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Table of Contents
 Introduction 23 Undo and Redo 40

Welcome! 24

Introducing Reason Rack Plugin 24

Audio and MIDI Basics 41

Conventions in the manual 24 General audio and MIDI handling 42

Typical input/output configurations 42
About the Authorization system 26
Registering Reason (licensed version only) 26 The I/O device 45
Authorizing Reason Rack Plugin 26
Authorizing a computer for long-term off-line use 27 Audio settings 46
What happens if an authorization expires? 29 Render audio using host buffer size setting 46

About Rack Extensions 30 About Plugin Delay Compensation 46

The "Update Rack Extension Licenses" alert 30

Getting all your content 30

Using Reason Rack Plugin as
About automatic update checks 31 an Instrument 47
Creating an instrument 48
Overview 33
Browsing for patches 49
Adding Reason Rack Plugin in your
Adding effects 50
project 34
Layering instruments 52
The Reason Rack Plugin window 34
Zoom 35
About different Themes 35 Using separate audio outputs 53

Editing parameters 36 Adding Players 54

Knobs 36
Faders and sliders 36 Using Mixer devices 56
Buttons 36
Fold/Unfold buttons 37 Detailed control over MIDI note input 57
Multi Mode selectors 37
Numerical controls 38
Menus and alpha-numeric controls 38

Tool Tips 38 Using Reason Rack Plugin as

an Effect 59
Context menus 39
Parameter context menus 39 Creating an effect 60
Device context menus 39
Empty rack context menu 39
Browsing for patches 61

4 TABLE OF CONTENTS
 Creating effect chains 62 Sounds, Patches and the
Using sidechain inputs 62
Browser 81
About patches 82
About the “Load Default Sound in New Devices” setting
82
Working in the Rack 65
Loading patches 82
Creating devices 66
The Device Palette 67
Saving patches 85
Copying and pasting patches between devices 85
Selecting devices 69 Initializing patches and resetting device parameters 85

Moving devices 70 Using the Browser 86

Browser overview 86
Re-routing devices 70 Creating a device by browsing patches 88
Browsing patches for a device 89
Deleting devices 71 Browsing samples for a device 90
Selecting multiple files 91
About patch formats and sampler devices 91
Replacing devices 71 About browse lists 91
Other Browse Focus modes 92
Cut, Copy, Paste and Duplicate devices Using the browser without Browse Focus 92
71 The Locations list 93
Filtering 95
Editing Tags and Categories 99
Naming devices 71 Other Browser functions 100

Handling Missing Sounds 101

Routing Audio and CV 73 About ReFills 102

Working on the back of the rack 74

Hiding cables 75
The I/O device 103
Signal types 76
Audio signals 76
CV/Gate signals 76 Introduction 104
About MIDI routing 77
The back panel 104
Manual routing 77
Dragging cables 77 The front panel 104
Using the routing menu 78 Level meters 104
Checking and following cable connections 78 Sum To Main (3-16 only) 105
Name labels 105
Auto-routing 78 Audio In/Out indicators 105
Shuffle 105

5 TABLE OF CONTENTS
 Dr. Octo Rex Loop Player 107 Connections 125
Modulation Inputs 126
Modulation Outputs 126
Introduction 108 Gate Inputs 126
ReCycled Loops 108 Gate Output 126
Slice Outputs 126
About REX file formats 109 Main Outputs 126

Loading and saving Dr. Octo Rex patches

109
About the Dr. Octo Rex patch format 109 Europa Shapeshifting
Synthesizer 127
Playing Loops 109
Switching playback between Loop Slots 110
Introduction 128
Adding Loops 111
Loading Loops “On the Fly” 111 Panel overview 129
Removing Loops 112
Cut/Copy and Paste Loops between Loop Slots 112 Signal flow 130

Playing individual Loop Slices 112 Playing and using Europa 131
Loading and saving patches 131
Slice handling 113 Global output controls 131
Selecting Slices 113 Global performance and “play” controls 131
Editing Slices in the Waveform Display 114
The Slice Edit Mode 115 Panel reference 133
Sound Engines On/Off and Edit Focus section 133
Dr. Octo Rex panel parameters 116 The Oscillator section 133
Pitch and Mod wheels 116 The Modifiers section 136
Trig Next Loop 116 The Spectral Filter 138
Note To Slot 116 The Harmonics section 140
Loop Slot buttons 117 The Unison section 142
Enable Loop Playback and Run 117 The User Wave and Mixer section 143
Volume 117 The Filter section 144
Global Transpose 117 The Amplifier section 146
The Envelopes section 148
The LFO section 152
Dr. Octo Rex synth parameters 118 The Effects section 153
Select Loop & Load Slot 118 The Modulation Bus section 156
Copy MIDI Notes to a sequencer track 118
Loop Transpose 119
Loop Level 119 Connections 160
Oscillator section 119 Sequencer Control inputs 160
Mod. Wheel 120 CV Modulation inputs and outputs 160
Velocity section 120 Audio Output 160
The Filter Section 121
Envelope section 122 Tips and Tricks 161
LFO section 123 Creating an individual “pre amp envelope” for a Sound
Pitch Bend Range 124 Engine 161
Setting number of voices - polyphony 125 Recording display movements in the sequencer 162
Audio Quality settings 125

6 TABLE OF CONTENTS
 Grain Sample Manipulator 163 The Filter section 202
The Amp section 204
The Delay section 206
Introduction 164 The Reverb section 207
A few words about granular synthesis 165
Connections 209
Panel overview 166 Sequencer Control inputs 209
Modulation Inputs 209
Playing and using Grain 167 Audio Out 209
Loading and saving patches 167
Global performance and “play” controls 167
Global output controls 168
ID8 Instrument Device 211
Panel reference 169
The Sample section 169
Introduction 212
The Playback Algorithms section 171
The Sounds 212
The Oscillator section 176
The Filter section 177
The Amplifier section 178 Using the ID8 213
The Envelopes section 179 Selecting Sounds 213
The LFO section 183 Controlling Sounds 213
The Effects section 183 About saving edited Sounds 214
The Modulation Bus section 187

Connections 191
Sequencer Control inputs 191 Klang Tuned Percussion 215
CV Modulation inputs and outputs 191
Audio Output 191
Introduction 216
Tips and Tricks 192
Automating sample playback parameters from the
Panel overview 216
sequencer 192
Using Klang 217
Loading and saving patches 217

Humana Vocal Ensemble 193 Global performance and “play” controls

217
Introduction 194
Panel controls 218
The Instruments section 218
Panel overview 194 The Filter section 223
The Amp section 225
Using Humana 195 The Delay section 226
Loading and saving patches 195 The Reverb section 228

Global performance and “play” controls Connections 229

195 Sequencer Control inputs 229
Modulation Inputs 229
Audio Out 229
Panel controls 196
The Instruments section 196

7 TABLE OF CONTENTS
 Kong Drum Designer 231 Compressor 264
Filter 265
Parametric EQ 265
Introduction 232 Ring Modulator 266
Rattler 267
Overview 232 Tape Echo 267
The Pad Section 232 Overdrive/Resonator 268
The Drum Control Panel 233
The Drum and FX Section 233 Connections 269
About using custom backdrops 233 Sequencer Control 269
Modulation Input 269
About file formats 233 Aux Send Out 269
Gate In and Out 270
Audio Out 3-16 270
Using patches 234 Main Audio Out 270
Loading a Kit Patch 234
Checking the sounds in a Kit Patch 235
Creating a new Kit Patch 235 Using Kong as an effect device 270
Creating an empty Kit Patch 236
Saving Kit Patches 236 Using external effects with Kong 271

