dynamic microphones a good choice
for rock guitar tones; useful if you want guitars to cut through other tracks in a mix. Choose
from: Dynamic 20, Dynamic 57, Dynamic 421, and Dynamic 609.
•Ribbon 121: 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 dierent microphones in each
Amp Designer instance while retaining identical settings for all other parameters, then set
track signal levels.

Chapter 1 Amps and pedals 27
Bass Amp Designer
Bass Amp Designer overview
Bass Amp Designer emulates the sound of three famous bass guitar ampliers and the speaker
cabinets used with them. Each precongured model combines an amp and cabinet that
recreates a well-known bass guitar amplier sound. The amp and cabinet can be combined
with integrated compression and EQ units to alter the tone. You can process signals directly,
reproducing the sound of your bass played through these amplication systems. Virtual
microphones are used to pick up the signal of the emulated amplier and cabinet. You can
choose from, and position, three dierent microphone types.
When recording, many bass players use a direct connection to a mixing board or other recording
equipment, often using a passive (non powered) or active (powered) D.I. box (Direct Injection
box). The use of a pre-amp with passive or active EQ and a hardware compressor instead of, or in
addition to, a D.I. box is extremely popular too. Bass Amp Designer emulates a professional-level
American D.I. box.
Bass Amp Designer has a two channel design—one for the pre-amp and one for the D.I. box.
This enables you to exibly change the signal ow for the following playing and recording
congurations: pre-amp with passive or active EQ, compressor, a straight power amp, just the
sound of the cabinets and microphones, D.I. box alone, bass amp alone, or both in parallel. See
Amplier signal ow and Pre-amp signal ow.
Model parameters Microphone parameters
Amp parameters Effects parameters Amp parameters Output slider
The Bass Amp Designer interface is divided into four main parameter sections.
•Model parameters: The Model pop-up menu at the left of the black bar at the bottom is used
to choose a precongured model, consisting of an amplier, a cabinet, and a microphone type.
The other menus in the black bar enable you to independently choose the type of amplier,
cabinet, and microphone. See Build a custom Bass Amp Designer combo on page 29.
•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 Bass Amp Designer amplier controls on
page 33.
•Eects parameters: Located in the center of the knobs section, these parameters control the
integrated EQ and compressor eects. A further graphic or parametric EQ is shown above
the compressor controls when the EQ button is turned on. See Bass Amp Designer eects
overview on page 34.
•Microphone parameters: Located at the right of the interface, these parameters set the type
and position of the microphone that captures the amplier and cabinet sound. See Bass Amp
Designer microphone parameters on page 37.

Chapter 1 Amps and pedals 28
•Output slider: The Output slider is found at the lower-right corner of the interface. It serves as
the nal level control for Bass Amp Designer’s output that is fed to the ensuing Insert slots in
the channel strip, or directly to the channel strip output.
Note: This parameter is dierent from the Master control, which serves the dual purpose of
sound design as well as controlling the level of the Amp section.
Choose a Bass Amp Designer model
mChoose a precongured model, consisting of an amplier, a cabinet, and a microphone type,
from the Model pop-up menu.
You can use the Model pop-up menu to choose a precongured model, or you can build a
customized model using the Amp, Cabinet, and Mic pop-up menus. See Build a custom Bass
Amp Designer combo. Your choices remain visible in the pop-up menus, and they are also
illustrated in the visual display above them.
Bass Amp Designer models
Bass amplier models
Bass Amp Designer emulates the three most iconic tube bass amps and cabinets from the 1960s,
1970s, and 1980s. The table includes the cabinets that each amplier is normally matched with.
Amp model Cabinet Description
Classic Amp 8 x 10 inch speakers Emulates a classic six-tube bass
amp with a tuned, closed-back
cabinet introduced in 1960. This
model is good for a range of
musical styles.
Flip Top Amp 1 x 15 inch speaker Emulates a 300-watt tube head
introduced in 1969. It is ideal for
full, fundamental tones.
Modern Amp 3-way speaker array Emulates a 12-tube 360-watt head
introduced in 1989. It is suitable
for many musical styles and is the
ideal choice for highly articulated
performances.

Chapter 1 Amps and pedals 29
Bass cabinet models
The table below outlines the properties of each cabinet model available in Bass Amp Designer.
Cabinet Description
Modern Cabinet 15" 1 x 15 inch speaker, closed-back design. Very deep and
full tone.
Modern Cabinet 10" 1 x 10 inch speaker, closed-back design. A punchy
tone.
Modern Cabinet 6" 1 x 6 inch speaker, closed-back design.
Classic Cabinet 8 X 10" 8 x 10 inch speakers, closed-back design.
Flip Top Cabinet 1 X 15" 1 x 15 inch speaker, closed-back design.
Modern 3 Way 1 x 15 inch speaker, 1 x 10 inch speaker, and 1 x 6 inch
speaker. You can move the microphone vertically and
can position it 20, 30, or 40 cm away from the cabinet.
Direct (PowerAmp Out) A direct signal from the power stage of the emulated
amplier. The cabinet and microphone are removed
from the signal path.
Direct (PreAmp Out) A direct signal from the pre-amplier stage of the
emulated amplier. The cabinet, microphone, and
power amp are removed from the signal path.
Build a custom Bass Amp Designer combo
You can use one of the default models or you can create your own hybrid of dierent ampliers,
cabinets, and so on, using the Amp, Cabinet, and Mic pop-up menus.
Note: If you create your own hybrid amp combo, you can use the Settings pop-up menu to save
it as a setting le, which also includes any parameter changes you have made.
Choose a Bass Amp Designer amplier
mChoose an amplier from the Amp pop-up menu in the black bar at the bottom of the
Amp Designer interface. See Bass amplier models for details on the characteristics of
each amplier.
Choose a Bass Amp Designer cabinet
Cabinets have a huge impact on the character of a bass guitar sound (see Bass cabinet
models on page 29).
Whereas certain amplier and cabinet pairings have been popular for decades, departing from
them is an eective way to create fresh-sounding tones. You can try 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 amplier and cabinet combinations.
mChoose a cabinet from the Cabinet pop-up menu in the black bar at the bottom of the Bass
Amp Designer interface.
•Old or new speakers: Some Bass Amp Designer models 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: Try several sizes and choose the one that works best for
your music.

Chapter 1 Amps and pedals 30
•Single speakers or multiple speakers: The number of speakers is less important than it may
appear. Phase cancelations occur between the speakers, adding texture and interest to
the tone.
Choose a microphone type and placement
1 Click the Mic pop-up menu to choose a microphone model.
•Condenser 87: Emulates the sound of a high-end German studio condenser microphone. The
sound of condenser microphones is ne, transparent, and well-balanced.
•Dynamic 20: Emulates the sound of popular American dynamic cardioid microphones. This
microphone type sounds brighter and more cutting than the Condenser 87 model. The lower-
mid frequencies are rolled o, making it a good choice for miking rock tones. It is especially
useful if you want your bass guitar part to cut through other tracks in a mix.
•Dynamic 421: Emulates the sound of a German dynamic cardioid microphone. It can capture a
wide frequency range and has a slight emphasis of the treble range. It is useful for clean tones.
2 Drag the white dot in the graphic above the Mic pop-up menu to set the microphone position
and distance relative to the cabinet.
Bass Amp Designer signal ow
Amplier signal ow
Bass Amp Designer has a two-channel design—one for the pre-amp and one for the D.I. box.
You can use these independently or can blend them by using the controls on the black bar at
the bottom.
Important: The two channels are always used in parallel if the Blend slider is not set to the far
right or to the far left position.
The channel signal ow changes when you choose dierent models from the Cabinet
pop-up menu.
Cabinet Blend slider position Channel 1 routing Channel 2 routing
Any speaker cabinet
model
Middle Pre-amp, power amp,
cabinet, mic
D.I. box
Direct (PowerAmp Out) Middle Pre-amp, power amp D.I. box
Direct (PreAmp Out) Middle Pre-amp D.I. box
Any speaker cabinet
model
Far left Pre-amp, power amp,
cabinet, mic
Direct (PowerAmp Out) Far left Pre-amp, power amp
Direct (PreAmp Out) Far left Pre-amp
Direct (PreAmp Out) Far right D.I. box

Chapter 1 Amps and pedals 31
Pre-amp signal ow
The pre-amp section is very exible, and can be used in several ways when you use dierent
combinations of On/O and Pre/Post switches. The signal ow indicated in the Mode column is
in series when multiple processors are used—that is, the output of one processor signal is fed
into the next processor.
Mode EQ On/O Compressor On/
O
Additional EQ On/
O
Pre/Post switch
All o O O O
EQ only On O O
Compressor only O On O
Additional EQ only O O On
EQ into Compressor
only
On On O
EQ into Additional
EQ only
On O On
Additional EQ into
Compressor only
O On On Pre
Compressor into
Additional EQ only
O On On Post
All on (EQ into
Additional EQ into
Compressor)
On On On Pre
All on (EQ into
Compressor into
Additional EQ)
On On On Post

Chapter 1 Amps and pedals 32
Use the D.I. box
The D.I. box is modeled on a highly regarded American D.I. unit.
D.I. box parameters
•Boost knob: Rotate to set the input gain of the D.I. box.
•HF Cut button: Click to turn on a highpass lter. This is used to reduce noise.
•Tone knob: Rotate to set the tonal color of the D.I. box. Choose from the following preset EQ
curves:
•1: An EQ curve with a -6 dB scoop from 100 Hz to 10 kHz, most pronounced around 800 Hz.
Suitable for acoustic and string instruments, electric bass, and keyboards.
•2: An EQ curve with a very pronounced -24 dB v-shaped notch from 100 Hz to 10 kHz,
centered around 800 Hz. Suitable for electric bass guitar.
•3: An EQ curve with a -3 dB scoop from 100 Hz to 10 kHz, most pronounced around 800 to
1200 Hz. Suitable for acoustics, strings, electric and bass guitar, and keyboards.
•4: An EQ curve with a -3 dB scoop from 1 kHz to 10 kHz, most pronounced around 8
kHz. Frequencies between 60 Hz and 1 kHz have a slight boost of 1 or 2 dB above unity.
Frequencies above 10 kHz have a +3 dB boost. Suitable for acoustics, strings, electric and
bass guitars, and keyboards.
•5: A sloped EQ curve that ramps up from -24 dB at 10 Hz to + 3 dB at around 900 Hz. Suitable
for acoustic and electric guitar.
•6: A sloped EQ curve that ramps up from -24 dB at 10 Hz to +3 dB at around 900 Hz. The
signal rolls o by approximately 6 dB between 10 and 20 kHz. Suitable for electric and
bass guitar.
•Tone on/o button: Click to turn on the tone control.
Use the D.I. box only
mDrag the Blend slider located in the black bar to the far right.
Use the D.I. box and the amplier in parallel
mDrag the Blend slider located in the black bar to any central position—not to the far right or the
far left.

Chapter 1 Amps and pedals 33
Bass Amp Designer amplier controls
The amp parameters include controls for channel selection, input lter and gain, and master
output. The Gain knob is located to the left in the knobs section and the Master knob and
Output slider are located at the far right.
Bright switch
Gain knob
Channel I/II switch
Master knob
Output slider
Amplier parameters
•Channel I/II switch: Click to switch between channel I and channel II.
•Channel I is active, with a gain of 0 dB.
•Channel II is passive, with a gain of -15 dB.
•Bright switch: Click to switch between normal and bright modes. In the bright position, highs
and upper mids are added to the tone.
Note: The increased mid and high range may lead to a perceived low end roll-o. Use the Bass
EQ knob if you feel the bottom end needs a boost.
•Gain knob: Rotate to set the amount of pre-amplication applied to the input signal. The Gain
knob aects amp models dierently.
•Master knob: Rotate to set the output volume of the amplier signal sent to the cabinet.
Increasing the Master level typically produces a compressed and saturated sound, resulting in
a more distorted and louder signal.
Note: If you choose Direct PowerAmp from the Cabinet pop-up menu, the output signal is
routed directly to the Amp/Direct Box Blend fader. However, if you choose Direct PreAmp from
the Cabinet pop-up menu, the Master knob acts as pre-amp master gain control before the
output signal is routed to the Amp/Direct Box Blend fader.
•Output slider: Drag to set the nal output level of Bass Amp Designer.

Chapter 1 Amps and pedals 34
Bass Amp Designer eects
Bass Amp Designer eects overview
Bass Amp Designer provides multiple EQ types to sculpt your instrument tones.
It provides a basic EQ that mirrors the tonal qualities of the integrated EQ of the amplier model
you choose, if applicable. All amplier model EQs have identical controls: Bass, Mids, and Treble.
See Bass Amp Designer EQ.
Bass Amp Designer also oers an additional Graphic or Parametric EQ that you turn on with the
EQ switch above the Master knob at the far right. See Bass Amp Designer Graphic EQ and Bass
Amp Designer Parametric EQ.
Bass Amp Designer also integrates a dedicated, custom-built compression circuit that is
optimized for electric bass. See Bass Amp Designer compressor.
Bass Amp Designer EQ
The EQ section contains a larger and more inclusive set of the EQ units found in the three
original bass amps emulated by Bass Amp Designer.
EQ parameters
•EQ on/o switch: Click to turn the EQ on or o.
•Bass, Mids, and Treble knobs: Rotate to adjust the frequency ranges of the EQ, similar to the tone
knobs on a hardware amplier.
•Low switch: Click to switch between two positions that aect the tone and behavior of the Bass
EQ knob.
•1-2-3 switch: Click to switch between three positions that aect the tone and behavior of the
Mids EQ knob.
•High switch: Click to switch between two positions that aect the tone and behavior of the
Treble EQ knob.

Chapter 1 Amps and pedals 35
Bass Amp Designer compressor
The internal compression circuit is custom-built for use with Bass Amp Designer. It features an
AutoGain function that compensates for volume reductions caused by compression.
Compressor parameters
•Compressor on/o switch: Click to turn the Compressor on or o.
•Fast/Easy switch: Click to switch between two compression algorithms:
•Fast: Stronger compression, with good control over levels, which makes it easier to t the
bass into an arrangement.
•Easy: Compression with a slow attack and longer sustain phase.
•Comp(ression) knob: Rotate to set the amount of compression intensity applied to the
input signal.
•Gain knob: Rotate to add gain to, or subtract gain from, the gain staging of the internal
AutoGain feature.
Note: AutoGain is always active.
Bass Amp Designer Graphic EQ
Bass Amp Designer oers an additional Graphic or Parametric EQ that you turn on with the EQ
switch above the Master knob at the far right.
Note: The Graphic EQ in a pre-compressor signal ow is enabled by default.
Graphic EQ parameters
•Type switch: Click the up position to choose the Graphic EQ. Click the down position to choose
the Parametric EQ.
Graphic and Parametric EQ parameter settings are retained when switching between EQ types
and when the additional EQ is turned o. This enables you to quickly make AB comparisons.
•Pre/Post switch: Click to determine if the additional EQ is inserted before or after—pre or
post—the compressor section within the signal ow.
Note: This parameter is relevant only if the Compressor is turned on.
•Frequency sliders: Drag to set the amount of boost or cut for each frequency band.

Chapter 1 Amps and pedals 36
Bass Amp Designer Parametric EQ
Bass Amp Designer oers an additional Graphic or Parametric EQ that you turn on with the EQ
switch above the Master knob at the far right. The Parametric EQ provides two EQ bands:
•HiMid: Controls frequencies in the high and high-mid range.
•LoMid: Controls frequencies in the low and low-mid range.
Parametric EQ parameters
•Type switch: Click the up position to choose the Graphic EQ. Click the down position to choose
the Parametric EQ.
Graphic and Parametric EQ parameter settings are retained when switching between EQ types
and when the additional EQ is turned o. This enables you to quickly make AB comparisons.
•Pre/Post switch: Click to determine if the additional EQ is inserted before or after—pre or
post—the compressor section within the signal ow.
Note: This parameter is relevant only if the Compressor is turned on.
•Gain knobs: Rotate to adjust the amount of cut or boost applied to the frequency range set
with the kHz knob.
•kHz knobs: Rotate to set the frequency range that you want to cut or boost with the Gain knob.
•Q knobs: Rotate to set the width of the band surrounding the frequency set with the
kHz knobs.
The lower the Q knob value, the wider the band, which means that more frequencies will
be aected. The higher the Q knob value, the narrower the band, which means that only the
frequencies nearest to the frequency set with the kHz knob will be aected.