Pad Settings 237

Assigning Drums to Pads 237
Renaming Pads 238 Malström Synthesizer 273
Copying & Pasting Drums between Pads 238
Assigning Hit Type to Pads 238
Muting and Soloing Pads 239 Introduction 274
Working with Pad Groups 240 Features 274
Theory of operation 275
Loading and Saving Patches 275
The Drum and FX section 241
Signal flow 242
The Drum Control Panel 244 The Oscillator section 276
The Drum Module slot 246 Setting oscillator frequency 277
The FX slots 247 Controlling playback of the graintable 277
The amplitude envelopes 278
The Drum modules 248
NN-Nano Sampler 248 The Modulator section 279
Nurse Rex Loop Player 251 Modulator parameters 279
Physical Bass Drum, Snare Drum and Tom Tom 256 Destinations 280
Synth Bass Drum, Snare Drum and Tom Tom 258
Synth Hi-hat 259 The Filter section 281
The Filters 282
The Support Generator modules 260 The Filter Envelope 284
Noise Generator 260 The Shaper 285
Tone Generator 261
Routing 287
The FX modules 262 Routing examples 288
Using CV modulation of Bus FX and Master FX The output controls 291
parameters 262
Drum Room Reverb 263 The play controls 291
Transient Shaper 263 Polyphony - setting the number of voices 292

8 TABLE OF CONTENTS
 Porta (portamento) 292 Panel reference 312
Legato 292 The Sample section 312
The Pitch Bend and Modulation wheels 293 Start Position 314
The Velocity controls 293 Speed and Speed Mod 315
The Modulation wheel controls 294 Stretch 316
Slices 319
Connections 295 Pitch 320
Audio Output 295 The Filter section 321
Audio Input 295 The Filter Envelope and Amp Envelope sections 324
Sequencer Control 295 The LFO section 326
Gate Input 296 The Amp section 326
Modulation Input 296 The Compressor 327
Modulation Output 296 The Effect section 327
The EQ section 328
The Send section 328
Routing external audio to the filters 297
Connections 329
Sequencer Control inputs 329
CV In 1-4 329
MIDI Out Device 299 Slot Out 1-8 329
FX Send Out 329
Introduction 300 Audio Output 329

Using the MIDI Out Device 300 Tips and Tricks 330
Setting up for MIDI controlling an external track/plugin Optimizing performance/DSP Load 330
300 Creating a “velocity layered” instrument 330
Modulating MIDI Controllers from CV signals 301 Extending the sample “tail” (without looping) 331
Automating the sample Start and End markers 332
Connections 302
Sequencer Control 302
CV In to MIDI CC Out 302
Monotone Bass Synthesizer
333
Mimic Creative Sampler 303 Introduction 334

Introduction 304 Panel overview 335

Panel overview 305 Signal flow 336

Signal flow 306 Playing and using Monotone 337

Loading and saving patches 337
Playing and using Mimic 307 Global output controls 337
Loading and saving patches 307 Global performance and “play” controls 337
Global output controls 307
Slot Mode 307 Panel reference 338
Slot Select 310 The Oscillator section 338
Performance and “play” controls 310 The Filter section 339
The Amplifier section 341

9 TABLE OF CONTENTS
 Chorus 342 Managing Zones and Samples 363
Delay 343 Creating a Key Map 363
The LFO section 343 About file formats and REX slices 364
The Envelope section 344 Adding more samples to the Key Map 364
Replacing a sample 364
Connections 345 Quick browsing through samples 365
Sequencer Control inputs 345 Removing samples 365
Modulation inputs 345 Auditioning samples 365
Audio Output 345 Adding empty Zones 365
Duplicating Zones 366
Removing Zones 366
Rearranging Zones in the List 366

NN-XT Sampler 347 Working with Grouping 366

About Groups 366
Introduction 348 Creating a Group 366
Moving a Group to another position in the List 367
Panel overview 349 Moving a Zone from one Group to another 367
The main panel 349 Selecting a Group and/or Zones in a Group 368
The Remote Editor panel 349 The Group Parameters 368

Loading complete Patches and REX files Working with Key Ranges 368
350 About Key Ranges 368
Setting up Key Ranges 368
Loading NN-XT Patches 350
About the Lock Root Keys function 372
Loading NN-19 Patches 350
About the Solo Sample function 373
Loading SoundFonts 351
Sorting Zones by Note 374
Loading complete REX files as Patches 351

Using the main panel 352 Setting Root Notes and Tuning 375
The Pitch and Modulation wheels 352 About the Root Key 375
The External Control wheel 352 Setting the Root Note manually 375
High Quality Interpolation 353 Tuning samples manually 375
Global Controls 353 Setting the Root Note and Tuning using pitch detection
376
About changing the pitch of samples 376
Overview of the Remote Editor panel 355
The Key Map display 355
Sample parameters 356
Using Automap 376
Group parameters 356
Synth Parameters 357 Layered, crossfaded and velocity
switched sounds 377
About Samples and Zones 357 Creating layered sounds 377
About velocity ranges 377
Setting velocity range for a Zone 379
Selections and Edit Focus 358 About Crossfading Between Zones 379
Selecting Zones 359
Setting crossfading for a Zone 381
Moving Edit Focus 361

Using Alternate 381

Adjusting parameters 361 About the Alternate function 381
Adjusting Synth parameters 361
Adjusting Group parameters 361
Sample parameters 362 Sample parameters 382
Root Note and Tune 382

10 TABLE OF CONTENTS
 Sample Start and End 382 Selecting Key Zones 404
Loop Start and End 382 Setting the Key Zone Range 404
Play Mode 383 Deleting a Key Zone 405
Lo Key and Hi Key 383 About Key zones, assigned and unassigned samples
Lo Vel and Hi Vel 383 405
Fade In and Fade Out 383 Adding sample(s) to a Key Map 405
Alt 383 Setting the Root Key 406
Out 383 Removing sample(s) from a Key Map 406
Removing all unassigned samples 406
Group parameters 384 Rearranging samples in a Key Map 406
Key Poly 384 Setting Sample Level 406
Legato and Retrig 385 Tuning samples 407
LFO 1 Rate 385 Looping Samples 407
Portamento 385 About the Solo Sample function 407

Synth parameters 386 Automap Samples 408

The Modulation controls 386 Mapping samples without Root Key or Tuning
The Velocity controls 388 information 408
The Pitch section 389 How Mapping Information is saved 408
The Filter section 390
The Modulation Envelope 391 NN-19 synth parameters 409
The Amplitude Envelope 393 The Oscillator Section 409
The LFOs 394 The Filter Section 410
Envelope Section 411
Connections 397 LFO Section 412
Sequencer Control 397
Modulation Input 397 Play Parameters 414
Gate Input 397 Velocity Control 414
Audio Output 397 Pitch Bend and Modulation Wheels 415
Legato 415
Retrig 415
Portamento (Time) 416
Setting Number of Voices - Polyphony 416
NN-19 Sampler 399 Voice Spread 416
Low Bandwidth 416
Introduction 400 Controller Section 416

General sampling principles 400 Connections 417

Background 400 Audio Outputs 417
Multisampling vs. single samples 400 Mono Sequencer Control 417
Modulation Inputs 417
About audio file formats 401 Modulation Outputs 417
Gate Inputs 418
Loading a Sample Patch 401
Loading REX Files as Patches 401

About Key Zones and samples 402

Loading a Sample into an empty NN-19 402
Loading SoundFont samples 403
Loading REX slices as samples 403
Creating Key Zones 404

11 TABLE OF CONTENTS
 Pangea World Instruments The Oscillator section 444
The Mixer 446
419 The Filter section 446
The Filter Envelope section 449
The Amp section 450
Introduction 420
The Amp Envelope section 450
The Mod LFO section 452
Panel overview 420 The Mod Envelope section 453
The Global LFO section 454
Using Pangea 421 The Chorus section 455
Loading and saving patches 421 The Reverb section 455