Chapter 1 Amps and pedals 37
Bass Amp Designer microphone parameters
Bass Amp Designer oers three virtual microphone types. As with other components in the tone
chain, dierent selections can yield dierent results. After choosing a cabinet, you can choose the
type of microphone to emulate and you can adjust the position of the microphone, relative to
the cabinet.
The Mic pop-up menu is near the right end of the black bar. The speaker-adjustment graphic
appears when you move your mouse in the area above the Mic pop-up menu.
Move your mouse above the
Mic pop-up menu to display
the speaker-adjustment
graphic.
Microphone parameters
•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. If you
place the microphone on the rim of the speaker (o-axis), you obtain a brighter, thinner tone.
Moving the microphone closer to the speaker emphasizes bass response.
The microphone position is shown on the cabinet and is indicated by the white dot in the
speaker-adjustment graphic. Drag the white dot to change the microphone position and
distance, relative to the cabinet. Placement is limited to near-eld positioning.
•Mic pop-up menu: Choose a microphone model:
•Condenser 87: Emulates the sound of a high-end German studio condenser microphone. The
sound of condenser microphones is ne, transparent, and well-balanced.
•Dynamic 20: Emulates the sound of popular American dynamic cardioid microphones. This
microphone type sounds brighter and more cutting than the Condenser 87 model. The
lower-mid frequencies are rolled o, making it a good choice for miking rock tones. It is
especially useful if you want your bass guitar part to cut through other tracks in a mix.
•Dynamic 421: Emulates the sound of a German dynamic cardioid microphone. It can capture
a wide frequency range and has a slight emphasis of the treble range. It is useful for
clean tones.
Tip: Combining multiple microphone types can produce an interesting sound. Duplicate the
bass guitar track, and insert Bass Amp Designer on both tracks. Select dierent microphones
in each Bass Amp Designer instance while retaining identical settings for all other parameters,
then set track signal levels.

Chapter 1 Amps and pedals 38
Pedalboard
Pedalboard overview
Pedalboard simulates the sound of a number of famous “stompbox” pedal eects. You can
process any audio signal with a combination of stompboxes.
You can add, remove, and reorder pedals. The signal ow 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.
All stompbox knobs, switches, and sliders can be automated. Eight Macro controls enable real-
time changes to any pedal parameter with a MIDI controller.
Macro Controls area
Routing area
Pedal Browser
Pedal area
•Pedal Browser: Shows all pedal eects and utilities. These can be dragged into the Pedal area as
part of the signal chain. See Use the Pedal Browser on page 39. This interface area is also used
for the alternative import mode. See Use Pedalboard’s import mode on page 40.
•Pedal area: This is where you determine the order of eects and set eect parameters. You can
add, replace, and remove stompboxes here. See Use the Pedal area on page 41.
•Router: Used to control signal ow in the two eects busses (Bus A and Bus B) available in
Pedalboard. See Use Pedalboard’s Router on page 42.
•Macro Controls: Used to assign eight MIDI controllers, which can be used to control any
stompbox parameter in real time. See Use Pedalboard’s Macro Controls on page 44.

Chapter 1 Amps and pedals 39
Use the Pedal Browser
Pedalboard oers dozens of pedal eects and utilities in the Pedal Browser on the right side of
the interface. Each eect and utility is grouped into a category, such as distortion, modulation,
and so on. The eect and utility pedals are described in the following sections:
•Pedalboard distortion pedals on page 45
•Pedalboard modulation pedals on page 46
•Pedalboard delay pedals on page 50
•Pedalboard lter pedals on page 51
•Pedalboard dynamics pedals on page 51
•Pedalboard utility pedals on page 52
View pop-up menu Import Mode button
Hide or show the Pedal Browser
mClick the disclosure triangle in the lower-right corner of the Pedal area.
Show specic pedal groups in the Pedal Browser
mChoose Distortion, Modulation, Delay, Filter, Dynamics, or Utility from the View pop-up menu.
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.
Add a stompbox to the Pedal area
Do one of the following:
mDrag the eect that you want to insert from the Pedal Browser to the appropriate Pedal area
position. This can be to the left, to the right, or in between existing pedals.
mDouble-click an eect in the Pedal Browser to add it to the right of all existing stompboxes in the
Pedal area.
Note: Double-clicking a stompbox in the Pedal Browser when a stompbox is selected in the
Pedal area replaces the selected pedal.

Chapter 1 Amps and pedals 40
Use Pedalboard’s import mode
Pedalboard has a feature you can use to import parameter settings for each type of pedal. In
contrast to the plug-in window Settings pop-up 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 specic stompbox type.
Turn import mode on or o
mClick the Import Mode button to show all pedals used in the most recent Pedalboard setting.
When the Import Mode button is active, the Pedal Browser switches to an alternate view mode
that displays imported settings. When import mode is inactive, the normal Pedal Browser view
is shown.
Import pedal settings into the Pedal Browser
1 Click the Import Mode button to activate import mode.
Note that the View menu changes to the Select Setting button.
Note: If this is your rst attempt to import settings, a dialog opens where you can select a setting
to import.
2 Click the Select Setting button and select a setting, then click Open.
Depending on the setting you chose, one or more stompboxes appear in the Pedal Browser. The
name of the imported setting is shown at the bottom of the Pedal Browser.
Add an imported pedal to the Pedal area
Do one of the following:
mDrag the stompbox that you want to add from the Pedal Browser to the appropriate Pedal area
position. This can be to the left, to the right, or in between existing pedals.
mMake sure that no pedal is selected in the Pedal area, then double-click a stompbox in the Pedal
Browser to add it to the right of all existing eects in the Pedal area.
Note: The parameter settings of pedals added in import mode are also imported.

Chapter 1 Amps and pedals 41
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 is 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 o 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.
Use the Pedal area
Pedalboard’s stompbox eect pedals not only resemble their physical counterparts; they are also
used in much the same way, albeit without patch cords, power supplies, and screws or locking
mechanisms. The Pedal area layout mirrors a traditional pedalboard, with signals running from
left to right.
Add a pedal to the Pedal area
Do one of the following:
mDrag the stompbox that you want to insert from the Pedal Browser to the appropriate Pedal area
position. This can be to the left, to the right, or in between existing pedals.
mMake sure that no pedal is selected in the Pedal area, then double-click a stompbox in the Pedal
Browser to add it to the right of all existing eects in the Pedal area.
Note: You insert Mixer and Splitter utility pedals in a dierent way. See Use Pedalboard’s
Router on page 42.
Change an eect pedal position in the Pedal area
mDrag the stompbox to a new position, either to the right or the left.
Automation and bus routings, if active, are moved with the eect pedal. For information about
automation and bus routings, see Use Pedalboard’s Router on page 42.

Chapter 1 Amps and pedals 42
Replace a pedal in the Pedal area
Do one of the following:
mDrag the stompbox from the Pedal Browser directly over the pedal you want to replace in the
Pedal area.
mClick to select the stompbox you want to replace in the Pedal area, then double-click the
appropriate pedal in the Pedal Browser.
Note: You can replace “eect” pedals, but not the Mixer or Splitter utilities. Bus routings, if active,
are not changed when an eect pedal is replaced. See Use Pedalboard’s Router on page 42.
Remove a pedal from the Pedal area
Do one of the following:
mDrag the pedal out of the Pedal area.
mClick the pedal to select it, and press the Delete key.
Use Pedalboard’s Router
Pedalboard has two discrete signal busses—Bus A and Bus B—that appear as two horizontal
gray lines in the Router above the Pedal area. These busses provide a great deal of exibility
when you are setting up signal processing chains. All stompboxes that you drag into the Pedal
area are inserted into Bus A—the lower of the two lines—by default.
Note: The Router 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 Router remains open even when your pointer is not over it. You can close the Router
by clicking the small latch button at the top; the Router then opens or closes automatically when
you move your pointer over it.
Notes on Splitter utility 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 point—the leftmost point—within the signal chain.
You cannot drag a Mixer utility to the rst slot in the Pedal area, to between an inserted Splitter
and Mixer utility combination, or directly to the right of an inserted Mixer utility.
Chapter 1 Amps and pedals 43
Create a second bus routing
Do one of the following:
mMove your pointer immediately above the Pedal area to open the Router, and click the name of a
stompbox in the Router.
Two gray lines appear in the Router—the lower one representing Bus A and the upper one Bus
B—and the pedal name moves to the upper line. The chosen stompbox is now routed to Bus B,
and a Mixer utility pedal is automatically added to the end of the signal chain.
mDrag a Splitter utility pedal into the Pedal area when more than one pedal is inserted.
This also inserts a Mixer at the end of the signal chain if one doesn’t already exist.
Remove the second bus routing
Do one of the following:
mDelete the Mixer and Splitter utility pedals from the Pedal area.
mRemove all stompboxes from the Pedal area. This automatically removes any Mixer utility.
Remove an eect from the second bus
mClick the name of the pedal in the Router. (You can also click the lower gray line immediately
above the pedal to remove the pedal from the second bus.)
Note: The removal of all eects from Bus B does not remove the second bus. The Mixer utility
pedal remains in the Pedal area, even when a single stompbox (eect) is in the Pedal area. This
enables parallel routing of wet and dry signals. Only when all pedal eects are removed from the
Pedal area are the Mixer utility and second bus removed.
Determine the split point between busses
When more than one bus is active, a number of dots appear along the “cables”—the gray lines—
in the Router. These represent the output (the socket) of the pedal to the lower left of the dot.
mClick the appropriate dot to determine the split point—the point where the signal is routed
between busses.
A cable appears between the busses when you click a dot.
Note: You cannot create a split point directly before or after the Mixer utility.
Switch between a Splitter utility and a bus split point
mTo replace a bus split point with a Splitter utility: Double-click the bus split point dot in the Router.
The Splitter utility appears in the Pedal area.
mTo replace a Splitter utility with a bus split point: Double-click the Splitter label in the Router.
The Splitter utility is removed from the Pedal area.
Change a Mixer utility position in the Pedal area
mDrag the Mixer utility to a new position, either to the left or to the right.
If you move the Mixer utility to the left, the “downmix” of Bus A and Bus B occurs at the earlier
insertion point. Relevant eect pedals are moved to the right and are inserted into Bus A.
If you move the Mixer utility to the right, the “downmix” of Bus A and Bus B occurs at the later
insertion point. Relevant eect pedals are moved to the left and are inserted into Bus A.
Note: A Mixer pedal cannot be moved to a position directly following or preceding a
corresponding split point or Splitter utility.

Chapter 1 Amps and pedals 44
Change a Splitter utility position in the Pedal area
mDrag the Splitter utility to a new position, either to the left or to the right.
If you move the Splitter utility to the left, the split between Bus A and Bus B occurs at the earlier
insertion point. Relevant eect pedals are moved to the right and are inserted into Bus A.
If you move the Splitter utility to the right, the split between Bus A and Bus B occurs at the later
insertion point. Relevant eect 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.
Use Pedalboard’s Macro Controls
Pedalboard provides eight Macro Targets—A through H—that are found in the Macro Controls
area below the Pedal area. These enable you to map any parameter of an inserted stompbox as a
Macro A–H target. You can save dierent mappings with each Pedalboard setting.
In Logic Pro X, 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 Pro X Help for details.
Click the disclosure 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 elds: Set, and display, the current value for the parameter chosen
from the corresponding Macro Target pop-up menu.
Assign a Macro A–H Target
Do one of the following:
mChoose the parameter that you want to control from any of the Macro A–H Target pop-up
menus.
Each stompbox parameter is shown in the following way: “Slot number—Pedal Name—
Parameter”—for example, “Slot 1—Blue Echo—Time,” or “Slot 2—Roswell Ringer—Feedback.”
The Slot number refers to the position among the pedals, as they appear from left to right in the
Pedal area.
mChoose the “Auto assign” item from any Macro A–H Target pop-up menu, then click the
appropriate parameter in any inserted pedal.
Note: The chosen parameter is displayed in the Macro A–H Target pop-up menu.

Chapter 1 Amps and pedals 45
Pedalboard distortion pedals
This table below describes the distortion eects pedals.
Stompbox Description
Candy Fuzz A bright, “nasty” distortion eect. Drive controls the
input signal gain. Level sets the eect volume.
Double Dragon A deluxe distortion eect. It oers 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 cuto frequency.
The Squash knob sets the threshold for the internal
compression circuit. Contour sets the amount of
nonlinear distortion applied to the signal. Mix sets the
ratio between the source and distorted signals. The
Bright/Fat switch changes between two xed, high
shelving lter frequencies. Blue and red LEDs indicate
the Bright and Fat switch positions, respectively.
Fuzz Machine An American “fuzz” distortion eect. Fuzz controls
the input gain. Overall output gain is set with Level.
The Tone knob increases treble, while simultaneously
reducing low frequencies, as you move it to higher
values.
Grinder Grinder is a lo-, 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 xed
Gain/Q lter settings. At the Full position, ltering is
less pronounced than at the Scoop position. Overall
output level is controlled with the Level knob.
Grit A hard and nasty ltered distortion eect that sounds
great on keyboards and guitars.
Happy Face Fuzz A softer, full-sounding distortion eect. Fuzz sets the
amount of saturation applied to the input signal.
Volume sets the output level.
Hi-Drive An overdrive eect that can emphasize high
frequency content in the signal. Level controls the
eect output. The Treble/Full switch sets a xed
shelving frequency, allowing either the treble portion
or the full range input signal to be processed.
Monster Fuzz A saturated, slightly harsh distortion. Roar sets the
amount of gain applied to the input signal. Growl
sets the 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 eect output is controlled with the Level
knob.
Octafuzz A fat fuzz eect 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 cuto frequency of the highpass
lter.

Chapter 1 Amps and pedals 46
Stompbox Description
Rawk! Distortion A metal/hard rock distortion eect. 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.
Tube Burner A vacuum tube-based distortion that provides a wide
palette of sounds, ranging from warm grain to crispy
overdrive.
Vintage Drive Overdrive eect that emulates the distortion
produced by a eld-eect transistor (FET), commonly
used in solid-state ampliers. When saturated, FETs
generate a warmer sounding distortion than bipolar
transistors, such as those emulated by Grinder. Drive
sets the saturation amount for the input signal. Tone
sets the frequency for the high cut lter, 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 eect.
Pedalboard modulation pedals
This table describes the modulation eects pedals.
Stompbox Description
Dr Octave A classic octaver eect with two independent octave
controls plus an integrated overdrive.
Flange Factory A deluxe anging eect that allows precise control of
every aspect of your sound.
Heavenly Chorus A rich, sweet-sounding chorus eect that thickens
the sound. Rate sets the modulation speed and can
either run freely or be synchronized with the host
application tempo when you enable the Sync button.
When synchronized, you can specify bar, beat, and
note values, including triplets and dotted notes.
Depth sets the strength of the eect. Feedback sends
the output of the eect back in to the input, further
thickening the sound or leading to intermodulations.
Delay sets the ratio between the original and eect
signals. The upper Bright switch position applies
a xed frequency internal EQ to the signal. At the
bottom position, the EQ is bypassed.
Phase Tripper A simple phasing eect. Rate sets the modulation
speed and can either run freely or be synchronized
with the host application tempo when you enable the
Sync button. When synchronized, you can specify bar,
beat, and note values, including triplets and dotted
notes. Depth sets the strength of the eect. Feedback
determines the amount of eect signal that is routed
back into the input. This can change the tonal color,
make the sweeping eect more pronounced, or do
both.

Chapter 1 Amps and pedals 47
Stompbox Description
Phaze 2 A exible dual-phaser eect. LFO 1 and LFO 2 Rate
set the modulation speed and can run freely, or
be synchronized with the host application tempo
when you enable the Sync button. Ceiling and Floor
determine the frequency range that is swept. Order
switches between dierent algorithms, with higher
(even) numbers resulting in a heavier phasing eect.
Odd order numbers result in more subtle comb-
ltering eects. Feedback determines the amount of
eect signal that is routed back into the input. This
can change the tonal color, make the phasing eect
more pronounced, or do both. Tone works from the
center position; turn it to the left to increase the
amount of lowpass ltering, or turn it to the right to
increase the amount of highpass ltering. Mix sets the
level ratio between each phaser.
Retro Chorus A subtle, vintage chorus eect. Rate sets the
modulation speed and can either run freely or be
synchronized with the host application tempo when
you enable the Sync button. When synchronized,
you can specify bar, beat, and note values, including
triplets and dotted notes. Depth sets the strength of
the eect.
Robo Flanger Flexible anging eect. Rate sets the modulation
speed and can either run freely or be synchronized
with the host application tempo when you enable the
Sync button. When synchronized, you can specify bar,
beat, and note values, including triplets and dotted
notes. Depth sets the strength of the eect. Feedback
determines the amount of eect signal that is routed
back into the input. This can change the tonal color,
make the anging eect more pronounced, or do
both. The Manual knob sets a delay time between the
source and eect signals. This can result in anger-
chorus eects, or in metallic-sounding modulations,
particularly when used with high Feedback values.
Roswell Ringer A ring modulation eect that can make incoming
audio sound metallic, or unrecognizable, and can
deliver tremolos, brighten up signals, and more. The
Freq knob sets the core lter cuto frequency. Fine
is a ne tuning knob for the lter frequency. The
Lin/Exp switch determines if the frequency curve is
linear—with 12 notes per octave—or exponential. FB
(feedback) determines the amount of eect signal
that is routed back into the input. This can change
the tonal color, make the eect more pronounced,
or do both. Balance between the original and eect
signals is set with the Mix knob. See Ringshifter
overview on page 206 for background information on
ring modulation.

Chapter 1 Amps and pedals 48
Stompbox Description
Roto Phase A phaser eect that adds movement to, and alters the
phase of, the signal. Rate sets the modulation speed
and can either run freely or be synchronized with the
host application tempo when you enable the Sync
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
eect. The Vintage/Modern switch activates a xed-
frequency internal EQ when switched to Vintage and
deactivates it when switched to Modern.
Spin Box 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—this applies only when the 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 lter 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; and Brake
stops the speaker rotation. See Rotor Cabinet eect
overview on page 212 for background information on
the Leslie eect.
Total Tremolo A exible tremolo eect—modulation of the signal
level. Rate sets the modulation speed and can
either run freely or be synchronized with the host
application tempo when you enable the Sync button.
When synchronized, you can specify bar, beat, and
note values, including triplets and dotted notes. Depth
sets the strength of the eect. Wave and Smooth work
in combination to change the LFO waveform shape.
This enables you to create oating changes in level,
or abrupt steps. Volume determines the output level
of the eect. The 1/2 Speed and 2x 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.
Trem-o-Tone A tremolo eect—modulation of the signal level.
Rate sets the modulation speed and can either run
freely or be synchronized with the host application
tempo when you enable the Sync button. When
synchronized, you can specify bar, beat, and note
values, including triplets and dotted notes. Depth sets
the strength of the eect. Level sets the post-tremolo
gain.

Chapter 1 Amps and pedals 49
Stompbox Description
the Vibe A vibrato/chorus eect 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 either run freely or be synchronized with
the host application tempo when you enable the
Sync button. When synchronized, you can specify bar,
beat, and note values, including triplets and dotted
notes. Depth sets the strength of the eect. See
Scanner Vibrato eect on page 216 for background
information on this eect.
Wham A pedal-controlled pitch shifter. Mix sets the level
balance between the source and pitch-shifted signals.

Chapter 1 Amps and pedals 50
Pedalboard delay pedals
This table describes the Delay eects pedals.
Stompbox Description
Blue Echo A delay eect. Time sets the modulation speed and
can either run freely or be synchronized with the
host application tempo when you enable 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 a
xed frequency internal lter circuit that allows more
low (Lo) or high (Hi) frequency content to be heard.
You can also disable this lter circuit by choosing O.
Through passes the source signal through to the next
pedal, while delay repeats continue.
Spring Box A spring reverb pedal. Time sets the length of the
reverberation to short, medium, or long values.
Tone controls the cuto frequency, making the
eect brighter or darker. Style switches between
algorithms, each with dierent characteristics. You can
choose Boutique, Simple, Vintage, Bright, or Resonant.
Mix sets the ratio between the source and eect
signals.
Tie Dye Delay A warm-sounding reverse delay eect that is perfect
for fans of 1960s and 1970s psychedelic rock.
Tru-Tape Delay A vintage tape delay eect. 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 xed
frequency lter. Dirt sets the amount of input signal
gain, which can introduce an overdriven, saturated
quality. Flutter emulates speed uctuations in the
tape transport mechanism. Time sets the modulation
speed and can either run freely or be synchronized
with the host application tempo when you enable the
Sync button. When synchronized, you can specify bar,
beat, and note values, including triplets and dotted
notes. Feedback determines the amount of eect
signal that is routed back into the input. The buildup
of repeating signals can be used creatively for dub-
delay and other eects by adjusting Feedback in real
time. Mix sets the balance between the source and
eect signals.

Chapter 1 Amps and pedals 51
Pedalboard lter pedals
This table describes the lter eects pedals.
Stompbox Description
Auto-Funk An auto-wah (lter) eect. Sensitivity sets a threshold
that determines how the lter responds to incoming
signal levels. Cuto sets the center frequency for the
lter. The BP/LP switch enables either a bandpass or
lowpass lter circuit. Signal frequencies just above and
below the cuto point are ltered when the BP switch
position is chosen. When the LP switch position is
active, only signals below the cuto point are allowed
through the lter. The Hi/Lo switch determines one of
two preset (lter) resonance settings. The Up/Down
switch activates a positive or negative modulation
direction—the “wah” ltering occurs above or below
the source signal frequency.
Classic Wah A funky wah eect, straight from 1970s TV police show
soundtracks. You control it by dragging the pedal.
Graphic EQ A classic 7-band EQ pedal.
Modern Wah A more aggressive wah eect. 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 dierent
tonal quality. The Q knob determines the resonant
characteristics. Low Q values aect a wider frequency
range, resulting in softer resonances. High Q values
aect a narrower frequency range, resulting in more
pronounced emphasis.
Pedalboard dynamics pedals
This table describes the dynamics pedals.
Stompbox Description
Squash Compressor A simple compressor. Sustain sets 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 with fast attack transients,
such as drums, or to Slow for signals with slow attack
phases, such as strings.

Chapter 1 Amps and pedals 52
Pedalboard utility pedals
This table describes the parameters of the Mixer and Splitter pedals.
Stompbox Description
Mixer Controls 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 Use
Pedalboard’s Router on page 42 for more information.
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 A utility that can be inserted anywhere in the signal
chain. Splitter has two modes:
Freq: 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.
Split: The incoming signal is routed equally to both
buses. The Frequency knob has no impact in this
mode.
See Use Pedalboard’s Router on page 42 for more
information.
53
Delay eects overview
Delay eects store the input signal—and hold it for a short time—before sending it to the eect
input or output.
The held, and delayed, signal is repeated after a given time period, creating a repeating echo
eect. Each subsequent repeat is a little quieter than the previous one. Most delays also allow
you to feed a percentage of the delayed signal back to the input. This can result in a subtle,
chorus-like eect 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 eects, to place the sound in a large “space,” to generate rhythmic
eects, or to enhance the stereo position of tracks in a mix.
Delay eects are generally used as channel insert or bussed eects. They are rarely used on an
overall mix (in an output channel), unless you’re trying to achieve an unusual eect.
Delay eects 2

Chapter 2 Delay eects 54
Delay Designer
Delay Designer overview
Delay Designer is a multitap delay. Unlike traditional delay units that oer only one or two delays
(or taps) that may or may not be fed back into the circuit, Delay Designer provides up to 26
individual taps. These taps are all fed from the source signal and can be edited to create unique
delay eects.
Delay Designer provides control over the level, pan position, and pitch of each tap. Each tap can
also be lowpass or highpass ltered.
Further eect-wide parameters include synchronization, quantization, and feedback.
As the name implies, Delay Designer oers signicant sound design potential. You can use it
for everything from a basic echo eect to an audio pattern sequencer. You can create complex,
evolving, moving rhythms by synchronizing the placement of taps. This leads to further musical
possibilities when coupled with judicious use of transposition and ltering. Alternatively, you can
set up numerous taps as repeats of other taps, much as you would use the feedback control of a
simple delay eect, but with individual control over each repeat.
Tap pads
Master section
Sync section
Tap parameter bar
Main display
The Delay Designer interface consists of ve main sections:
•Main display: Provides a visual representation of all taps. You can see and edit the parameters
of each tap in this area. See Delay Designer main display on page 55.
•Tap parameter bar: Oers a numeric overview of the current parameter settings for the
selected tap. You can view and edit the parameters of each tap in this area. See Delay Designer
Tap parameter bar on page 68.
•Tap pads: You can use these two pads to create taps in Delay Designer. See Create taps in
Delay Designer on page 59.
•Sync section: You can set all Delay Designer synchronization and quantization parameters in
this area. See Delay Designer sync mode on page 69.
•Master section: This area contains the global Mix and Feedback parameters. See Delay Designer
master parameters on page 70.

Chapter 2 Delay eects 55
Delay Designer main display
Delay Designer’s main display is used to view and edit tap parameters. You can choose the
parameter to show and quickly zoom or navigate through all taps.
View buttons
Overview display
Autozoom buttonToggle buttons
Identification bar
Tap display
Main display parameters
•View buttons: Click to choose the parameter or parameters shown in the Tap display. See Use
the Delay Designer Tap display.
•Autozoom button: Zooms the Tap display out, making all taps visible. Turn Autozoom o if
you want to zoom the display in (by dragging vertically in the Overview display) to view
specic taps.
•Overview display: Shows all taps in the time range.
•Toggle buttons: Click to turn the parameters of a particular tap on or o. The parameter being
toggled is selected with the view buttons. The label at the left of the Toggle bar indicates the
parameter. See Use Delay Designer’s tap toggle buttons on page 57.
•Tap display: Represents each tap as a shaded line. Each tap contains a bright bar (or dot for
stereo panning) that indicates the value of the parameter. You can directly edit tap parameters
in the Tap display area. See Edit parameters in the Tap display on page 63.
•Identication bar: Shows an identication letter for each tap. This also serves as a time position
indicator for each tap. You can move taps backward or forward in time along this bar/timeline.
See Select, move, and delete taps on page 61.

Chapter 2 Delay eects 56
Use the Delay Designer Tap display
The view buttons determine the parameter shown in Delay Designer’s Tap display.
The Toggle bar is shown below the view buttons. You can use it to turn parameters on or o for
each tap.
You can use Delay Designer’s Overview display to zoom and to navigate the Tap display area.
Overview display
Tip: If the Overview display is hidden behind a tap, you can move it to the foreground by
holding down Shift.
Select the parameter shown in the Tap display
mClick one (or more) of the following buttons to select the parameter or parameters shown in the
Tap display.
•Cuto button: Shows the highpass and lowpass lter cuto frequencies of taps.
•Reso(nance) button: Shows the lter resonance value of each tap.
•Transp(ose) button: Shows the pitch transposition of each tap.
•Pan button: Shows the pan parameter of each tap.
•For mono to stereo channels, each tap contains a line showing its pan position.
•For stereo to stereo channels, each tap contains a dot showing its stereo balance. A line
extending outward from the dot indicates the tap’s stereo spread.
•For surround channels, each tap contains a line representing its surround angle. See Use
Delay Designer in surround on page 71.
•Level button: Shows the relative volume level of each tap.
Tip: Press Command-Option to temporarily switch the Tap display to Level view from
another view.

Chapter 2 Delay eects 57
Use Delay Designer’s tap toggle buttons
Each tap has its own toggle button in the Toggle bar. These buttons provide a quick way to
graphically turn parameters on and o. The parameter being toggled is determined by the
current view button selection.
1 Click the view button for the parameter you want to toggle.
2 Click the toggle button of each tap that you want to change:
•Cuto view: Turn the lter on or o.
•Reso view: Switch the lter slope between 6 dB and 12 dB.
•Pitch view: Switch pitch transposition on or o.
•Pan view: Switch between the Flip modes.
•Level view: Mute or unmute the tap.
Note: The rst time you edit a lter or pitch transpose parameter, the lter or pitch transposition
module automatically turns on. After you manually turn o either of these modules, however,
you need to manually switch it back on.
Temporarily switch the mute state of taps
mIn any view, Command-Option-click a toggle button.
When you release the Command and Option keys, the toggle buttons return to their standard
functionality in the active view.

Chapter 2 Delay eects 58
Zoom the Tap display
Do one of the following:
mVertically drag the highlighted section (the bright rectangle) in the Overview display.
mHorizontally drag the highlighted bars—to the left or right of the bright rectangle—in the
Overview display.
Note: The Autozoom button needs to be turned o when you manually zoom in the Overview
display. When you zoom in on a small group of taps, the Overview display continues to
show all taps. The area shown in the Tap display is indicated by the bright rectangle in the
Overview display.
Move to dierent sections of the Tap display
mHorizontally drag the middle of the bright rectangle in the Overview display.
The zoomed view in the Tap display updates as you drag.

Chapter 2 Delay eects 59
Create taps in Delay Designer
You can create new delay taps in three dierent ways: by using the Tap pads, by creating them in
the Identication bar, or by copying existing taps.
The fastest way to create multiple taps is to use the Tap pads. If you have a specic rhythm in
mind, you might nd it easier to tap out your rhythm on dedicated hardware controller buttons,
instead of using mouse or trackpad clicks. If you have a MIDI controller, you can assign the Tap
pads to buttons on your device. For information about assigning controllers, see the Control
Surfaces Support Help.
After a tap has been created, you can adjust its position, or you can remove it. See Select, move,
and delete taps on page 61.
Taps are assigned letters, based on their order of creation. The rst tap to be created is assigned
as Tap A, the second tap is assigned as Tap B, and so on. Once assigned, each tap is always
identied by the same letter, even when moved in time, and therefore reordered. For example,
if you initially create three taps, they are named Tap A, Tap B, and Tap C. If you then change the
delay time of Tap B so that it precedes Tap A, it is still called Tap B.
The Identication bar shows the letter of each visible tap. The Tap Delay eld of the Tap
parameter bar displays the letter of the currently selected tap or the letter of the tap being
edited when multiple taps are selected (for details, see Select, move, and delete taps on page 61).
Create taps with the Tap pad
1 Click the upper pad (Start).
Note: Whenever you click the Start pad, it automatically erases all existing taps. Because of this
behavior, after you create your initial taps, you will want to create subsequent taps by clicking in
the Identication bar.
The upper pad label changes to Tap, and a red tap recording bar appears in the strip below the
view buttons.
2 To begin recording new taps, click the Tap button.
3 To create new taps, click the Tap button.
These are created at the exact moments in time of each click, adopting the rhythm of your
click pattern.
4 To nish creating taps, click the Last Tap button.
The nal tap is added, ending tap recording, and assigning the last tap as the feedback tap (for
more information about the feedback tap, see Delay Designer master parameters on page 70).
Note: If you do not click the Last Tap button, tap recording automatically stops after 10 seconds
or when the 26th tap is created, whichever comes rst.

Chapter 2 Delay eects 60
Create a tap in the Identication bar
mClick the position where you want to add a tap.
Copy taps in the Identication bar
mOption-drag a selection of one or more taps to the position where you want to add the tap
or taps.
The delay time of copied taps is set to the drag position.