Global performance and “play” controls Connections 456

421 Sequencer Control 456
Mix/Morph 456
CV Modulation inputs 456
Panel controls 422 Audio Output 456
The Instruments section 422 Wheels 456
The Filter section 427
The Amp section 429
The Delay section 431
The Reverb section 432
Radical Piano 457
Connections 434
Sequencer Control inputs 434 Introduction 458
Modulation Inputs 434 The pianos 458
Audio Out 434
Using Radical Piano 460
Loading and saving patches 460
Selecting piano sound sets 460
Polytone Dual-Layer Character 460
Volume 461
Synthesizer 435 Velocity Response 461
Tune 462
Introduction 436 Sustain 462
Resonance 463
Envelope 463
Panel overview 437 Mechanics 464
EQ 464
Signal flow 438 Ambience 465
Output 465
Playing and using Polytone Dual-Layer
Synthesizer 439 Connections 466
Loading and saving patches 439 Sequencer Control 466
Global output controls 439 Modulation In 466
Global performance and “play” controls 439 Audio In 466
Layer controls 440 Audio Out 466

Panel reference 442 Additional external control 467

Glide 442
Wheels 443
P.Bend Range 443

12 TABLE OF CONTENTS
 Redrum Drum Computer 469 The Drum Channel sections 491
Auditioning samples 491
Selecting a Drum Channel 491
Introduction 470 Muting and soloing Drum Channels 491
Setting the Drum Channel volumes 491
About file formats 471 Setting the Send Effect levels 492

Using patches 472 The Sample Playback section 492

Loading a patch 472 Selecting Samples 492
Checking the sounds in a patch 472 Setting the Sample Start and End 493
Creating a new patch 473 Setting the Panning 493
Creating an empty patch 473 Setting the Pitch 493
Setting Fade In and Fade Out 494
Programming patterns 474
Pattern basics 474 The Insert Effects section 494
Pattern tutorial 475 Distortion 494
Setting pattern length 476 Low Cut and Hi Cut Filter 495
Setting pattern resolution 477
Step dynamics 477 The Master FX section 495
Pattern Shuffle 478 Delay 496
Flam 478 Reverb 497
The Pattern Enable switch 478 Compressor 498
The Enable Pattern Section switch 479 Master Pitch 500
Pattern functions 479 Master Reverb 500
Chaining patterns 479 Master Filter 500
Copy MIDI Notes to a sequencer track 480 Master Volume 501

Redrum parameters 481 Connections 502

Drum sound settings 481 Drum Gate In/Out 502
Global settings 484 Separate Outputs 502
Main Output L & R 502
Using Redrum as a sound module 485

Connections 486
Subtractor Synthesizer 503
Introduction 504
Rytmik Drum Machine 487 Loading and Saving Patches 504

Introduction 488 The Oscillator Section 505

Oscillator 1 Waveform 505
Panel overview 489 Setting Oscillator 1 Frequency - Octave/Semitone/Cent
507
Oscillator Keyboard Tracking 507
Signal flow 490 Using Oscillator 2 507
Oscillator 2 Waveform 508
Global controls 490 Noise Generator 508
Loading and saving patches 490 Phase Offset Modulation 509
Frequency Modulation (FM) 511
Ring Modulation 512

13 TABLE OF CONTENTS
 The Filter Section 513 Mix section 545
Filter 1 Type 513 Filter slots 545
Filter 1 Frequency 516 Shaper 549
Resonance 516 Amp section 549
Filter Keyboard Track (Kbd) 516 LFO 1 550
Filter 2 517 Envelope sections 551
Global section 552
Envelopes - General 518
Amplitude Envelope 519 Modulation bus routing section 554
Filter Envelope 519
Mod Envelope 520 Step Sequencer 563
Basic operation 563
LFO Section 521
LFO 1 Parameters 521 Connections 567
LFO 2 Parameters 522

Play Parameters 523

Velocity Control 523
Pitch Bend and Modulation Wheels 524
Alligator Triple Filtered Gate
Legato 525 569
Retrig 525
Portamento (Time) 525
Introduction 570
Setting Number of Voices - Polyphony 526
About the Patch format 570
About the Low Bandwidth button 526

Overview and signal flow 570

External Modulation 526
Parameters 572
Connections 527
Common effect device parameters 572
Audio Output 527
Pattern section 572
Sequencer Control 527
Gate and Amp Envelope 573
Modulation Inputs 528
Filters and Modulation 574
Modulation Outputs 528
Effects 576
Gate Inputs 528
Mix controls 578

Audio connections 579

Main Inputs and Outputs 579
Thor Polysonic Synthesizer Separate Outputs 579
529
CV connections 580
Gate inputs 580
Introduction 530
CV Modulation inputs 580
Loading and Saving Patches 530
Gate Outputs 580
LFO CV Out 580
Thor elements 531
The built-in patterns 581
The Controller panel 532
Methods and Tips 582
Using the Programmer 534 Playing the Alligator live 582
Basic connections - a tutorial 535 Playing the gates from Matrix patterns 582
The Oscillator section 538 Controlling other sounds and effects 582

14 TABLE OF CONTENTS
 Audiomatic Retro Transformer Channel Dynamics
583 Compressor & Gate 605
Introduction 584 Introduction 606

Using Audiomatic Retro Transformer 584 Panel reference 606

Gain 584 Global controls 606
The Presets 585 The Compressor section 607
Transform 586 The Gate/Expander section 608
Dry/Wet 586 External Sidechain 609
Volume 587
Connections 610
Connections 587 CV Outputs 610
CV Modulation In 587 Sidechain Input Left & Right 610
Input L&R 587 Input Left & Right 610
Output L&R 587 Output Left & Right 610

BV512 Vocoder 589 Channel EQ Equalizer 611

Introduction 590 Introduction 612
How does a vocoder work? 590
Panel reference 612
Setting up for vocoding 591 Global controls 612
The Filter section 612
Using the BV512 as an equalizer 592 The Equalizer section 613

BV512 parameters 593 Connections 616

CV Inputs 616
Input Left & Right 616
Connections 595 Output Left & Right 616

Tips and tricks 596

Choosing a carrier sound 596
Choosing a modulator sound 597
Using the modulator as carrier 597
Master Bus Compressor 617
Controlling the Hold function 598
Using the individual band level connections 598 Introduction 618
“Playing” the vocoder from a MIDI keyboard 600
Using the BV512 as a reverb 601 Panel reference 618
Global controls 618
Compression controls 619
External Sidechain 620

Connections 621
Comp Gain Reduction 621

15 TABLE OF CONTENTS
 Sidechain Input Left & Right 621 Bend and Vibrato wheels 644
Input Left & Right 621 Input signal type 645
Output Left & Right 621 MIDI Input 645
Pitch Adjust section 646
Transpose section 647
Formant section 647
The Output Mixer section 647
The MClass Effects 623
Connections 648
The MClass effects 624 Sequencer Control 648
CV In 648
The MClass Equalizer 625 CV Out 649
Audio In 649
The MClass Stereo Imager 626 Voice Synth Out 649
Audio Out 649

The MClass Compressor 627

The MClass Maximizer 630 Pulveriser 651

Introduction 652
About the Patch format 652
Neptune Pitch Adjuster and
Voice Synth 631 Parameters 652
Common effect device parameters 652
Introduction 632 Signal Routing selector 653
Typical use cases 632 The Squash section 654
The Dirt section 654
The Filter section 655
Overview and basic concepts 633 The Tremor section 656
Sections overview 633 The Follower section 657
The display 634 Blend 658
Volume 658
Setting up for pitch processing 634
Modulation inputs and outputs 659
Using pitch correction 634 CV Modulation inputs 659
Basic settings for pitch correction 635 Audio Modulation inputs 659
Using automatic pitch correction 636 CV Modulation outputs 660
Using manual pitch correction 640
Demolition tips and tricks 660
Using pitch shifting (Transpose) 641 Beef up your sounds 660
Make your pads tremble 660
Using Formant control 642
What are formants? 642
Using the Formant function 643
Quartet Chorus Ensemble 661
Using the Voice Synth 643
Introduction 662
Panel parameters 644
Level Meter and Bypass/On/Off switch 644