Chapter 2 Delay eects 61
Select, move, and delete taps
There is always at least one selected tap. You can easily distinguish selected taps by color—the
Toggle bar icons and the Identication bar letters of selected taps are white.
You can move a tap backward or forward in time or completely remove it.
Note: When you move a tap, you are actually editing its delay time.
Select a tap
Do one of the following:
mClick a tap in the Tap display.
mClick the tap letter in the Identication bar.
mClick one of the arrows to the left of the Tap name to select the next or previous tap.
mChoose the tap letter from the pop-up menu to the right of the Tap name.

Chapter 2 Delay eects 62
Select multiple taps
Do one of the following:
mTo select multiple taps: Drag across the background of the Tap display.
mTo select multiple nonadjacent taps: Shift-click specic taps in the Tap display.
Move a selected tap in time
mIn the Identication bar, drag a tap to the left to go forward in time, or to the right to go
backward in time.
This method also works when more than one tap is selected.
Note: Editing the Delay Time parameter in the Tap Delay eld of the Tap parameter bar
also moves a tap in time. For more details about the Tap Delay eld and editing taps, see
Delay Designer Tap parameter bar on page 68.
Delete a tap
Do one of the following:
mSelect a tap, then press the Delete key.
mIn the Identication bar, drag a tap letter downward, out of the Tap display.
This method also works when more than one tap is selected.
Delete all selected taps
mControl-click (or right-click) a tap, then choose “Delete tap(s)” from the shortcut menu.

Chapter 2 Delay eects 63
Edit parameters in the Tap display
You can graphically edit any tap parameter that is represented as a vertical line in
Delay Designer’s Tap display. The Tap display is ideal if you want to edit the parameters of one
tap relative to other taps or when you need to edit or align multiple taps simultaneously.
Edit a tap parameter in the Tap display
1 Click the view button of the parameter you want to edit.
2 Vertically drag the bright line of the tap you want to edit (or one of the selected taps, if multiple
taps are selected).
If you selected multiple taps, the values of all selected taps are changed relative to each other.
Note: The method outlined above is slightly dierent for the Filter Cuto and Pan parameters.
See the tasks below.
Set the values of multiple taps
mCommand-drag horizontally and vertically across several taps in the Tap display.
Parameter values change to match the pointer position as you drag across the taps. Command-
dragging across several taps lets you draw value curves, much like using a pencil to create a
curved line on a piece of paper.

Chapter 2 Delay eects 64
Align the values of several taps
1 Command-click in the Tap display, and drag while holding down the Command key.
A line trails behind the pointer as you drag.
2 Click the appropriate position to mark the end point of the line.
The values of taps that fall between the start and end points are aligned along the line.
Reset the value of a tap
You can use Delay Designer’s Tap display or Tap parameter bar to reset tap parameters to their
default values.
mTo reset a parameter to its default setting in the Tap display: Option-click a tap to reset the selected
parameter to its default setting.
If multiple taps are selected, Option-clicking any tap resets the chosen parameter to its default
value for all selected taps.
mTo reset a parameter to its default setting in the Tap parameter bar: Option-click a parameter value
to reset it to the default setting.
If multiple taps are selected, Option-clicking a parameter of any tap resets all selected taps to the
default value for that parameter.

Chapter 2 Delay eects 65
Edit lter cuto in the Tap display
In Cuto view, each tap actually shows two parameters: highpass and lowpass lter
cuto frequency.
mDrag the cuto frequency line—the upper line is lowpass and the lower line is highpass—to
independently adjust lter cuto values. Both cuto frequencies can be adjusted simultaneously
by dragging in the area between them.
When the highpass lter cuto frequency value is lower than that of the lowpass cuto
frequency, only one line is shown. This line represents the frequency band that passes through
the lters—in other words, the lters act as a bandpass lter. In this conguration, the two lters
operate serially, meaning that the tap passes through one lter rst, then the other.
If the highpass lter’s cuto frequency value is above that of the lowpass lter cuto
frequency, the lter switches from serial operation to parallel operation, meaning that the
tap passes through both lters simultaneously. In this case, the space between the two cuto
frequencies represents the frequency band being rejected—in other words, the lters act as a
band-rejection lter.

Chapter 2 Delay eects 66
Edit pan in the Tap display
The way the Pan parameter is represented in the Pan view is entirely dependent on the input
channel conguration—mono to stereo, stereo to stereo, or surround.
•In mono input/stereo output congurations, all taps are initially panned to the center.
•In stereo input/stereo output congurations, the Pan parameter adjusts the stereo balance,
not the position of the tap in the stereo eld.
Note: Pan is not available in mono congurations.
mTo edit the pan position in mono input/stereo output congurations: Drag vertically from the center
of the tap in the direction you want to pan the tap or taps.
A white line extends outward from the center in the direction you have dragged, reecting the
pan position of the tap or taps.
Lines above the center position indicate pans to the left, and lines below the center position
denote pans to the right. Left (blue) and right (green) channels are easily identied.

Chapter 2 Delay eects 67
mTo adjust the stereo balance in stereo input/stereo output congurations: Drag the Pan parameter—
which appears as a dot on the tap—up or down the tap to adjust the stereo balance.
By default, stereo spread is set to 100%. To adjust the spread width, drag either side of the dot.
As you do so, the width of the line extending outward from the dot changes. Keep an eye on the
Spread parameter in the Tap parameter bar while you are adjusting.
Note: In Surround congurations, the bright line represents the surround angle. See Use
Delay Designer in surround on page 71.
Edit taps with shortcut menu commands
mControl-click (or right-click) a tap in the Tap display, then choose one of the following commands
from the shortcut menu:
•Copy sound parameters: Copies all parameters (except the delay time) of the selected tap or
taps to the Clipboard.
•Paste sound parameters: Pastes the tap parameters from the Clipboard into the selected tap or
taps. If there are more taps in the Clipboard than are selected in the Tap display, the extra taps
in the Clipboard are ignored.
•Reset sound parameters to default values: Resets all parameters of all selected taps (except the
delay time) to the default values.
•2 x delay time: Doubles the delay time of all selected taps. For example, the delay times of three
taps are set as follows: Tap A = 250 ms, Tap B = 500 ms, and Tap C = 750 ms. If you select these
three taps and choose “2 x delay time,” the taps are changed as follows: Tap A = 500 ms, Tap B
= 1000 ms, and Tap C = 1500 ms. In other words, a rhythmic delay pattern unfolds half as fast.
(In musical terms, it is played in half time.)
•1/2 x delay time: Halves the delay time of all selected taps. Using the example above, choosing
“1/2 x delay time” changes the taps as follows: Tap A = 125 ms, Tap B = 250 ms, and Tap C =
375 ms. In other words, a rhythmic delay pattern unfolds twice as fast. (In musical terms, it is
played in double time.)
•Delete tap(s): Deletes all selected taps.

Chapter 2 Delay eects 68
Delay Designer Tap parameter bar
The Tap parameter bar provides access to all parameters of the selected tap. It also shows several
parameters that are not available in the Tap display, such as Transpose and Flip.
Editing the parameters of a single, selected tap is fast and precise because all parameters are
visible, with no need to switch display views or estimate values with vertical lines. If you choose
multiple taps in the Tap display, the values of all selected taps are changed relative to each other.
Option-click a parameter value to reset it to the default setting. If multiple taps are selected,
Option-clicking a parameter of any tap resets all selected taps to the default value for
that parameter.
Tap parameter bar controls
•Filter On/O button: Turns the highpass and lowpass lters on or o (for the selected tap).
•HP-Cuto-LP elds: Drag to set the cuto frequencies (in Hz) for the highpass and
lowpass lters.
•Slope buttons: Determine the steepness of the highpass and lowpass lter slope. Click the
6 dB button for a gentler lter slope, or click the 12 dB button for a steeper, more pronounced
ltering eect.
Note: You cannot set the slope of the highpass and lowpass lters independently.
•Reso(nance) eld: Drag to set the amount of lter resonance for both lters.
•Tap Delay elds: Show the number and name of the selected tap in the upper section and the
delay time in the lower section.
•Pitch On/O button: Click to turn pitch transposition on or o (for the selected tap).
•Transp(ose) elds: The left eld transposes pitch in semitones. The right eld ne-tunes each
semitone step in cents (1/100th of a semitone).
•Flip buttons: Swap the left and right side of the stereo or surround image. Clicking these
buttons reverses the tap position from left to right, or vice versa. For example, if a tap is set to
55% left, clicking the ip button swaps it to 55% right.
•Pan eld: Drag to set pan position for mono signals, stereo balance for stereo signals, and
surround angle when used in surround congurations.
•Pan displays a percentage between 100% (full left) and −100% (full right), which
represents the pan position or balance of the tap. A value of 0% represents the center
panorama position.
•When used in surround, a surround panner replaces the percentage representation. See Use
Delay Designer in surround on page 71.
•Spread eld: Drag to set the width of the stereo spread for the selected tap (in stereo-to-stereo
or stereo-to-surround instances).
•Mute button: Click to mute (silence) or unmute the selected tap.
•Level eld: Drag to set the output level for the selected tap.

Chapter 2 Delay eects 69
Delay Designer sync mode
Delay Designer can either synchronize to the project tempo or can run independently. When you
are in synchronized mode (sync mode), taps snap to a grid of musically relevant positions, based
on note durations. You can also set a Swing value in sync mode, which varies the precise timing
of the grid, resulting in a laid-back, less robotic feel for each tap. When you are not in sync mode,
taps don’t snap to a grid, nor can you apply the Swing value.
When sync mode is on, a grid that matches the chosen Grid parameter value is shown in
the Identication bar. All taps are moved toward the closest delay time value on the grid.
Subsequently created or moved taps are snapped to positions on the grid.
When you save a Delay Designer setting, the sync mode status, Grid, grid position of each tap,
and Swing values are all saved. This ensures that a setting loaded into a project with a dierent
tempo to that of the project the setting was created in retains the relative positions, and rhythm,
of all taps—at the new tempo.
Note: Delay Designer has a maximum delay time of 10 seconds. This means that if you load a
setting into a project with a slower tempo (than the setting’s tempo), some taps may fall outside
the 10 second limit. In such cases, these taps do not play but are retained as part of the setting.
Sync parameters
•Sync button: Turns synchronized mode on or o.
•Grid pop-up menu: Choose a grid resolution from several musical note durations. The grid
resolution, along with the project tempo, determines the length of each grid increment.
As you change grid resolutions, the increments shown in the Identication bar change
accordingly. This also determines a step limitation for all taps.
For example, imagine a project with a tempo of 120 bpm. The Grid pop-up menu value is set
to 1/16 notes. At this tempo and grid resolution, each grid increment is 125 milliseconds (ms)
apart. If Tap A is currently set to 380 ms, turning on sync mode shifts Tap A to 375 ms. If you try
to move Tap A forward in time, it snaps to 500 ms, 625 ms, 750 ms, and so on. At a resolution
of 1/8 notes, the steps are 250 milliseconds apart, so Tap A automatically snaps to the nearest
division (500 ms) and could be moved to 750 ms, 1000 ms, 1250 ms, and so on.

Chapter 2 Delay eects 70
•Swing eld: Drag to determine how close to the absolute grid position every second grid
increment will be.
•A setting of 50% means that every grid increment has the same value.
•Settings below 50% result in every second increment being shorter in time.
•Settings above 50% result in every second grid increment being longer in time.
Tip: Use subtle grid position variations of every second increment (values between 45% and
55%) to create a less rigid rhythmic feel. High Swing values are unsubtle because they place
every second increment directly beside the subsequent increment. Make use of higher values
to create interesting and intricate double rhythms with some taps, while retaining the grid to
lock other taps into more rigid synchronization with the project tempo.
Delay Designer master parameters
The Master section incorporates two global functions: delay feedback and dry/wet mix.
In simple delays, the only way for the delay to repeat is to use feedback. Because Delay Designer
oers 26 taps, you can use these taps to create repeats, rather than requiring discrete feedback
controls for each tap.
Delay Designer’s global Feedback parameter does, however, enable you to send the output of
one user-dened tap back through the eect input, to create a self-sustaining rhythm or pattern.
This tap is known as the feedback tap.
Master parameters
•Feedback button: Turn the feedback tap on or o.
•Feedback Tap pop-up menu: Choose a tap as the feedback tap.
•Feedback Level knob: Rotate to set the feedback tap output level (before it is routed back into
Delay Designer’s input).
•A value of 0% equals no feedback.
•A value of 100% sends the feedback tap back into Delay Designer’s input at full volume.
Note: If Feedback is enabled and you begin creating taps with the Tap pads, Feedback
is automatically turned o. When you stop creating taps with the Tap pads, Feedback is
automatically re-enabled.
•Mix sliders: Drag to independently set the levels of the dry input signal and the post-
processing wet signal.

Chapter 2 Delay eects 71
Use Delay Designer in surround
Delay Designer is optimized for use in surround congurations. With 26 taps that can be
positioned in the surround eld, you can create interesting rhythmic and spatial eects.
Note: Delay Designer generates separate automation data for stereo pan and surround pan
operations. This means that when you use it in surround channels, it does not react to existing
stereo pan automation data, and vice versa.
Delay Designer always processes each input channel independently.
•In a mono/stereo input and surround output conguration, Delay Designer processes the two
stereo channels independently, and the surround panner lets you place each delay around the
surround eld.
•In a surround input and surround output conguration, Delay Designer processes each
surround channel independently and the surround panner lets you adjust the surround
balance of each tap in the surround eld.
When you use Delay Designer in any surround conguration, the Pan parameter on the Tap
parameter bar is replaced with a surround panner, which lets you determine the surround
position of each tap.
Note: In the Tap display’s Pan view, you can adjust only the angle of taps. You must use the
surround panner on the Tap Parameter bar to adjust diversity.
Adjust surround parameters
mTo adjust diversity: Command-drag.
mTo adjust the angle: Command-Option-drag.
mTo reset the angle and diversity: Option-click the blue dot.

Chapter 2 Delay eects 72
Echo
This simple echo eect always synchronizes the delay time to the project tempo, enabling you to
quickly create echo eects that run in time with your composition.
Echo parameters
•Time pop-up menu: Choose 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 eld: Drag to determine how often the delay eect is repeated.
•Color slider and eld: Drag to set the harmonic content (color) of the delay signal.
•Dry and Wet sliders and elds: Drag to set the amount of original and eect signal.
Sample Delay
Sample Delay is more a utility than an eect—you can use it to delay a channel by single
sample values.
When used in conjunction with the phase inversion capabilities of the Gain eect, Sample Delay
is useful for correcting timing problems that may occur with multichannel microphones. It can
also be used creatively to emulate stereo microphone channel separation.
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 emulates an
acoustic (microphone) separation of 10 centimeters.
Sample Delay parameters
•Delay slider and eld (L and R in stereo version): Drag to set the number of samples that the
incoming signal is delayed by.
•Link L & R button (only in stereo version): Turn on to make sure that the number of samples is
identical for both channels. Adjusting one channel value adjusts the other.

Chapter 2 Delay eects 73
Stereo Delay
Stereo Delay lets you set the Delay, Feedback, and Mix parameters separately for the left and
right channels. The Crossfeed knob (for each stereo side) sets the feedback intensity level of each
signal being routed to the opposite stereo side. You can use Stereo Delay on mono tracks or
busses when you want to create independent delays for the two stereo sides.
Note: If you use Stereo Delay 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.
The parameters for the left and right delays are identical. The descriptions below describe the
left channel—the right channel parameter title is provided in parentheses, if named dierently.
Parameters common to both channels are described after the channel parameters.
Channel parameters
•Left (Right) Input pop-up menu: Choose the input signal for the two stereo sides. Options
include O, Left, Right, L + R, and L − R.
•Left (Right) Delay eld: Drag to set the delay time in milliseconds. (The parameter is dimmed
when you synchronize the delay time with the project tempo.)
•Groove slider and eld: Drag to determine the proximity of every second delay repeat to the
absolute grid position—in other words, how close every second delay repeat is.
•Note buttons: Click to 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 eld: Rotate to set the amount of feedback for the left and right
delay signals.
•Crossfeed Left to Right (Crossfeed Right to Left) knob and eld: Rotate to transfer the feedback
signal of the left channel to the right channel, and vice versa.
•Feedback Phase button: Click to invert the phase of the corresponding channel’s
feedback signal.
•Crossfeed Phase button: Click to invert the phase of the crossfed feedback signals.