16 TABLE OF CONTENTS
 Panel reference 662 The main panel 691
Global controls 662 The Remote Programmer 691
Chorus 663
BBD 665 Reverb algorithms and parameters 692
FFT 666 Common effect device parameters 692
Grain 668 About the main panel parameters 692
Selecting an algorithm 692
Connections 670 Small Space 693
CV Input 670 Room 693
Input Left & Right 670 Hall 694
Output Left & Right 670 Arena 694
Plate 694
Spring 694
Echo 695
Multi Tap 695
Ripley Space Delay 671 Reverse 696
Convolution 697
Introduction 672
The EQ section 700
Panel overview 672
The Gate section 701
Signal flow 673
CV Inputs 702
Panel reference 674
Loading and saving patches 674
The Delay section 674
The Feedback Filter section 676 Scream 4
The Reverb section 678
The Noise section 678 Sound Destruction Unit 703
The Distortion section 679
The Digital section 679 Scream 4 Sound Destruction Unit 704
The EQ section 680
About the Patch format 704
The Output section 681
The Ducker section 681
The Modulation and Modulation Matrix sections 681 Parameters 704
Common effect device parameters 704
Damage section controls 704
Connections 686 Description of the various Damage Type algorithms 705
Audio jacks 686 Cut section (EQ) 706
CV Inputs 687 Body section 706
CV Outputs 687 About the Master level control 707
CV inputs and outputs 708

Screamy tips and tricks 709

RV7000 Mk II Advanced Creating a heavy drum sound 709
Reverb 689 Warming up a mix with the Tape effect 709
Using the Body section as a sound enhancer/phaser/
wah 709
Overview 690 Emulating vintage digital gear 710
About the Patch format 690 Creating a real dynamic wah effect with the envelope
Connections 690 follower 710

17 TABLE OF CONTENTS
 Sidechain Tool 711 Connections 728
CV In 728
Input Left & Right 728
Introduction 712 Output Left & Right 728

Panel reference 712

Loading and saving patches 712
Auto Pump/Trigger/Sidechain mode switch 712
Ducking curve parameters (Auto Pump and Trigger
Sweeper Modulation Effect
modes) 712 729
Auto Pump mode specific parameters 714
Trigger mode specific parameters 714
Sidechain mode specific parameters 715
Introduction 730
Common parameters for all modes 717
Panel reference 730
Connections 717 Global controls 730
The Phaser 733
CV Input 717
The Flanger 736
CV Output 718
The Filter 738
Sidechain Input Left & Right 718
LFO 741
Input Left & Right 718
The Envelope Modulator 742
Output Left & Right 718
The Audio Follower Modulator 745

Connections 746
CV Input 746
Softube Amps 719 CV Output 746
Input Left & Right 747
Introduction 720 Output Left & Right 747
Basic usage 720
Front panel 720

Using the Softube Amps 721 Synchronous

Loading and saving patches 721
Selecting Amp and Cabinet model 721
Timed Effect Modulator 749
About the Amp and Cabinet models 722
Amp panel controls 723 Introduction 750
Connections 724
Panel overview 751

Using Synchronous 752

Stereo Tool 725 Loading and saving patches 752
Drawing and assigning modulation curves - a tutorial
Introduction 726 752
Editing modulation curves - a tutorial 754
Panel reference 726
Widening 726 Panel reference 755
Low Bypass 726 The display section 755
Frequency Adjust 726 Modulation controls 758
Mono 727 Dist section 759
Filter 760
Delay 762
Reverb 764

18 TABLE OF CONTENTS
 Level 765 Spider CV Merger & Splitter 787
Master Controls 765
RV-7 Digital Reverb 791
About automation of display section
parameters 766 D-11 Foldback Distortion 793

Connections 767 ECF-42 Envelope Controlled Filter 794

CV In 767
CV Out 768
Audio In L&R 768 PH-90 Phaser 796
Audio Out L&R 768
UN-16 Unison 798

COMP-01 Auto Make-up Gain

The Echo 769 Compressor 799

Introduction 770 PEQ-2 Two Band Parametric EQ 800

About the Patch format 770

Parameters 770
Common effect device parameters 770 The Combinator 801
The Mode section 771
The Delay section 772
Introduction 802
The Feedback section 773
The Color section 774
The Modulation section 775 Combinator overview 803
The Output section 776
Creating a Combinator device 804
CV/Gate inputs 777 Creating an empty Combinator device 804
Creating a Combinator by combining devices 804
Creating a Combinator by browsing patches 804
The Breakout Jacks 777
About internal and external audio
Tips and Tricks 778
Using the Roll function 778 connections 805
Creating “pitched” delay 778 External audio connections 805
Distorted external feedback 778 Internal audio connections 805
About External Routing 806

Adding devices to a Combi 807

Creating new devices in a Combi 807
Half-Rack Effects 779 Adding existing devices to the Combi from the rack 808
Combining two Combis 808
Common effect device features 780
Combinator handling 809
DDL-1 Digital Delay Line 782 Moving the entire Combi 809
Moving devices within a Combi 809
CF-101 Chorus/Flanger 783 Moving devices out of a Combi 809
Deleting devices in a Combi 809
Uncombining devices 809
Spider Audio Merger & Splitter 785

19 TABLE OF CONTENTS
 Configuring the Combinator panel 810 The Line Mixer 6:2 835
Opening the Configuration panel 810
Setting the front panel size 811
Selecting front panel background color 811 Introduction 836
Selecting a backdrop image 811
Selecting, positioning and editing front panel controls Channel parameters 836
813
The Auxiliary Return section 836
Assigning panel controls to parameters
in the Editor 817 Master level 836
Key Mapping instrument devices 817
Setting Velocity Ranges for instrument devices 819
Connections 837
Using Modulation Routing 820
About Rotary/Slider and Switch controls 821
About the CV Inputs 821
Assigning panel controls to device parameters 821 Matrix Pattern Sequencer 839
CV Connections 826 Introduction 840
Sequencer Control inputs 826 About the three Output types 840
Control CV In 826
Wheel CV In and Source CV In 827 Programming patterns 841
Pattern basics 841
Tutorial 843
Using Curve Patterns 845
Gain Tool 829 Setting Pattern Length 846
Using Tied Notes 846
Setting Pattern Resolution 847
Introduction 830 Pattern Shuffle 847
Pattern Mute 847
Signal flow 830 Pattern Functions 847
Chaining Patterns 848
Panel reference 830 Copy MIDI Notes to a sequencer track 848
The input mode switch 830
The Gain input mode 831 Example usage 849
The Mix input mode 831 Using the Matrix for modulation 849
The X-Fade input mode 831 Programming “Acid Style” lead lines 850
The output mode switch 832 Triggering samples 850
The Width/Pan output mode 832
The Dual Pan output mode 832
The Router output mode 833
The Output buttons 833
The output level meters 833
Mixer 14:2 851

Connections 834 Introduction 852

Audio Inputs 834
CV Inputs 834 The Channel Strip 852
Audio Outputs 834 Channel Strip Controls 853