Chapter 2 Delay eects 74
Common parameters
•Beat Sync button: Turn on to synchronize delay repeats to the project tempo.
•Output Mix (Left and Right) sliders and elds: Drag to independently control the level of the left
and right channel signals.
•Low Cut and High Cut sliders and elds: Drag to cut frequencies below the Low Cut value and
above the High Cut value from the source signal.
Tape Delay
Tape Delay simulates the sound of vintage tape echo machines. It can run at a free rate or can
be synchronized with the project tempo. The eect is equipped with a highpass and lowpass
lter in the feedback loop, making it easy to create authentic dub echo eects. Tape Delay also
includes an LFO for delay time modulation. This can be used to produce chorus eects, even on
long delays.
Tape Delay parameters
•Feedback slider: Drag to set the amount of delayed and ltered signal that is routed back to
the input. Set to the lowest possible value to generate a single echo. Set to 100% to endlessly
repeat the signal. The levels of the original signal and taps (echo repeats) tend to accumulate
and may cause distortion. Use the internal tape saturation circuit to make these overdriven
signals sound pleasant.
•Freeze button: Captures the current delay repeats and sustains them until Freeze is turned o.
•Delay eld: Drag to set the delay time in milliseconds. (This parameter is dimmed when you
synchronize the delay time to the project tempo.)
•Sync button: Click to synchronize delay repeats with the project tempo (including tempo
changes).
•Tempo eld: Drag to set the delay time in beats per minute. (This parameter is dimmed when
you synchronize the delay time to the project tempo.)
•Groove slider and eld: Drag to determine 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.
Chapter 2 Delay eects 75
•Low Cut and High Cut sliders and elds: Drag to cut frequencies below the Low Cut value and
above the High Cut value from the source signal. You can shape the sound of taps (delay
repeats) with the highpass and lowpass lters. The lters are located in the feedback circuit,
which means that the ltering eect increases in intensity with each delay repeat. If you want
an increasingly muddy and confused tone, move the High Cut slider toward the left. For ever
thinner echoes, move the Low Cut slider toward the right. If you can’t hear the eect, check
the Dry and Wet controls and the lter settings.
•Smooth slider and eld: Drag to even out the LFO and utter eect.
•LFO Rate knob and eld: Rotate to set the speed of the LFO.
•LFO Depth knob and eld: Rotate to set the amount of LFO modulation. A value of 0 turns delay
modulation o.
•Flutter Rate and Intensity sliders and elds: Simulate the speed irregularities of the tape
transports used in analog tape delay units.
•Flutter Rate: Drag to set the speed variation.
•Flutter Intensity: Drag to determine the intensity of the eect.
•Dry and Wet sliders and elds: Drag to independently control the amount of original and
eect signal.
•Distortion Level slider and eld (Extended Parameters area): Drag to set the level of the distorted
(tape saturation) signal.

76
Distortion eects overview
Distortion eects simulate the distortion created by vacuum tubes, transistors, or digital circuits.
Vacuum tubes were used in audio ampliers before the development of digital audio technology.
They are still used in musical instrument ampliers today. When overdriven, tubes produce a
musically pleasing distortion that 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 eects 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 tone, or to create sound eects.
Distortion eects include parameters for tone, which let you shape the way the distortion alters
the signal (often as a frequency-based lter), 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 eects can damage your hearing—and
your speakers. When you adjust eect settings, it is recommended that you lower the output
level of the track, and raise the level gradually when you are nished.
Distortion eects 3

Chapter 3 Distortion eects 77
Bitcrusher
Bitcrusher is a low-resolution digital distortion eect. You can use it to emulate the sound of early
digital audio devices, to create articial aliasing by dividing the sample rate, or to distort signals
until they are unrecognizable.
Bitcrusher parameters
•Drive slider and eld: Drag to set the amount of gain applied to the input signal (in decibels).
Note: Raising the Drive level tends to increase the amount of clipping at the output of the
Bitcrusher as well.
•Resolution slider and eld: Drag to set 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 slider and eld: Drag to reduce the sample rate. A value of 1x has no eect on
the signal, a value of 2x halves the sample rate, and a value of 10x reduces the sample rate to
one-tenth of the original. (For example, if you set Downsampling to 10x, 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 signicant impact on the behavior of all three modes.
This is reected in the Waveform display, so try each mode button and adjust the Clip Level
slider to get a feel for how this works.
•Folded button: The center portion of the signal is halved in level above the threshold,
resulting in a softer distortion. The start and end levels of the clipped signal are unchanged.
•Cut button: Causes abrupt distortion when the clipping threshold is exceeded. Clipping that
occurs in most digital systems is closest to Cut mode.
•Displaced button: The start, mid, and end levels of the signal (above the threshold) are oset,
resulting in less severe distortion as signal levels cross the threshold. The center portion of
the clipped signal is also softer than in Cut mode.
•Clip Level slider and eld: Drag to set the point (below the clipping threshold of the channel) at
which the signal starts clipping.
•Mix slider and eld (Extended Parameters area): Drag to set the balance between dry (original)
and wet (eect) signals.

Chapter 3 Distortion eects 78
Clip Distortion
Clip Distortion is a nonlinear distortion eect that produces unpredictable spectra. It can
simulate warm, overdriven tube sounds and can also generate heavy distortions.
Clip Distortion has an unusual combination of serially connected lters. The incoming signal
is amplied by the Drive value, passes through a highpass lter, then is subjected to nonlinear
distortion. Following the distortion, the signal passes through a lowpass lter. The eect signal is
then recombined with the original signal and this mixed signal is sent through a further lowpass
lter. All three lters have a slope of 6 dB/octave.
This unique combination of lters allows for gaps in the frequency spectra that can sound good
with this sort of nonlinear distortion.
Clip Distortion parameters
•Drive slider and eld: Drag to set the amount of gain applied to the input signal. After being
amplied by the Drive value, the signal passes through a highpass lter.
•Tone slider and eld: Drag to set the cuto frequency (in hertz) of the highpass lter.
•Clip Circuit display: Shows the impact of all parameters, with the exception of the High Shelving
lter parameters.
•Symmetry slider and eld: Drag to set the amount of nonlinear (asymmetrical) distortion
applied to the signal.
•Clip Filter slider and eld: Drag to set the cuto frequency (in hertz) of the rst lowpass lter.
•Mix slider and eld: Drag to set the ratio between the eect (wet) signal and original (dry)
signals, following the Clip Filter.
•Sum LPF knob and eld: Drag to set the cuto frequency (in hertz) of the lowpass lter. This
processes the mixed signal.
•(High Shelving) Frequency knob and eld: Rotate to set the frequency (in hertz) of the high
shelving lter. 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- amplier. 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 and eld: Rotate to set the amount of gain applied to the output of
signals above the high shelving lter frequency.
•Input Gain eld and slider (Extended Parameters area): Drag to set the amount of gain applied to
the input signal.
•Output Gain eld and slider (Extended Parameters area): Drag to set the amount of gain applied
to the output signal.

Chapter 3 Distortion eects 79
Distortion eect
The Distortion eect simulates the low delity distortion generated by a bipolar transistor. You
can use it to simulate playing a musical instrument through a highly overdriven amplier or to
create unique distorted sounds.
Distortion parameters
•Drive slider and eld: Drag to set the amount of saturation applied to the signal.
•Display: Shows the impact of parameters on the signal.
•Tone knob and eld: Rotate to set the frequency for the high cut lter. Filtering the harmonically
rich distorted signal produces a softer tone.
•Output slider and eld: Drag to set the output level. This enables you to compensate for
increases in loudness caused by adding distortion.
•Level Compensation checkbox (Extended parameter): Turn on to reference the overall processing
of the signal to 0 dB, making the output louder.
Distortion II
Distortion II emulates the distortion circuit of a Hammond B3 organ. You can use it on musical
instruments to recreate this classic eect or can use it creatively when designing new sounds.
Distortion II parameters
•PreGain knob: Rotate to set the amount of gain applied to the input signal.
•Drive knob: Rotate to set the amount of saturation applied to the signal.
•Tone knob: Rotate to set the frequency of the highpass lter. Filtering the harmonically rich
distorted signal produces a softer tone.
•Type pop-up menu: Choose the type of distortion.
•Growl: Emulates a two-stage tube amplier 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.

Chapter 3 Distortion eects 80
Overdrive
Overdrive emulates the distortion produced by a eld eect transistor (FET), commonly used
in solid-state musical instrument ampliers and hardware eects devices. When saturated, FETs
generate a warmer-sounding distortion than bipolar transistors, such as those emulated by the
Distortion eect.
Overdrive parameters
•Drive slider and eld: Drag to set the saturation amount for the simulated transistor.
•Display: Shows the impact of parameters on the signal.
•Tone knob and eld: Rotate to set the frequency for the high cut lter. Filtering the harmonically
rich distorted signal produces a softer tone.
•Output slider and eld: Drag to set the output level. This enables you to compensate for
increases in loudness caused by using Overdrive.

Chapter 3 Distortion eects 81
Phase Distortion
The Phase Distortion eect is based on a modulated delay line, similar to a chorus or anger
eect (see Modulation eects overview on page 19 9). Unlike these eects, however, the delay
time is not modulated by a low frequency oscillator (LFO) but rather by a lowpass-ltered
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 aected by any other process. The Mix
parameter blends the eect signal with the original signal.
Phase Distortion parameters
•Monitor button: Turn on to hear the input signal in isolation. Turn o to hear the mixed signal.
•Cuto knob and eld: Rotate to set the (center) cuto frequency of the lowpass lter.
•Resonance knob and eld: Rotate to emphasize frequencies surrounding the cuto frequency.
•Display: Shows the impact of parameters on the signal.
•Mix slider and eld: Drag to set the percentage of the eect signal mixed with the
original signal.
•Max Modulation slider and eld: Drag to set the maximum delay time.
•Intensity slider and eld: Drag to set the amount of modulation applied to the signal.
•Phase Reverse checkbox (Extended Parameters area): Select to reduce the delay time on the right
channel when input signals that exceed the cuto frequency are received. Available only for
stereo instances of the Phase Distortion eect.
82
Dynamics processors overview
Dynamics processors control the perceived loudness of your audio, add focus and punch to
tracks and projects, and optimize the sound for playback in dierent 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 les, tracks, or an overall project. This can be
to increase the perceived loudness or to highlight the most important sounds, while ensuring
that softer sounds are not lost in the mix.
There are four types of dynamics processors. These are each used for dierent audio
processing tasks.
•Compressors: Downward compressors behave like an automatic volume control, lowering the
volume whenever it rises above a certain level, called the threshold.
By reducing the highest parts of the signal, called peaks, a compressor 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 dierent audio
environments. For example, the speakers of a television 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-delity 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 eect tracks, but they are rarely used
on ambience tracks.
Some compressors—multiband compressors—can divide the incoming signal into dierent
frequency bands and apply dierent 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 dierence 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.
Dynamics processors 4

Chapter 4 Dynamics processors 83
•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 o. Noise gates are often used to eliminate low-level noise or
hum from an audio signal.
Adaptive Limiter
Adaptive Limiter is a versatile tool for controlling the perceived loudness of sounds. It works by
rounding and smoothing peaks in the signal, producing an eect similar to an analog amplier
being driven hard. Like an amplier, it can slightly color the sound of the signal. You can use
Adaptive Limiter to achieve maximum gain, without introducing unwanted distortion and
clipping, which can occur when the signal exceeds 0 dBFS.
Adaptive Limiter is typically used on the nal mix, where it can be placed after a compressor,
such as Multipressor, and before a nal gain control, resulting in a mix of maximum loudness.
Adaptive Limiter can produce a louder-sounding mix than can be achieved by normalizing
the signal.
Note: Using Adaptive Limiter adds latency when the Lookahead parameter is active. The eect
is typically used for mixing and mastering previously recorded tracks, not while recording. You
should bypass Adaptive Limiter while recording.
Adaptive Limiter parameters
•Input meters: Show input levels in real time. The Margin eld shows the highest input level. You
can reset the Margin eld by clicking it.
•Input Scale knob and eld: Rotate to scale the input level. Scaling is useful for handling very
high-level or low-level input signals. It “squeezes” the higher and lower signal levels into a
range that allows the Gain knob to work eectively. Avoid input levels above 0 dBFS, which
can result in unwanted distortion.
•Gain knob and eld: Rotate to set the amount of gain after input scaling.
•Out Ceiling knob and eld: Rotate to set the maximum output level, or ceiling. The signal will
not rise above this.

Chapter 4 Dynamics processors 84
•Output meters: Show output levels, enabling you to see the results of the limiting process. The
Margin eld shows the highest output level. You can reset the Margin eld by clicking it.
•Mode buttons (Extended Parameters area): Click to choose the type of peak smoothing:
•OptFit: Limiting follows a linear curve, which allows signal peaks above 0 dB.
•NoOver: Avoids distortion artifacts from the output hardware by ensuring that the signal
does not exceed 0 dB.
•Lookahead eld and slider (Extended Parameters area): Drag to adjust the playback buer size
(how far in the future the le is analyzed for peaks).
•Remove DC checkbox (Extended Parameters area): Select to activate a highpass lter that
removes direct current (DC) from the signal. DC can be introduced by lower-quality
audio hardware.
•Inter Sample Peak Detection checkbox (Extended Parameters area): Select to detect inter-sample
peaks in the signal.
Compressor
Compressor overview
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 Compressor with individual tracks, including vocal, instrumental, and eects tracks,
as well as on the overall mix. Usually you insert the Compressor directly into a channel strip.
Chapter 4 Dynamics processors 85
Compressor parameters
•Circuit Type pop-up menu: Choose the type of circuit emulated by Compressor. The choices are
Platinum, Studio or Vintage VCA or FET, and Vintage Opto.
•Side Chain Detection pop-up menu: Choose the signal type to exceed or fall below the
threshold. Max uses the maximum level of each side-chained signal. Sum uses the summed
level of all side-chained signals.
•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 eld: Rotate to set the time it takes for Compressor to react when the signal
exceeds the threshold.
•Compression curve display: Shows the compression curve created by the combination of Ratio
and Knee parameter values. The input (level) is shown on the x-axis and output (level) on the
y-axis.
•Release knob and eld: Rotate to set the time it takes for Compressor to stop reducing the
signal after the signal level falls below the threshold.
•Auto button: Turn on to make the release time dynamically adjust to the audio material.
•Ratio slider and eld: Drag to set the compression ratio—the ratio of signal reduction when the
threshold is exceeded.
•Knee slider and eld: Drag to set 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 eld: Drag to set the threshold level—signals above this
threshold value are reduced in level.
•Peak/RMS buttons: Click to determine if signal analysis uses the Peak or RMS method when
using the Platinum circuit type.
•Gain slider and eld: Drag to set 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 O, 0 dB, and −12 dB.
•Limiter Threshold slider and eld: Drag to set the threshold level for the limiter.
•Limiter button: Turns the integrated limiter on or o.
•Output Distortion pop-up menu (Extended Parameters area): Choose whether to apply clipping
above 0 dB, and the type of clipping. The choices are O, Soft, Hard, and Clip.
•Activity pop-up menu (Extended Parameters area): Turns the side chain on or o.
•Mode pop-up menu (Extended Parameters area): Choose the type of lter used for the side
chain. The choices are LP (lowpass), BP (bandpass), HP (highpass), ParEQ (parametric), and HS
(high shelving).
•Frequency slider and eld (Extended Parameters area): Drag to set the center frequency for the
side-chain lter.
•Q slider and eld (Extended Parameters area): Drag to set the width of the frequency band
aected by the side-chain lter.
•Gain slider and eld (Extended Parameters area): Drag to set the amount of gain applied to the
side-chain signal.
•Mix slider and eld (Extended Parameters area): Drag to set the balance between dry (source)
and wet (eect) signals.
Chapter 4 Dynamics processors 86
Use Compressor
The following explains how to eectively use the main Compressor parameters.
Compressor Threshold and Ratio
The most important Compressor parameters are Threshold and Ratio. The Threshold parameter
sets the oor level in decibels. Signals that exceed this level are reduced by the amount set as
the Ratio.
The Ratio parameter 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.
For 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.
Compressor envelope times
The Attack and Release parameters shape the dynamic response of Compressor. The Attack
parameter determines the time it takes after the signal exceeds the threshold level before
Compressor starts reducing the signal.
Many sounds, including voices and musical instruments, rely on the initial attack phase to dene
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 dierences in
the signal. Set lower Release values if you want to emphasize dynamic dierences.
Important: The results of your settings for the Attack and Release parameters depend not only
on the type of source material but on the compression ratio and threshold settings.
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.
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.
Chapter 4 Dynamics processors 87
Other Compressor parameters
As 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, Compressor can analyze the signal using one of two
methods: Peak or root mean square (RMS). While Peak is technically more accurate, RMS provides
a better indication of how people perceive the signal loudness.
Note: If you turn on Auto Gain and RMS simultaneously, the signal may become oversaturated.
If you hear any distortion, turn o Auto Gain and adjust the Gain slider until the distortion
is inaudible.
Use a side chain with Compressor
Use of a side chain with a compressor is common. The dynamics (level changes) of another
channel strip is used 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.
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.
1 Insert Compressor into a channel strip.
2 In the Compressor plug-in window header, choose the channel strip that carries the desired
signal (side-chain source) from the Side Chain pop-up menu.
3 Choose the desired analysis method (Max or Sum) from the Side Chain Detection pop-up menu.
4 Adjust Compressor parameters.
DeEsser
DeEsser is a frequency-specic 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.
The advantage of using 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. DeEsser has extremely fast attack and release times.
When using DeEsser, you can set the frequency range being compressed (the Suppressor
frequency) independently of the frequency range being analyzed (the Detector frequency). The
two ranges can be compared in 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.

Chapter 4 Dynamics processors 88
DeEsser does not use a frequency-dividing network—a crossover utilizing lowpass and highpass
lters. 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 DeEsser’s interface, and the Suppressor
parameters are on the right. The center section includes the Detector and Suppressor displays
and the Smoothing slider.
DeEsser Detector parameters
•Detector Frequency knob and eld: Rotate to set the frequency range for analysis.
•Detector Sensitivity knob and eld: Rotate to set the degree of responsiveness to the
input signal.
•Monitor pop-up menu: Choose the signal type that you want to monitor. Choose Det(ector)
to monitor the isolated Detector signal, Sup(pressor) to monitor the ltered Suppressor
signal, Sens(itivity) to remove the sound from the input signal in response to the Sensitivity
parameter, or O to hear the DeEsser output.
DeEsser Suppressor parameters
•Suppressor Frequency knob and eld: Rotate to set the frequency band that is reduced when the
Detector sensitivity threshold is exceeded.
•Strength knob and eld: Rotate to set the amount of gain reduction for signals that surround
the Suppressor frequency.
•Activity LED: Indicates active suppression in real time.
DeEsser common parameters
•Detector and Suppressor frequency displays: The upper display shows the Detector frequency
range. The lower display shows the Suppressor frequency range (in hertz).
•Smoothing slider: Drag to set the reaction speed for the gain reduction start and end phases.
Smoothing controls both the attack and release times, as they are used by compressors.

Chapter 4 Dynamics processors 89
Use 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 nished, 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 be inserted only in output and aux channel strips.
Ducker parameters
•Ducking O/On buttons: Turn ducking on or o.
•Lookahead O/On buttons: Turn on to make sure that Ducker reads the incoming signal before
processing. This results in no latency—it is primarily intended for slower computers.
•Amount slider and eld: Drag to set the amount of volume reduction of the music mix channel
strip—in eect, the output signal.
•Threshold slider and eld: Drag to set 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 Amount slider. If
the side-chain signal level doesn’t reach the threshold, the music mix channel strip volume is
not aected.
•Attack slider and eld: Drag to control 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. The Attack 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. Logic Pro needs to analyze the signal level before it is played back to predene the
point where ducking begins.
•Hold slider and eld: Drag to dene the length of time that the music mix channel strip volume
is reduced. This control prevents a chattering eect 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.

Chapter 4 Dynamics processors 90
•Release slider and eld: Drag to control 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.
Use the Ducker plug-in
1 Insert Ducker 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 In the Ducker plug-in window header, choose the bus that carries the ducking (vocal) signal from
the Side Chain pop-up menu.
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 voiceover—is heard at the output.
4 Adjust the Ducker parameters.
Enveloper
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 dier from other dynamics processors. In contrast to a compressor or expander,
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.
The most important Enveloper parameters are the two Gain sliders, one on each side of the
central display. These govern the Attack and Release levels of each respective phase.
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
eect 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 aected channel strip.
Conversely, attenuating the release phase makes reverb-drenched tracks sound drier. This is
particularly useful when you are working with drum loops, but it has many other applications
as well.
Chapter 4 Dynamics processors 91
Enveloper parameters
•Threshold slider and eld: Drag to set the threshold level. Signals that exceed the threshold
have their attack and release phase levels altered. In general, you should set the Threshold
to the minimum value and leave it there. Only when you signicantly raise the release phase
level, which also boosts any noise in the original recording, should you raise the Threshold
slider slightly. This limits Enveloper to aecting only the useful part of the signal.
•(Attack) Gain slider and eld: Drag to boost or attenuate the attack phase of the signal. When
the Gain slider is set to the center position—0%—the signal is unaected.
•Lookahead slider and eld: Drag to set the pre-read analysis time for the incoming signal. The
Lookahead slider denes how far into the future of the incoming signal Enveloper looks,
to anticipate upcoming events. You generally do not 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.
•Display: Shows the attack and release curves applied to the signal.
•(Attack) Time knob and eld: Rotate to set the time it takes for the signal to increase from the
threshold level to the maximum Gain level. Attack Time values of around 20 ms and Release
Time values of 1500 ms are a good starting point.
•(Release) Time knob and eld: Rotate to set the time it takes for the signal to fall from the
maximum gain level to the threshold level.
•(Release) Gain slider and eld: Drag to boost or attenuate the release phase of the signal. When
the Gain slider is set to the center position—0%—the signal is unaected.
•Out Level slider and eld: Drag to set the level of the output 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.

Chapter 4 Dynamics processors 92
Expander
Expander is similar in concept to a compressor, but increases, rather than reduces, the dynamic
range above the threshold level. You can use Expander to add liveliness and freshness to your
audio signals.
Expander parameters
•Threshold slider and eld: Drag to set the threshold level. Signals above this level are expanded.
•Peak/RMS buttons: Click to determine whether the Peak or RMS method is used to analyze
the signal.
•Attack knob and eld: Rotate to set the time it takes for Expander to respond to signals that
exceed the threshold level.
•Expansion display: Shows the expansion curve applied to the signal.
•Release knob and eld: Rotate to set the time it takes for Expander to stop processing the signal
after it falls below the threshold level.
•Ratio slider and eld: Drag to set the expansion ratio—the ratio of signal expansion when the
threshold is exceeded.
Note: Because 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 eld: Drag to determine 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 eld: Drag to set the amount of output gain.
•Auto Gain button: Turn on to compensate 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 adjusts the signal appropriately.

Chapter 4 Dynamics processors 93
Limiter
Limiter works much like a compressor but with one important dierence: where a compressor
proportionally reduces the signal when it exceeds the threshold, a limiter reduces any peak
above the threshold to the threshold level, eectively limiting the signal to this level.
Limiter is used primarily when mastering. Typically, you apply 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.
Limiter is designed in such a way that if set to 0 dB Gain and 0 dB Output Level, it has no eect
on a normalized signal. If the signal clips, Limiter reduces the level before clipping can occur.
Limiter cannot, however, x audio that is clipped during recording.
Limiter parameters
•Gain reduction meter: Shows the amount of limiting in real time.
•Gain slider and eld: Drag to set the amount of gain applied to the input signal.
•Lookahead slider and eld: Drag to adjust how far ahead (in milliseconds) 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 eect when you use Limiter as a
mastering eect on prerecorded material. Set it to higher values if you want the limiting eect
to occur before the maximum level is reached, thus creating a smoother transition.
•Release slider and eld: Drag to set the time it takes for Limiter to stop processing, after the
signal falls below the threshold level.
•Output Level knob and eld: Rotate to set the output level of the signal.
•Softknee button: Turn on to limit the signal only when it reaches the threshold. The transition to
full limiting is nonlinear, producing a softer, less abrupt eect, and reducing distortion artifacts
that can be produced by hard limiting.
•Inter Sample Peak Detection checkbox (Extended Parameters area): Select to detect inter-sample
peaks in the signal.

Chapter 4 Dynamics processors 94
Multipressor
Multipressor overview
Multipressor (an abbreviation for multiband compressor) is a versatile audio mastering tool. It
splits the incoming signal into dierent frequency bands—up to four—and enables you to
independently compress each band. After compression is applied, the bands are combined into a
single output signal.
The advantage of compressing dierent frequency bands separately is that it allows more
compression to be applied to bands that need it, without aecting other bands. This avoids the
“pumping” eect often associated with high amounts of compression.
Because the use of higher compression ratios on specic frequency bands is possible,
Multipressor can achieve a higher average volume without causing audible artifacts.
Raising the overall volume level can result in a corresponding increase in the existing noise oor.
Each frequency band features downward expansion, which enables you to reduce or suppress
this noise.
Downward expansion works as a counterpart to compression. Whereas a compressor compresses
the dynamic range of higher volume levels, the downward expander expands the dynamic range
of the lower volume levels. With downward expansion, the signal is reduced in level when it falls
below the threshold level. This works in a similar way to a noise gate, but rather than abruptly
cutting o the sound, it smoothly fades the volume with an adjustable ratio.
Multipressor Display parameters
Multipressor’s parameters are grouped into three main areas: the graphic display in the upper
section, the set of controls for each frequency band in the lower section, and the output
parameters on the right.
Frequency band section
Graphic display section
Output section

Chapter 4 Dynamics processors 95
Display parameters
•Graphic display: Shows and allows adjustment of frequency and gain for each frequency band.
The amount of gain change from 0 dB is indicated by blue bars. The band number appears in
the center of active bands. You can adjust each frequency band in the following ways:
•Drag the horizontal bar up or down to adjust the gain makeup for that band.
•Drag the vertical edges of a band to the left or right to set the crossover frequencies, which
adjusts the band’s frequency range.
•Crossover elds: Drag to set the crossover frequency between adjacent bands.
•Gain Make-up elds: Drag to set the amount of the gain make-up for each band.
Multipressor Frequency Band parameters
Multipressor’s parameters are grouped into three main areas: the graphic display in the upper
section, the set of controls for each frequency band in the lower section, and the output
parameters on the right.
Frequency band section
Graphic display section
Output section
Frequency band parameters
•Compr Thresh (Compression Threshold) elds: Drag to set the compression threshold for the
selected band. Setting the parameter to 0 dB results in no compression of the band.
•Compr Ratio (Compression Ratio) elds: Drag to set the compression ratio for the selected band.
Setting the parameter to 1:1 results in no compression of the band.
•Expnd Thresh (Expand Threshold) elds: Drag to set the expansion threshold for the selected
band. Setting the parameter to its minimum value (−60 dB) means that only signals that fall
below this level are expanded.
•Expnd Ratio (Expand Ratio) elds: Drag to set the expansion ratio for the selected band.
•Expnd Reduction (Expand Reduction) elds: Drag to set the amount of downward expansion for
the selected band.
•Peak/RMS elds: Drag to set a smaller value for shorter peak detection, or a larger value for RMS
detection, in milliseconds.

Chapter 4 Dynamics processors 96
•Attack elds: Drag to set the time before compression starts for the selected band, after the
signal exceeds the threshold.
•Release elds: Drag to set the time before compression stops on the selected band, after the
signal falls below the threshold.
•Band on/o buttons: Turn each band on or o. When enabled, the button is highlighted, and
the corresponding band appears in the graphic display area above.
•Byp(ass) buttons: Turn on to bypass the selected frequency band.
•Solo buttons: Turn on to hear compression on only the selected frequency band.
•Level meters: The bar on the left shows the input level, and the bar on the right shows the
output level.
•Threshold arrows: Two arrows appear to the left of each Level meter.
•Click the upper arrow to adjust the Compression Threshold (Compr Thresh).
•Click the lower arrow to adjust the Expansion Threshold (Expnd Thresh).
Multipressor Output parameters
Multipressor’s parameters are grouped into three main areas: the graphic display in the upper
section, the set of controls for each frequency band in the lower section, and the output
parameters on the right.
Frequency band section
Graphic display section
Output section
Output parameters
•Auto Gain button: Turn on to reference the overall processing of the signal to 0 dB, making the
output louder.
•Lookahead value eld: Drag to set how far ahead the eect analyzes the incoming signal,
allowing faster reactions to peak volumes.
•Out slider: Drag to set the overall gain at Multipressor’s output.
•Level meter: Shows the overall output level.
Chapter 4 Dynamics processors 97
Use Multipressor
In the graphic display, the blue bars show the gain change—not merely the gain reduction—as
with a standard compressor. The gain change display is a composite value consisting of the
compression reduction, plus the expander reduction, plus the auto gain compensation, plus the
gain make-up.
Compression parameters
The Compression Threshold and Compression Ratio parameters are the key parameters for
controlling compression. Usually the most useful combinations of these two settings are a low
Compression Threshold with a low Compression Ratio, or a high Compression Threshold with a
high Compression Ratio.
Downward Expansion parameters
The Expansion Threshold, Expansion Ratio, and Expansion Reduction parameters are the key
parameters for controlling downward expansion. They determine the strength of the expansion
applied to the chosen range.
Peak/RMS and Envelope parameters
Adjusting the parameter between Peak (0 ms, minimum value) and RMS (root mean square
−200 ms, maximum value) is dependent on the type of signal you want to compress. An
extremely short Peak detection setting is suitable for compression of short and high peaks of low
power, which do not typically occur in music. The RMS detection method measures the power
of the audio material over time and thus works much more musically. This is because human
hearing is more responsive to the overall power of the signal than to single peaks. As a basic
setting for most applications, the centered position is recommended.
Output parameters
The Out slider sets the overall output level. Set Lookahead to higher values when the Peak/RMS
elds are set to higher values (farther towards RMS). Set Auto Gain to On to reference the overall
processing to 0 dB, making the output louder.