The Mixer signal flow 854

20 TABLE OF CONTENTS
 About the EQ modes 854 CV connections 878

The Auxiliary Return Section 855 Tips and tricks 880

The Master Fader 855

Connections 855 Working with Players 881

Chaining several Mixer 14:2 devices 857 About this chapter 882

Overview 882
General recording principle 883
Pulsar Dual LFO 859
Using Players 883
Introduction 860 Creating Players 883
Chaining Players 884
Replacing Players 884
Panel parameters 860 Deleting Players 884
LFO 1&2 common parameters 860 Naming Players 884
LFO 1 specific parameters 862 About Players in Combinators 884
LFO 2 specific parameters 862 Common Player device parameters 884
LFO 2 to LFO 1 modulation parameters 862 Getting the Player MIDI output onto a track in your DAW
Envelope 863 885
KBD Follow 865
Dual Arpeggio 886
Modulation inputs and outputs 866 The Display sections 888
LFO 1&2 input sections 866
LFO 1&2 output sections 867
Output LFO 1+2 867 Note Echo 894
Envelope connections 867
Scales & Chords 895
Tips and Tricks 868 Scales 895
Patch between LFO 1 and LFO 2 on the back for more Filter Notes 896
flexibility 868 Chords 897
Using Pulsar as a monophonic synth 868
Beat Map 900
Included content 900
The front panel 901
Map Select 902
RPG-8 Arpeggiator 869 Density 902
Lock Position 903
Introduction 870 Mirror notes 903
Setting MIDI note numbers 904
Using the RPG-8 871 Global settings 905
Beat Map and the main sequencer 905
Setting up 871
Editing the drum notes 906
Using CV 906
RPG-8 Parameters 874
MIDI-CV Converter parameters 874
Arpeggiator parameters 875
Pattern editor 877

21 TABLE OF CONTENTS
 Tips & Tricks 907
Generating scale-correct arpeggios from single notes
907
Generating chord arpeggios 907
Creating parallel chords 907
Using a Scales & Chords device as a “MIDI Note monitor”
908

Settings 909
The Reason Rack Plugin Settings dialog
910
Browsing 910
Do not index these folders 910
User Interface 911
Reason Rack Plugin info 911
Rack Extensions and Content 912
Audio 912
Account and Authorization 912

Index 913

22 TABLE OF CONTENTS
Chapter 1
Introduction
 Welcome!
This is the Operation Manual for Reason Rack Plugin, part of the Reason Version 13 music production software from
Reason Studios. The information in this manual is also available as html files in the on-line Help system.
If you're using Reason mainly as plugin in another DAW host, this is the manual for you! If you're using Reason as a
standalone music application in itself, you should check out the main Reason 13 Operation Manual.
Also, be sure to regularly check out www.reasonstudios.com for the latest news!
! The information in this document is subject to change without notice and does not represent a commitment on
the part of Reason Studios.

Introducing Reason Rack Plugin

Reason Rack Plugin is an instrument and effect plugin in VST3, AUv2 (Mac) and AAX formats. Reason Rack Plugin
is installed when you install the standalone Reason program.
Like the standalone Reason 13 application, Reason Rack Plugin works on two platforms: Windows 10 (64-bit) and
macOS 10.15 (Catalina) (64-bit) or later. All functions are the same. If your DAW host supports cross-platform func-
tionality (saving a project on one platform and opening it on another), Reason Rack Plugin will open and work the
same on both platforms (provided of course that Reason is installed on both computers).
The screenshots in this manual were taken from both platform versions of Reason Rack Plugin. Since the layout is
more or less identical in these versions, there shouldn't be any problem following the instructions.

Conventions in the manual

This manual describes both the Windows and macOS versions of Reason Rack Plugin; wherever the versions differ
this is clearly stated in the text.

Text conventions
The text conventions are pretty straightforward. The examples below describe when certain text styles are used:
D This style instructs the user to perform the task(s) described in the sentence.
! This text style means IMPORTANT INFORMATION. Read carefully to avoid problems!
q This text style is used for tips and additional info.

Key command conventions

In the manual, computer keyboard commands are indicated with brackets. For example:
D Hold down [Shift] and press [C].
However, some modifier keys are different on Windows and Mac computers. Whenever this is the case, the manual
separates the commands with “(Win)” and “(Mac)” indications as in the following example:
D Hold down [Ctrl](Win) or [Cmd](Mac) and press [C] to copy.

References to context menus

Whenever the manual instructs you to select an item from the “context menu”, it means that you should right-click (or
[Ctrl]-click if you’re using a Mac with single-button mouse) on the specific area, section or device, and then select the
item from the pop-up menu that appears - the context menu. The item list in context menus varies depending on
where in the application you click.

24 INTRODUCTION
 Frames and circles (call-outs)
I/O device

Player

In pictures throughout this manual there might be circles and/or rectangles highlighting certain areas or objects.
These are indicated by filled lines according to the examples in the picture above. Sometimes these highlighting
frames/circles might also be accompanied by descriptive texts. The different colors of the frames and texts are only
to enhance the contrast to the background pictures.

Dashed arrows

A dashed arrow in a picture indicates the directions in which the pointer (or other tool) should be dragged to perform
the desired operation. The example in the picture above shows in which directions (up and down) to drag the pointer
to change the knob’s setting.

25 INTRODUCTION
 About the Authorization system
! Note that the information below is for Reason 13 and later. The information applies both to Reason+ subscrip-
tions and to licensed Reason versions, unless noted.
Reason Rack Plugin is authorized in the same way as the standalone Reason application, and uses the same license
or Reason+ account.
• The core of the authorization system is your Reason Studios account (and license number, if you are using a li-
censed version of Reason 13).

Registering Reason (licensed version only)

To be able to run your licensed version of Reason Rack Plugin, Reason must be registered to your account on the
Reason Studios web site.
D If you purchased a Reason license in a music store, follow the instructions you received from the shop.
! If you purchased Reason directly from the Reason Studios Shop web site, the Reason license has already been
automatically registered to your user account and you can use Reason by running it on a computer connected
to the Internet.

Authorizing Reason Rack Plugin

! Note that Internet connection is initially required for the procedures described below.

1. Launch your DAW and create a Reason Rack Plugin instance.

2. Enter you Username/e-mail address and Password for your Reason Studios account and click the Log in but-
ton.
Reason Rack Plugin launches in Authorized Mode.

26 INTRODUCTION
 • The first time you log in to the "Welcome to Reason" login dialog, the program will automatically store your
user credentials.
The next time you create a Reason Rack Plugin instance, it will start without showing the login dialog. This also au-
tomatically enables off-line use, allowing Reason Rack Plugin to run even if you have no Internet connection. Rea-
son Rack Plugin will then run off-line for a month. This period is continuously renewed when you use Reason Rack
Plugin with an Internet connection. You do not have to do anything manually, except connecting to the Internet
once a month.

Logging out from Reason Rack Plugin

Logging out means removing the stored login credentials - do this if you’re not going to use Reason Rack Plugin on
that computer, or if you need to log in to another Reason Studios user account.
D In the Settings dialog in Reason Rack Plugin, click the “Log Out on Quit” button:

The next time you run Reason, you will need to enter your Username/e-mail and password in the "Welcome to
Reason" dialog again. Internet connection is then required again.

Authorizing a computer for long-term off-line use

In addition, if you have an Annual (yearly) Reason+ subscription or own a licensed version of Reason 13, you can au-
thorize your computer for long-term off-line use. This can be useful if you are traveling or working in a studio without
Internet access, etc. The long-term authorization allows for staying off-line on the authorized computer for one (1)
year. If you are a Reason+ subscriber you are limited by the subscription end date, though. In most situations, autho-
rizing a computer for long-term use is not necessary, though.

27 INTRODUCTION
 Authorizing Reason Rack Plugin for long-term off-line use
1. Open the Settings dialog.
2. Click the “Add long-term authorization” button:

• The authorization expiry date for your computer is shown in the Offline Authorization section.
• You can authorize up to three (3) computers for long-term off-line use.

Removing a long-term authorization for a computer

If you've already long-term authorized three computers and would like to authorize another computer, you will need to
remove the authorization for one of the computers first:
1. Launch your DAW and add a Reason Rack Plugin instance on one of the computers that already has a long-
term authorization.
2. Open the Settings dialog.

28 INTRODUCTION
 3. Click the “Remove long-term authorization” button:

• If you can no longer access any of your authorized computers, you need to contact support to have them re-
move the authorization(s). This can be done here.

What happens if an authorization expires?

• If a Reason Rack Plugin authorization should expire during a session, you can continue to work and save your
song as usual.
• The next time you open Reason Rack Plugin you will have to log in in the “Welcome to Reason” dialog again.

29 INTRODUCTION
 About Rack Extensions
Rack Extensions are additional devices that can be purchased or trialed from the Reason Studios web shop. Rack
Extensions can be instruments, effects or utility devices, such as mixers and CV processors. Rack Extension devices
are developed by Reason Studios as well as by 3rd party companies.
Once installed, Rack Extensions will be available both in standalone Reason and in Reason Rack Plugin. In the pro-
gram or plugin, they behave just like built-in devices.
D To browse, trial or purchase Rack Extensions, visit reasonstudios.com/shop
D To download, install and manage Rack Extensions that you own, use the Reason Companion app, which can
be downloaded here.
You can open all these directly from the Rack Extensions section in the Settings dialog, see “Rack Extensions and
Content”.

The "Update Rack Extension Licenses" alert

Your Rack Extensions are authorized in the same way as Reason Rack Plugin. However, if you are opening the plugin
when connected to the Internet, you may get an alert asking you to "Update Rack Extension Licenses". This means
the Rack Extension license on your user account has changed, and no longer matches the license components on
the computer (for example if you have added new Rack Extensions to your account, or if Trial licenses have timed
out). Choose whether to update the license components on your computer or to run this session without Rack Exten-
sions.

Getting all your content

Reason and Reason Rack Plugin comes with some additional content (Rack Extensions and ReFills) that you can
download separately, either right after the Reason installation or at a later point. To download additional content, pro-
ceed as follows:
! Note that this procedure requires Internet connection!
1. Open the Reason Companion app and click the Devices tab.
2. At the top of the Devices tab, install the desired additional device(s).
Alternatively, click the “Install All” button to install all Rack Extensions in one go:

30 INTRODUCTION
 3. Then, click the “ReFills” tab and install the desired additional ReFill(s).
Alternatively, click the “Install All” button to install all ReFills in one go:

4. After installation, restart Reason Rack Plugin for the installed items to become available.
The additional content is stored on your computer as follows:
• The additional Rack Extension devices are stored in the following folders:
Windows: C:\Users\[your.user]\AppData\Roaming\Propellerhead Software\Optional REs
macOS: ~/Library/Application Support/Propellerhead Software/Optional REs
• The additional ReFills are stored in the “Music > Reason Studios > ReFills” sub-folder.

About automatic update checks

When you launch Reason Rack Plugin it automatically checks for new updates on the Reason Studios web site (pro-
vided that your computer has Internet connection). If a new update is found, an alert will be shown below the Global
Panel at the top of the Reason Rack. In the update bar you can choose to either download the update now or later:

If you click “Not Now” Reason Rack Plugin will wait a number of days before it checks for updates again.
If you click “Download” the update is downloaded to your computer via your default web browser.
1. Shut down Reason Rack Plugin and your host DAW.
2. Unzip and install the updated version of Reason and Reason Rack Plugin.
3. After installation, launch your DAW and add the updated Reason Rack Plugin.

31 INTRODUCTION
32 INTRODUCTION
Chapter 2
Overview
 Adding Reason Rack Plugin in your project
Reason Rack Plugin comes in two flavors: Reason Rack Plugin (for use as an instrument) and Reason Rack Plugin
Effect (for use as an audio effect, processing the sound from other instruments or audio tracks).
D Add Reason Rack Plugin instances to your project like you would with other VST plugins.
You can add as many instances of Reason Rack Plugin as your computer can handle.

The Reason Rack Plugin window

Global Panel

Device Palette

Rack

The Reason Rack Plugin has three main areas:

• The Global Panel
Here are buttons and functions that are global for Reason Rack Plugin.
• The Device Palette
The Device Palette is where you create devices. You can fold/unfold it by clicking the Devices button to the left in
the Global Panel.
• The Rack
The Rack is where you build your sound, by adding instrument and/or effect devices.

34 OVERVIEW
 The Reason Rack Plugin window can be resized vertically by dragging the lower window edge, which is quite useful
if you add many devices to your rack (and have a large monitor).

Scrolling in the rack is done by using scroll wheel, Page Up/Down buttons on your computer keyboard or by clicking
and dragging the side panels up or down.
Above the rack you'll find the Global Panel, which holds some important functions such as Show/Hide Device Pal-
ette, Show/Hide Browser, Undo/Redo, Flip Rack and a button for opening the Settings dialog:

Zoom
It’s possible to choose a suitable zoom level for the Reason Rack Plugin. This can be useful if you are using a very
high-resolution screen and want to make Reason Rack Plugin larger. Since all built-in rack devices and most Rack
Extensions are high-resolution, you will not lose any image quality when you enlarge Reason Rack Plugin.
D Click the Zoom button and select the desired Zoom factor:

! Note that the first time you select a new Zoom level, all devices in the song will recalculate their high-resolu-
tion panels in the background, which might take a little while.

About different Themes

In Reason Rack Plugin you can choose between two different visual themes (Light or Dark), i.e. how the user inter-
face should be visually presented. The selected theme affects the Browser, Global Panel, Settings dialog and Tutorial
area.
To select another Theme, click the Settings button and select an option from the Theme menu in the dialog that ap-
pears (see “Theme”). For the changes to take effect, you need to quit and relaunch your DAW host.

35 OVERVIEW
 Editing parameters
Since devices in Reason Rack Plugin are largely laid out like "real" hardware synths and effects, almost all controls
are designed like their real world counterparts - mixer faders, effect unit knobs, buttons, etc. How to adjust these con-
trols is described in the following paragraphs.

Knobs
D To “turn” a knob, point at it, hold down the mouse button and drag up or down (as if the knob was a vertical
slider).
Dragging upwards turns the knob clockwise and vice versa.

D If you press [Shift] and drag, the knob will turn slower, allowing for higher precision.
You can also adjust the knob precision with the “Mouse Knob Range” setting in the Settings dialog.
D To reset a knob to its default value (usually zero, center pan or similar), press [Ctrl](Win) or [Cmd](Mac) and
click on the knob.

Faders and sliders

D To move a fader or slider, click on the handle and drag in the fader/slider direction.

D You can also click anywhere on the fader/slider to instantly move the handle to that position.
D If you press [Shift] and drag, the fader/slider will move more slowly, allowing for higher precision.
D To reset a fader/slider to its default value (usually zero, 100, center pan or similar), press [Ctrl](Win) or
[Cmd](Mac) and click on the fader/slider handle.

Buttons

Many functions and modes are controlled by clicking buttons. Many of the buttons in Reason have a “built-in” LED, or
the button itself lights up, indicating whether the button is on or not.

36 OVERVIEW
 Fold/Unfold buttons
Fold/Unfold buttons are distinguished by a small triangle at the top to the left on a device. Clicking on a Fold/Unfold
button will unfold the device panel so that more controls are visible and can be accessed for editing on the screen.
On some devices, such as the RV7000 Advanced Reverb, there are more than one Fold/Unfold button. Clicking on
the second Fold/Unfold button on the unfolded front panel will open up the Remote Programmer panel from which
more parameters can be accessed:

Click on the Fold/

Unfold Button to unfold
the front panel.
Click on the second Fold/
Unfold Button on the
unfolded panel to bring up
the Remote Programmer.