Chapter 4 Dynamics processors 98
Noise Gate
Noise Gate overview
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.
Noise Gate works by allowing signals above the threshold level to pass unimpeded, while
reducing signals below the threshold level. This eectively removes lower-level parts of the
signal, while allowing the desired parts of the audio to pass.
Noise Gate parameters
•Threshold slider and eld: Drag to set the threshold level. Signals that fall below the threshold
are reduced in level.
•Reduction slider and eld: Drag to set the amount of signal reduction.
•Attack knob and eld: Rotate to set the time it takes to fully open the gate after the signal
exceeds the threshold.
•Hold knob and eld: Rotate to set the time the gate is kept open after the signal falls below
the threshold.
•Release knob and eld: Rotate to set the time it takes to reach maximum attenuation after the
signal falls below the threshold.
•Hysteresis slider and eld: Drag to set the dierence (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 level is close to the threshold level.
•Lookahead slider and eld: Drag to set how far ahead Noise Gate analyzes the incoming signal,
allowing the eect to respond more quickly to peak levels.
•Monitor button: Turn on to hear the side-chain signal, including the eect of the High Cut and
Low Cut lters.
•High Cut slider and eld: Drag to set the upper cuto frequency for the side-chain signal.
•Low Cut slider and eld: Drag to set the lower cuto frequency for the side-chain signal.
Note: When no external side chain is selected, the input signal is used as the side chain.
Chapter 4 Dynamics processors 99
Use 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 eects.
The Attack, Hold, and Release knobs modify the dynamic response of 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
dene a minimum Hold time. Use it to set the range between the threshold values that open
and close the gate. This is useful when the signal level hovers around the Threshold level, causing
Noise Gate to switch on and o repeatedly, and producing the undesirable chattering eect. The
Hysteresis slider essentially sets the gate to open at the Threshold level and remain open until
the level drops below another, lower, level. As long as the dierence between these two values
is large enough to accommodate the uctuating level of the incoming signal, 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 nd that the level of the signal you want to keep and the level of the
noise signal are close, making it dicult to separate them. For example, when you are recording
a drum kit and using 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 lters.
Important: The side-chain signal is used only as a detector/trigger in this situation. The lters are
used to isolate particular trigger signals in the side-chain source, but they have no inuence on
the actual gated signal—the audio being routed through Noise Gate.
Use the side-chain lters
1 Click the Monitor button to hear how the High Cut and Low Cut lters will aect the incoming
trigger signal.
2 Drag the High Cut slider to set the upper frequency.
Trigger signals above this are ltered.
3 Drag the Low Cut slider to set the lower frequency.
Trigger signals below this are ltered.
The lters allow only very high (loud) signal peaks to pass. In the drum kit example above, you
could remove the hi-hat signal, which is higher in frequency, with the High Cut lter and allow
the snare signal to pass. Turn o monitoring to set a suitable Threshold level more easily.

Chapter 4 Dynamics processors 100
Surround Compressor
Surround Compressor overview
Surround Compressor, based on the Compressor plug-in, is specically designed for compression
of complete surround mixes. It is commonly inserted in a surround output channel strip or in
audio or aux channel strips—busses—that carry multichannel audio.
You can adjust the compression ratio, knee, attack, and release for the main, side, surround, and
LFE channels, depending on the chosen surround format. All channels include an integrated
limiter and provide independent threshold and output level controls.
You can link channels by assigning them to one of three groups. When you adjust the threshold
or output parameter of any grouped channel, the parameter adjustment is mirrored by all
channels assigned to the group.
LFE sectionMain section
Link section
Surround Compressor’s interface is divided into three sections:
•The Link section at the top contains menus used to assign each channel to a group. See
Surround Compressor Link parameters on page 101.
•The Main section includes controls common to all the main channels and the threshold and
output controls for each channel. See Surround Compressor Main parameters on page 102.
•The LFE section on the lower right includes separate controls for the LFE channel. See
Surround Compressor LFE parameters on page 103.

Chapter 4 Dynamics processors 101
Surround Compressor Link parameters
Surround Compressor’s Link section provides the following parameters.
Link parameters
•Circuit Type pop-up menu: Choose the type of circuit emulated by Surround Compressor. The
choices are Platinum, Classic A_R, Classic A_U, VCA, FET, and Opto (optical).
•Grp. (Group) pop-up menus: Set group membership for each channel (A, B, C, or no group
(indicated by -). Moving the Threshold or Output Level slider for any grouped channel moves
the sliders for all channels assigned to that group.
Tip: Press Command and Option while moving the Threshold or Output Level slider of
a grouped channel to temporarily unlink the channel from the group. This lets you set
independent threshold settings while maintaining the side-chain detection link necessary for
a stable surround image.
•Byp (Bypass) buttons: Click to bypass the channel. If the channel belongs to a group, all
grouped channels are bypassed.
•Detection pop-up menu: Choose the signal type to exceed or fall below the threshold. Max uses
the maximum level of each signal. Sum uses the summed level of all signals.
•If Max is chosen and any of the surround channels exceeds or falls below the threshold, that
channel (or group of channels) is compressed.
•If Sum is chosen, the combined level of all channels must exceed the threshold before
compression occurs.

Chapter 4 Dynamics processors 102
Surround Compressor Main parameters
Surround Compressor’s Main section provides the following parameters.
Main parameters
•Ratio knob and eld: Rotate to set the ratio of signal reduction when the threshold is exceeded.
•Knee knob and eld: Rotate to set the ratio of compression at levels close to the threshold.
•Attack knob and eld: Rotate to set the time it takes to reach full compression, after the signal
exceeds the threshold.
•Release knob and eld: Rotate to set the time it takes to return to zero compression, after the
signal falls below the threshold.
•Auto button: Turn on to dynamically adjust the release time to the audio material.
•Limiter button: Turns limiting for the main channels on or o.
•Threshold knob and eld: Rotate to set the threshold for the limiter on the main channels.
•Main Compressor Threshold sliders and elds: Drag to set the threshold level for each channel—
including the LFE channel, which also has independent controls.
•Main Output Levels sliders and elds: Drag to set the output level for each channel—including
the LFE channel, which has independent controls.

Chapter 4 Dynamics processors 103
Surround Compressor LFE parameters
Surround Compressor’s LFE section provides the following parameters.
LFE parameters
•Ratio knob and eld: Rotate to set the compression ratio for the LFE channel.
•Knee knob and eld: Rotate to set the knee for the LFE channel.
•Attack knob and eld: Rotate to set the attack time for the LFE channel.
•Release knob and eld: Rotate to set the release time for the LFE channel.
•Auto button: Turn on to automatically adjust the release time to the audio signal.
•Threshold knob and eld: Rotate to set the threshold for the limiter on the LFE channel.
•Limiter button: Turns limiting on or o for the LFE channel.
104
Equalizers overview
An equalizer (commonly abbreviated as EQ) shapes the sound of incoming audio by changing
the level of specic frequency bands.
Equalization is one of the most-used audio processes, both for music projects and in post-
production work for video. You can use EQ to subtly or signicantly shape the sound of an
audio le, an instrument, a vocal performance, or a project by adjusting specic frequencies or
frequency ranges.
All EQs are specialized lters that allow certain frequencies to pass through unchanged while
raising (boosting) or lowering (cutting) the level of other frequencies. Some EQs can be used
in a “broad-brush” fashion, to boost or cut a large range of 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 Channel EQ or Linear Phase EQ) combine several lters 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 of any audio source, be it an individual audio
signal or an entire mix.
Channel EQ
Channel EQ overview
Channel EQ is a versatile multiband EQ. It provides eight frequency bands, including lowpass and
highpass lters, low and high shelving lters, and four exible parametric bands. It also features
an integrated Fast Fourier Transform (FFT) Analyzer that shows changes to the frequency curve
of the audio signal in real time, allowing you to see which parts of the frequency spectrum may
need adjustment.
You can use Channel EQ to shape the sound of individual tracks or audio les or for tone-shaping
on an overall project mix. The Analyzer and graphic display’s controls make it easy to view and
change the audio signal in real time.
Tip: The parameters of Channel EQ and Linear Phase EQ are identical, enabling you to freely
copy settings between them. In Logic Pro X, if you replace a Channel EQ with a Linear Phase EQ
(or vice versa) in the same Insert slot, the current settings are automatically transferred to the
new EQ.
Equalizers 5

Chapter 5 Equalizers 105
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 parameters
•Master Gain slider and eld: Drag to set the overall output level of the signal. Use it after
boosting or cutting individual frequency bands.
•Analyzer button: Turns the Analyzer on or o.
•Pre/Post EQ button: Determines whether the Analyzer shows the frequency curve before or
after EQ is applied, when Analyzer mode is active.
•Resolution pop-up menu: Choose the sample resolution for the Analyzer. Choose from the
following menu items: low (1024 points), medium (2048 points), and high (4096 points).
•Band On/O buttons: Turn the corresponding band on or o. Each button’s icon indicates the
lter type:
•Band 1 is a highpass lter.
•Band 2 is a low shelving lter.
•Bands 3 through 6 are parametric bell lters.
•Band 7 is a high shelving lter.
•Band 8 is a lowpass lter.
•Graphic display: Shows the current curve of each EQ band. The scale is shown in dB.
•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 reects your changes immediately.
•Drag the pivot point in each band to adjust the Q factor. Q is shown beside the pointer
when it is moved over a pivot point.
•Frequency elds: Drag to adjust the frequency of each band.
•Gain/Slope elds: Drag to set the amount of gain for each band. For bands 1 and 8, this
changes the slope of the lter.
•Q elds: Drag to adjust the Q factor or resonance for each band—the range of frequencies
around the center frequency that are aected.
Chapter 5 Equalizers 106
Note: The Q parameter of band 1 and band 8 has no eect when the slope is set to 6 dB/Oct.
When the Q parameter is set to an extremely high value, such as 100, these lters aect only a
very narrow frequency band and can be used as notch lters.
•Link button: Turns on 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): Click to choose Peak or RMS.
•Analyzer Decay slider and eld (Extended Parameters area): Drag to set 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.
•Strong: Preserves most of the perceived bandwidth.
•Light or medium: Allows some change as you raise or lower the gain.
•Asymmetric: These 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 dierent Gain-Q Couple Strength
setting, the actual Q values will be dierent than when the automation was recorded.
Channel EQ use tips
The way you use Channel EQ depends on the audio material and your intended outcome. A
useful workow for many situations is as follows: Set the Channel EQ to a at response (no
frequencies boosted or cut), turn on the Analyzer, and play the audio signal. Watch 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 attention to sections where the signal distorts or clips. Use
the graphic display or parameter controls to adjust the frequency bands.
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
aect a specic 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 or parameters so that only a narrow range of frequencies is aected, or widen it to
alter a broader frequency area.
Each EQ band has a dierent 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 area around the zero line, drag the dB
scale, on the left side of the graphic display upward. Drag downward to decrease the resolution.
Chapter 5 Equalizers 107
Channel EQ Analyzer
The Analyzer uses 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 simplies the task of setting
EQ curves to raise or lower the levels of frequencies and frequency ranges.
The bands derived from FFT analysis are scaled logarithmically—there are more bands in higher
octaves than in lower octaves.
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: High Analyzer resolutions require signicantly 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 turn o the Analyzer or close the Channel EQ window after setting
EQ parameters.
Linear Phase EQ
Linear Phase EQ overview
The high-quality Linear Phase EQ eect is similar to Channel EQ, sharing the same parameters
and eight-band layout. You can copy settings between them. In Logic Pro, if you replace
Channel EQ with Linear Phase EQ (or vice versa) in the same Insert slot, the current settings are
automatically transferred to the new EQ.
Linear Phase EQ uses a dierent underlying technology that preserves the phase of the audio
signal. Phase coherency is always maintained, even when you apply extreme EQ curves to the
sharpest signal transients.
A further dierence between Channel EQ and Linear Phase EQ is that the latter uses a xed
amount of CPU resources, regardless of the number of active bands. Linear Phase EQ also
introduces greater amounts of latency.
Note: It is recommended that you use Linear Phase EQ for mastering recorded audio and avoid
use when playing software instruments live, for example. You may need to use the latency
compensation feature of Logic Pro when using Linear Phase EQ.

Chapter 5 Equalizers 108
Linear Phase EQ parameters
The left side of the Channel EQ window incorporates the Gain and Analyzer controls. The central
area of the window includes the graphic display and parameters for shaping each EQ band.
Linear Phase EQ parameters
•Master Gain slider and eld: Drag to set the overall output level of the signal after boosting or
cutting individual frequency bands.
•Analyzer button: Click to turn the Analyzer on or o.
•Pre/Post EQ button: Click to determine if the Analyzer shows the frequency curve before or
after EQ is applied, when Analyzer mode is active.
•Resolution pop-up menu: Choose the sample resolution for the Analyzer. Choose from the
following menu items: low (1024 points), medium (2048 points), and high (4096 points).
•Band On/O buttons: Turn the corresponding band on or o. Each button’s icon indicates the
lter type:
•Band 1 is a highpass lter.
•Band 2 is a low shelving lter.
•Bands 3 through 6 are parametric bell lters.
•Band 7 is a high shelving lter.
•Band 8 is a lowpass lter.
•Graphic display: Shows the current curve of each EQ band. The scale is shown in dB.
•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 reects your changes immediately.
•Drag the pivot point in each band to adjust the Q factor. Q is shown beside the pointer
when it is moved over a pivot point.
•Frequency elds: Drag to adjust the frequency of each band.
•Gain/Slope elds: Drag to set the amount of gain for each band. For bands 1 and 8, this
changes the slope of the lter.
•Q elds: Drag to adjust the Q or resonance for each band—the range of frequencies around
the center frequency that are aected.
Chapter 5 Equalizers 109
Note: The Q parameter of band 1 and band 8 has no eect when the slope is set to 6 dB/Oct.
When the Q parameter is set to an extremely high value (such as 100), these lters aect only a
very narrow frequency band and can be used as notch lters.
•Link button: Click to turn on 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): Click to choose Peak or RMS.
•Analyzer Decay slider and eld (Extended Parameters area): Drag to adjust the decay rate
(in dB per second) of the Analyzer curve (peak decay in Peak mode or an averaged decay in
RMS mode).
•Gain-Q Couple Strength pop-up menu (Extended Parameters area): Choose the amount of Gain-Q
coupling.
•Strong: Preserves most of the perceived bandwidth.
•Light and medium: Allows some change as you raise or lower the gain.
•Asymmetric: Features 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 dierent Gain-Q Couple
Strength setting, the actual Q values will be dierent than they were when the automation
was recorded.
Linear Phase EQ use tips
Linear Phase EQ is typically used as a mastering tool that is inserted into master or output
channel strips. The way you use Linear Phase EQ depends on the audio material and your
intended outcome. A useful workow for many situations is as follows: Set Linear Phase EQ to
a at response (no frequencies boosted or cut), turn on the Analyzer, then play the audio signal.
Watch 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 attention to sections where the signal
distorts or clips. Use the graphic display or parameter controls to adjust the frequency bands.
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 aect a specic 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. Use the Q parameter or
parameters so that only a narrow range of frequencies is aected.
Each EQ band has a dierent 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 adjust the decibel scale of the graphic display by vertically dragging either the left
or right edge of the dB scale 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 area around the zero line, drag the left
side of the dB scale upward. Drag downward to decrease the resolution.
Chapter 5 Equalizers 110
Linear Phase EQ Analyzer
The Analyzer uses 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 simplies the task of setting
EQ curves to raise or lower the levels of frequencies or frequency ranges.
The bands derived from FFT analysis are scaled logarithmically—there are more bands in higher
octaves than in lower octaves.
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 −40 dB. The
Analyzer display is always dB-linear.
Note: High Analyzer resolutions require signicantly 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 turn o the Analyzer or close the Channel EQ window after setting
EQ parameters.
Match EQ
Match EQ overview
Match EQ allows you to analyze and store the average frequency spectrum of an audio le as a
template. You can apply this template to another audio signal so that it matches the spectrum
of the original le. This is also known as a ngerprint EQ, where one sonic ngerprint is applied to
another signal.
Match EQ enables you to acoustically match the tonal quality or overall sound of dierent songs
you plan to include on an album, for example, or to impart the color of any source recording to
your own projects.
Match EQ is a learning equalizer that analyzes the frequency spectrum of an audio signal such as
an audio le, a channel strip input signal, or a template. The average frequency spectrum of the
source le (the template) and of the current material (this can be the entire project or individual
channel strips within it) is analyzed. These two spectra are then matched, creating a lter curve.
This lter curve adapts the frequency response of the current material to match that of the
template. Before applying the lter curve, you can modify it by boosting or cutting any number
of frequencies or by inverting the curve.
The Analyzer allows you to visually compare the frequency spectrum of the source le and
the resulting curve, making it easier to make manual corrections at specic points within
the spectrum.
Note: Although Match EQ acoustically matches the frequency curve of two audio signals, it does
not match any dynamic dierences between the two signals.