The Fold/Unfold buttons on an RV7000 Advanced Reverb device.

• Pressing [Alt](Win) or [Option](Mac) and clicking a Fold button will fold or unfold all devices in the rack.

Multi Mode selectors

Some parameters allow you to select one of several modes. There are two different graphical representations of this
in Reason.
The multi mode selector type below consists of a button with the different modes listed above it:

D Click the button to step through the modes or click directly on one of the modes printed on the panel, or click
on the corresponding LED, to select mode.
The currently selected mode is indicated by a lit LED.
The multi mode selector type below is a switch with more than two positions:

D To change mode, click and drag the switch, or click directly at the desired switch position (just as when adjust-
ing a slider).

37 OVERVIEW
 Numerical controls
In Reason devices, numerical values are often displayed in numerical displays with “spin controls” (up/down arrow
buttons) on the side. Some parameter values, such as oscillator and LFO waveforms, are displayed graphically in the
displays. There are two ways of changing values in these types of controls:

or

D By using the up and down buttons on the spin controls.

To adjust a value in single steps, click on its up or down arrow button. To scroll a value continuously, click on an ar-
row button and keep the mouse button depressed.
D By clicking and holding the mouse button depressed in the actual display and then dragging the mouse up or
down.
This allows you to make coarse adjustments very quickly.

Menus and alpha-numeric controls

In Reason, alpha-numeric values and/or device presets are displayed in alpha-numeric readouts with “spin controls”
(up/down arrow buttons) on the side. There are two ways to change alpha-numeric/preset values:

or

D By using the up and down buttons on the spin controls.

To scroll continuously, click on an arrow button and keep the mouse button depressed.
D By clicking and holding the mouse button depressed in the actual alpha-numeric display and selecting from
the list that appears.
This allows you to make coarse adjustments very quickly or to immediately change to a preset anywhere in the list.
q This type of control is used to select, e.g., patch and reverb algorithms and some oscillator waveforms.

Tool Tips
If you hover with the mouse over a control on a device panel and wait a moment, a tool tip appears. The tool tip shows
the name of the parameter associated with that control and its current value. This helps you fine-tune settings, set
several parameters to the same value, etc.
D You can turn off the Tool Tips function by deactivating the option “Show parameter value tool tip” in the
Settings dialog (see “Show parameter value tool tip”).

38 OVERVIEW
 Context menus
Context menus are “tailored” to contain only menu items that are relevant to the current circumstances. Using the
various context menus allows you to work more quickly and more efficiently with Reason Rack Plugin.
D To bring up a context menu, right-click on the desired object, section or area in Reason Rack Plugin.
If you're using a Mac with a single-button mouse, press [Ctrl] and click.
The contents of the context menus depend on where you click. These are the primary types of context menus you will
encounter in Reason:

Parameter context menus

Depending on the host DAW implementation there may or may not be parameter context menus. If available, the host
DAW defines what items are available.

Device context menus

If you context-click anywhere on a device in the Rack (but not on a parameter or display), the context menu will con-
tain the following items:
• Undo and Redo functions.
• Cut, Copy, Paste, Delete and Duplicate Device items, allowing you to rearrange and manage the devices in the
rack.
• Submenus for creating new devices (Instruments, Effects, Utilities or Players).
• Combine and Uncombine are used when you want to use the selected device in, or exclude it from, a Combina-
tor setup.
• Auto-routing and Disconnect functions.
This allows you to automatically route (connect) or disconnect a selected device in a logical way.
• Reset Device resets all parameters to their default values and removes any loaded samples.
• If the device supports patches, the menu contains functions for copying, pasting and browsing patches.
• At the bottom of the menu are Flip Rack and Hide Cables items.
These work the same as the buttons on the Global Panel at the top of the rack.
• Additional device-specific items.
There may also be additional device-specific functions, for managing samples, handling patterns and more.

Empty rack context menu

If you click in an empty area of the rack, the context menu will contain the following items:
• Undo/Redo.
• A Paste item, allowing you to paste any copied or cut devices.
• Functions for creating new devices.
• Flip Rack and Hide Cables.

39 OVERVIEW
 Undo and Redo
While virtually all DAW hosts have Undo and Redo functions, many don't allow you to undo changes done within
plugins. This means that you might create a plugin instrument, change some parameters in the plugin and select
undo - only to have the program remove the plugin you created in the first step. The parameter changes aren't part
of the DAW host's undo history.
To avoid this, Reason Rack Plugin has its own Undo and Redo functions. These are available as buttons on the top
Global Panel, and as functions on the context menus:

! Each instance of Reason Rack Plugin has its own Undo history.

40 OVERVIEW
Chapter 3
Audio and MIDI Basics
 General audio and MIDI handling
Reason Rack Plugin doesn't communicate directly with your audio or MIDI hardware. Instead, this is handled by your
DAW host, which in turn passes on MIDI or audio to Reason Rack Plugin and gets audio back in return.
A Reason Rack Plugin instance can:
• Receive MIDI notes and other messages from the DAW host. It does not care about MIDI channels.
• Receive up to four audio channels (two stereo input pairs).
Typically, Reason Rack Plugin receives audio when used as an audio effect, but it's also possible to send audio to
an instrument (if your DAW permits this), for sidechaining and other effects.
• Send out up to 32 audio channels (16 stereo pairs) to the DAW host.
In most cases, the output will be a single stereo signal, but you may for example want to route different drum
sounds to different outputs for processing on separate channels in your DAW host's mixer.
• Reason Rack Plugin can also send MIDI to other tracks in your DAW - if you use the MIDI Out Device, see “MIDI
Out Device”.

Typical input/output configurations

Below are some examples of typical MIDI and audio configurations in Reason Rack Plugin:

Stereo instrument device

This setup involves MIDI note input sent to an instrument device, and stereo audio output sent from the instrument
device:
Reason Rack Plugin output signals

AUDIO IN
1-2 3-4 5-6 7-8 9-10 11-12 13-14 15-16 MAIN SIDECHAIN FADER

I/O MAIN OUT TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN AUDIO OUT
DEVICE
SHUFFLE L R

MUTE SOLO

MIDI Out from

Master Keyboard ID8 audio output signals
Grand A Delay
ID8 1

Piano Upright
Dance
B
C
Vibes D Chorus instrument device
PITCH MOD VOLUME
56

MIX CHAN...

Reason Rack Plugin

DAW Mixer
Live and playback MIDI data from Instrument Track in sequencer

MIDI In ID8 1

M S M

DAW Sequencer
Stereo instrument example.

42 AUDIO AND MIDI BASICS

 Multi-channel instrument device
This setup involves MIDI note input sent to an instrument device, and audio sent out from multiple audio outputs of
the instrument device. A typical scenario would be a drum machine device with multiple drum channels:
Reason Rack Plugin output signals

AUDIO IN
1-2 3-4 5-6 7-8 9-10 11-12 13-14 15-16 MAIN SIDECHAIN FADER FADER FADER FADER

I/O MAIN OUT TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN AUDIO OUT
DEVICE
SHUFFLE L R L R L R L R

MUTE SOLO MUTE SOLO MUTE SOLO MUTE SOLO

MIDI Out from

Master Keyboard DRM audio separate outputs
DRM 1

DRM
Reverb

DRUM KIT 1
Compression - - - -
VOLUME DRUM MACHINE
56 56 56 56

KICK SNARE HHCL HHOP TOM1 TOM2 TOM3 CYM MIX CHAN... MIX CHAN... MIX CHAN... MIX CHAN...

DAW Mixer
Reason Rack Plugin

Live and playback MIDI data from Instrument Track in sequencer

MIDI In ID8 1

M S M

DAW Sequencer
Multi-channel instrument example.

Stereo audio effect device

This setup involves a stereo audio input signal to an effect device, and stereo audio output from the effect device:

Reason Rack Plugin output signals

Input signals to Reason Rack Plugin

AUDIO IN
1-2 3-4 5-6 7-8 9-10 11-12 13-14 15-16 MAIN SIDECHAIN FADER FADER FADER FADER

I/O MAIN OUT TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN AUDIO OUT
DEVICE
SHUFFLE L R L R L R L R

MUTE SOLO MUTE SOLO MUTE SOLO MUTE SOLO

FX audio out
FX audio in
ECO 1

E:C:O
Reverb

ECHO CHAMBER
Delay
VOLUME MULTI FX MIX CHAN... MIX CHAN... MIX CHAN... MIX CHAN...

FX 1 FX 2 FX 3 FX 4

DAW Mixer
Reason Rack Plugin Effect
Stereo effect example.

43 AUDIO AND MIDI BASICS

 Stereo audio effect device with sidechain
This setup involves a stereo audio input signal to an effect device - plus a sidechain audio input signal to the effect
device - and stereo audio output from the effect device. A typical scenario would be a stereo compressor device with
sidechain inputs.
Reason Rack Plugin output signals

Input signals to Reason Rack Plugin

Sidechain signals to Reason Rack Plugin

AUDIO IN
1-2 3-4 5-6 7-8 9-10 11-12 13-14 15-16 MAIN SIDECHAIN FADER FADER FADER FADER

I/O MAIN OUT TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN TO MAIN AUDIO OUT
DEVICE
SHUFFLE L R L R L R L R

MUTE SOLO MUTE SOLO MUTE SOLO MUTE SOLO

12 12 12 12

FX audio out
FX audio in SC in
DUCK 1

Duck
Threshold

Ratio
VOLUME MULTIBAND COMPRESSOR MIX CHAN... MIX CHAN... MIX CHAN... MIX CHAN...

DAW Mixer
Reason Rack Plugin Effect
Stereo effect with sidechain example.

44 AUDIO AND MIDI BASICS

 The I/O device

The I/O device is always located at the top of the rack.

At the top of the rack is the i/o device (for "input/output"). This handles the audio communication between the de-
vices in the rack and the DAW host.
The input jacks deliver audio from the DAW host to devices in the rack. This is most often the case when Reason
Rack Plugin is used as an effect, and typically only the main (1-2) input jacks are used.
The output jacks deliver audio from devices in the rack to the DAW host. These are used both when Reason Rack
Plugin is an instrument and an effect.
On the front panel you find audio input and output indicators, lighting up whenever audio signals are received from
the DAW host (input) or sent back to the DAW host (output).
The first eight output pairs have more detailed meters, name labels matching the device names and buttons called
(sum) “To Main" (outputs 3-16 only). When sum To Main is on for an output pair, its signal is directed to the Main Out
(1-2) instead, and summed with any other signals there.
• If you want to layer several instrument devices on a single stereo channel in your DAW, leave "To Main" on.

The three instruments (connected to outputs 1-2, 3-4 and 5-6) will all be sent out on Main Out 1-2 to the DAW.
• If you want different devices in your rack to be routed to different audio channels in your DAW's mixer, turn off
"To Main" for these outputs.

The two instruments connected to outputs 1-2 and 3-4 will be sent out on Main Out 1-2 to the DAW. The instrument connected to
outputs 5-6 will be sent to the separate Out 5-6 outputs.

45 AUDIO AND MIDI BASICS

 Audio settings
Since Reason Rack Plugin doesn't communicate directly with the audio hardware, audio settings like sample rate are
all made in the DAW host. There is however one audio setting in the Settings dialog:

Render audio using host buffer size setting

When this is activated (default) all audio rendering will be done in batches corresponding to the buffer size selected
in the DAW host's audio settings. Selecting a higher buffer size there will improve the performance of Reason Rack
Plugin. However, if your rack contains feedback routings, these will be delayed with higher buffer sizes.
Turning this off will cause the plugin to render audio in batches of 64 samples (like in older versions of Reason). Use
this only if you want to minimize delay in feedback routings in Reason Rack Plugin.

About Plugin Delay Compensation

There is no delay compensation made in the internal device routings of the Reason Rack Plugin instance itself. How-
ever, the summed delay in the signal chain to the Main Outputs 1-2 is reported to the host DAW, to allow for delay
compensation against other tracks or channels.

46 AUDIO AND MIDI BASICS

Chapter 4
Using Reason Rack
Plugin as an Instrument
 Creating an instrument

When you add Reason Rack Plugin as an instrument the plugin window opens. When the rack is empty, an overlay is
shown with icons of the most popular instrument devices. Either:
D Double click an instrument icon to add that instrument,
D click "Browse Instruments" to open the Browser with “Add Instrument” focus,
D or click "Add other device" to add another device from the context menu that appears.
When you have added a device, the popular devices overlay goes away - it is only shown when the rack is empty.
Note that the list of popular devices will change over time to include the instrument devices you most often add!
When an instrument device is added, it will automatically receive MIDI input from the track in the DAW host - you
should be able to play it from your MIDI keyboard right away.
The output of the added instrument device is auto-routed to the first available output jack on the i/o device at the top
of the rack.

48 USING REASON RACK PLUGIN AS AN INSTRUMENT

 1. Click the arrow to the left to unfold the I/O device.

2. Click the Flip Rack button on the Global Panel to see the jacks on the back:

A Thor instrument automatically connected to Main Out (1-2) of the I/O device.

Browsing for patches

Most instrument devices come with patches for quickly changing sounds. There are three main ways to select a
patch for a device:
D Use the patch name display on the device panel.
Either click and select a patch from the menu or use the previous/next patch buttons next to the display:

D Click the Browse Patches button on the device (or select Browse Patches from the context menu).

This opens the Browser with Browse Focus for the device, where you can search and navigate the sound banks
and folders on your hard drive. Read more in the “Sounds, Patches and the Browser” chapter.
D Select a device and click the Browse icon on the global panel.

This opens the Browser with Browse Focus for the device, where you can search and navigate the sound banks
and folders on your hard drive. Read more in the “Sounds, Patches and the Browser” chapter.

49 USING REASON RACK PLUGIN AS AN INSTRUMENT

 Adding effects
After having created an instrument and selected a sound for it, you might want to add one or several effects. You can
either do this from the Browser or by right clicking the instrument device panel and selecting an effect device from
the Effect Devices sub-menu:

50 USING REASON RACK PLUGIN AS AN INSTRUMENT

 The effect device is automatically connected, so that the audio from the instrument is routed through the effect:

Adding more effects will connect them in series. You can always click Flip Rack to see how devices are connected on
the back side of the rack. Read more in the “Routing Audio and CV” chapter about how to do manual routing of sig-
nals, for more complex effect setups!

51 USING REASON RACK PLUGIN AS AN INSTRUMENT

 Layering instruments
Layering two or more instrument devices is a quick way to create thicker or more complex sounds. To do this in Rea-
son Rack Plugin, simply add another device below the first one (either by clicking Add Device at the bottom of the
rack or by using the Browser).
The new instrument devices will be automatically routed to the first free output on the I/O device at the top of the
rack:

By default, sum To Main is activated, which means the instruments will be mixed with the first one on the main stereo
output to the DAW:

All instruments in the rack will get the same MIDI input, i.e. when you play your MIDI keyboard, all instruments will re-
ceive notes and be heard at the same time. For more advanced layering techniques, you can use the Combinator de-
vice (see “The Combinator”).

52 USING REASON RACK PLUGIN AS AN INSTRUMENT

 