Chapter 5 Equalizers 111
Match EQ parameters
Match EQ oers the following parameters.
Match EQ parameters
•Analyzer button: Turns the Analyzer function on or o.
•Pre/Post button: Click to determine if the Analyzer looks at the signal before (Pre) or after (Post)
the lter curve is applied.
•View pop-up menu: Set the information shown in the graphic display. Choices are:
•Auto: Displays information for the current function, as set with the active button below the
graphic display.
•Template: Displays the learned frequency curve template for the source le. This is shown in
red.
•Current Material: Displays the frequency curve for the audio learned as current material. This
is shown in green.
•Filter: Displays the lter curve created by matching the template and the current material.
This is shown in yellow.
•View button: Determines if separate curves are displayed by the Analyzer (L&R for stereo, All
Cha for surround) or the summed maximum level is shown (LR Max for stereo, Cha Max for
surround).
Note: The View parameters are disabled when you use the eect on a mono channel.
•Select buttons: Apply changes to the lter curve (created by matching the template with the
current material) to: the left (L), right (R), or both channels (L+R).
Note: The Select parameters are disabled when you use the eect on a mono channel.
•Select pop-up menu (Surround instances only): Choose an individual channel or all channels.
Changes to the lter curve aect the chosen channel when a single channel is selected.
•Channel Link slider and eld: Drag to rene settings made with the Select buttons or Select
pop-up menu.
•When set to 100%, all channels (L and R for stereo, or all surround channels) are represented
by a common EQ curve.
•When set to 0%, a separate lter curve is displayed for each channel (chosen with the Select
buttons or Select pop-up menu).
Chapter 5 Equalizers 112
•Settings between 0 and 100% blend these values with your lter curve changes for each
channel. This results in a hybrid curve.
Note: The Channel Link parameters are disabled when you use the eect on a mono channel.
•LFE Handling buttons (Extended Parameters area): In surround instances, click to process or
bypass the LFE channel.
•Graphic display: Displays the lter curve created by matching the template to the current
material. You can edit the lter curve (see Edit the Match EQ lter curve on page 115 ).
•Template Learn button: Starts/stops the process of learning the frequency spectrum of the
source le.
•Current Material Learn button: Starts/stops the process of learning the frequency spectrum of
the project you want to match with the source le.
•Current Material Match button: Click to match the frequency spectrum of the current material
to that of the template (source) le.
•Phase pop-up menu: Switches the operational principle of the lter curve.
•Linear: Prevents processing from altering the signal phase, but latency is higher.
•Minimal: Alters the signal phase (slightly), but latency is reduced.
•Minimal, Zero Latency: Adds no latency, but has a higher CPU overhead than the
other options.
•Apply slider and eld: Drag to determine the impact of the lter curve on the signal.
•Values above 100% magnify the eect.
•Values below 100% reduce it.
•Negative values (−1% to −100%) invert the peaks and troughs in the lter curve.
•A value of 100% has no impact on the lter curve.
•Smoothing slider and eld: Drag to set the amount of smoothing for the lter curve, using a
constant bandwidth set in semitone steps. A value of 0.0 has no impact on the lter curve.
A value of 1.0 means a smoothing bandwidth of one semitone. A value of 4.0 means a
smoothing bandwidth of four semitones (a major third). A value of 12.0 means a smoothing
bandwidth of one octave, and so on.
Note: Smoothing has no eect on any manual changes you make to the lter curve.
•Fade Extremes checkbox (Extended Parameters area): Select to smooth the lter curve at the
high and low extremes of the frequency spectrum.
Chapter 5 Equalizers 113
Use Match EQ
These tasks are those commonly used with Match EQ to match the frequency spectrum of a mix
with the spectrum of a source audio le. You can adapt some, or all, to your own workow.
Learn or create a Match EQ template
You can drag an audio le to the Template Learn or Current Material Learn buttons for use as
either the template or the current material. A progress bar appears while Match EQ is analyzing
the le. You can also load a previously saved plug-in setting, or you can import the settings of
another unsaved Match EQ instance by copying and pasting.
Do one of the following:
mDrag an audio le from the Finder to the Template Learn button, and select the source channel
strip as a sidechain.
mUse Match EQ on the source channel strip and save a setting. Import this setting into the target
Match EQ instance.
The lter curve is updated automatically each time a new template or current material spectrum
is learned or loaded when the Match button is turned on. You can alternate between the
matched (and possibly scaled or manually modied) lter curve and a at response by turning
the Match button on or o.
Match the EQ of a project mix to the EQ of a source audio le
1 In the project you want to match to the source audio le, insert Match EQ (typically on Output
1-2).
2 Drag the source audio le to the Template Learn button.
3 Return to the start of your mix, click Current Material Learn, and play your mix (the current
material) from start to nish.
4 When you are done, click Current Material Match (this automatically turns o the Current Material
Learn button).
When you click either of the Learn buttons, the View parameter is set to Automatic and the
graphic display shows the frequency curve for the function. You can review any of the frequency
curves when no le is being processed by choosing one of the other View options.
Note: Only one of the Learn buttons can be turned on at a time. For example, if the Learn button
in the Template section is on and you click the Learn button in the Current Material section,
analysis of the template le stops, the current status is used as the spectral template, and analysis
of the incoming audio signal (Current Material) begins.
Edit spectra with the Match EQ shortcut menu
This menu provides commands that can be applied to the spectrums of either the template or
the current material.
mControl-click (or right-click) either Learn button, then choose one of the following from the
shortcut menu:
•Clear Current Material Spectrum: Clears the current spectrum.
•Copy Current Spectrum: Copies the current spectrum to the Clipboard (this can be used by any
Match EQ instance in the current project).
•Paste Current Spectrum: Pastes the Clipboard contents to the current Match EQ instance.
•Load Current Material Spectrum from setting le: Loads the spectrum from a stored setting le.
Chapter 5 Equalizers 114
•Generate Current Material Spectrum from audio le: Generates a frequency spectrum for an
audio le that you have chosen.
Rene the Match EQ curve
Each time you match two audio signals, either by loading or learning a new spectrum while
Match is activated or by turning on Match after a new spectrum has been loaded, any existing
changes to the lter curve are discarded and Apply is set to 100%.
Do either of the following:
mDrag the Apply slider down from the default 100% value to avoid extreme spectral changes to
your mix.
mDrag the Smoothing slider to adjust the spectral detail of the generated EQ curve—if required.
Use the matched EQ on a channel strip
Match EQ creates a lter curve based on the dierences between the spectrum of the template
and the current material. This curve automatically compensates for dierences in gain between
the template and the current material, with the resulting EQ curve referenced to 0 dB. A yellow
lter response curve appears in the graphic display, showing the average spectrum of your mix.
This curve approximates (mirrors) the average spectrum of your source audio le.
1 Choose the channel strip that you want to match from the Sidechain pop-up menu of the
Match EQ window.
2 Click the Template Learn button.
3 Play the entire source audio le from start to nish. To stop the learn process, click the Template
Learn button again.
4 Return to the start of your mix, click Current Material Learn, and play your mix (the current
material) from start to nish.
5 When you are done, click Current Material Match (this automatically turns o the Current Material
Learn button).
Chapter 5 Equalizers 115
Edit the Match EQ lter curve
You can edit the lter curve in the graphic display by adjusting the various points shown in each
band. As you drag a point, the current value appears in a small box inside the graphic display,
allowing precise changes.
Adjust Match EQ curve values
Do any of the following:
mTo shift the peak frequency for the band (over the entire spectrum), drag horizontally.
mTo adjust the gain of the band, drag vertically.
mTo adjust the Q Factor, Shift-drag vertically.
mTo reset the gain to 0 dB, Option-drag.
Note: If you manually modify the lter curve, you can restore it to the original (or at) curve by
Option-clicking the background of the Analyzer display. Option-click the background again to
restore the most recent curve.
The Q factor of the lter is determined (and set) by the vertical distance between the clicked
position and the curve.
Set the Q factor in Match EQ
Do either of the following:
mTo set the maximum Q value of 10 (for notch-like lters), click the curve.
mTo decrease the Q value, click above or below the curve. The farther you click from the curve, the
smaller the value (down to the minimum of 0.3).
Change the Match EQ scale range
The colors and modes of the dB scales on the left and right of the display are automatically
adapted to the active function. If the Analyzer is active, the left scale displays the average
spectrum in the signal, while the right scale serves as a reference for the peak values of the
Analyzer. A dynamic range of 60 dB is shown by default. If this is not precise enough for your
edits, you can increase the range.
mDrag either scale to set values of up to +20 dB and −100 dB.
Change Match EQ gain with the scales
mDrag either scale to adjust the overall gain of the lter curve from −30 to +30 dB.
The left scale—and the right, if the Analyzer is inactive—shows the dB values for the lter curve.
Chapter 5 Equalizers 116
Single-Band EQ
The single-band EQ can operate in several modes. When you choose an EQ from the EQ Mode
pop-up menu, the parameters shown below change. You can choose:
•Low Cut or High Cut Filter: Low Cut Filter attenuates the frequency range that falls below
the selected frequency. High Cut Filter attenuates the frequency range above the
selected frequency.
•High Shelf or Low Shelf EQ: Low Shelving EQ aects only the frequency range that falls
below the selected frequency. High Shelving EQ aects only the frequency range above the
selected frequency.
•Parametric EQ: Parametric EQ is a simple lter 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 lter with a very narrow range. A symmetrical frequency range on either side of
the center frequency is boosted or cut.
Single-Band EQ parameters
•Frequency slider and eld: Drag to set the cuto frequency.
•Gain slider and eld (Shelf and Parametric lters only): Drag to set the amount of cut or boost.
•Slope pop-up menu (Cut lters only): Choose the amount of cut, in decibels per octave. The
higher the value, the more pronounced the eect.
•Q-Factor slider and eld: Drag to set the Q (bandwidth).

117
Filter eects overview
Filters are used to emphasize or suppress frequencies in an audio signal, resulting in a change in
the tonal color of the audio.
Logic Pro X contains a variety of advanced lter-based eects that you can use to creatively
modify your audio. These eects are most often used to radically alter the frequency spectrum of
a sound or mix.
Note: Equalizers (EQs) are special types of lters. They are not usually used as “eects” per se,
but as tools to rene the frequency spectrum of a sound or mix. See Equalizers overview on
page 104.
AutoFilter
AutoFilter overview
AutoFilter is a versatile lter eect with several unique features. You can use it to create classic,
analog-style synthesizer eects, or as a tool for creative sound design.
The eect 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
lter cuto.
The AutoFilter allows you to choose between dierent lter types and slopes, control the amount
of resonance, add distortion for more aggressive sounds, and mix the original, dry signal with the
processed signal.
Filter
parameters
Threshold
parameter
Envelope
parameters
Distortion parameters
LFO parameters
Output
parameters
Filter eects 6

Chapter 6 Filter eects 118
The main areas of the AutoFilter window are the Threshold, Envelope, LFO, Filter, Distortion, and
Output parameter sections.
•Threshold slider: Sets an input level that—if exceeded—triggers the envelope or LFO that
dynamically modulates lter cuto frequency. See AutoFilter threshold on page 118 .
•Envelope parameters: Dene how the lter cuto frequency is modulated over time. See
AutoFilter envelope on page 118 .
•LFO parameters: Dene how the lter cuto frequency is modulated by the LFO. See AutoFilter
LFO on page 119 .
•Filter parameters: Control the tonal color of the ltered sound. See AutoFilter lter on page 120 .
•Distortion parameters: Distort the signal both before and after the lter. See AutoFilter
distortion on page 121.
•Output parameters: Set the level of both the dry and eect signal. See AutoFilter output on
page 121.
AutoFilter threshold
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.
Note: Retriggering of the envelope or LFO occurs only if the Retrigger button is active.
You can use the envelope and LFO to modulate the lter cuto frequency.
AutoFilter envelope
The envelope is used to shape the lter cuto over time. When the input signal exceeds the set
threshold level, the envelope is triggered.
Envelope parameters
•Attack knob and eld: Rotate to set the attack time for the envelope.
•Decay knob and eld: Rotate to set the decay time for the envelope.

Chapter 6 Filter eects 119
•Sustain knob and eld: Rotate to set 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 eld: Rotate to set the release time for the envelope. This is triggered as soon
as the input signal falls below the threshold.
•Dynamic knob and eld: Rotate to determine the input signal modulation amount. You can
modulate the peak value of the envelope section by varying this control.
•Cuto Mod. slider and eld: Drag to determine the impact of the envelope on the
cuto frequency.
AutoFilter LFO
The LFO is used as a modulation source for lter cuto.
LFO parameters
•Coarse Rate slider and eld: Drag to set the speed of LFO modulation. Use to set the LFO
frequency in hertz.
Note: The labels shown for the Rate knob, slider, and eld change when you activate Beat
Sync. Only the Rate knob and eld are then available.
•Fine Rate knob: Rotate to set the speed of LFO modulation. Use to ne-tune the LFO frequency.
•Beat Sync button: Turn on to synchronize the LFO to the host application tempo. You can
choose from bar values, triplet values, and more by using the Rate knob and eld.
•Phase knob and eld: Rotate to set the phase relationship between the LFO rate and the host
application tempo—when Beat Sync is active. This parameter is dimmed when Beat Sync
is disabled.
•Decay/Delay knob and eld: Rotate to set the time it takes for the LFO to go from 0 to its
maximum value.
•Rate Mod. knob and eld: Rotate to set 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 eld: Rotate to set the phase relationship of the LFO modulations
between the two channels (stereo only).
•Cuto Mod. slider and eld: Drag to determine the impact of the LFO on the cuto frequency.
•Retrigger button: Turn on to start the LFO waveform at 0 each time the threshold is exceeded.
•Waveform buttons: Click to select the shape of the LFO waveform: descending sawtooth,
ascending sawtooth, triangle, pulse wave, or random.
•Pulse Width slider and eld: Drag to alter the curve shape of the selected waveform.

Chapter 6 Filter eects 12 0
AutoFilter lter
The Filter parameters allow you to precisely tailor the tonal color.
Filter parameters
•Cuto knob and eld: Rotate to set the cuto frequency for the lter. Higher frequencies are
attenuated, whereas lower frequencies are allowed to pass through in a lowpass lter. The
reverse is true in a highpass lter. When the State Variable Filter is set to bandpass (BP) mode,
the lter cuto determines the center frequency of the frequency band that is allowed to pass.
•Resonance knob and eld: Rotate to boost or cut signals in the frequency band that surrounds
the cuto frequency. Very high Resonance values cause the lter to begin oscillating at the
cuto frequency. This self-oscillation occurs before you reach the maximum Resonance value.
•Fatness slider and eld: Drag to boost the level of low frequency content. When you set Fatness
to its maximum value, adjusting Resonance has no eect on frequencies below the cuto
frequency. This parameter is used to compensate for a weak or “brittle” sound caused by high
resonance values, when in the lowpass lter mode.
•State Variable Filter buttons: Switch the lter between highpass (HP), bandpass (BP), or lowpass
(LP) modes.
•4-Pole Lowpass Filter buttons: Click to set the slope of the lowpass lter to 6, 12, 18, or 24 dB
per octave.
Note: Clicking one of these buttons automatically chooses the lowpass (LP) lter and slope,
overriding any active State Variable Filter button.

Chapter 6 Filter eects 121
AutoFilter distortion
The Distortion parameters can be used to overdrive the lter input or lter output. The distortion
input and output modules are identical, but their dierent positions in the signal chain—before
and after the lter, respectively—result in remarkably dissimilar sounds.
Distortion parameters
•Input knob and eld: Rotate to set the amount of distortion applied before the lter section
processes the signal.
•Output knob and eld: Rotate to set the amount of distortion applied after the lter section
processes the signal.
AutoFilter output
The Output parameters are used to set the wet/dry balance and overall level.
Output parameters
•Dry Signal slider and eld: Drag to set the amount of original, dry signal added to the
ltered signal.
•Main Out slider and eld: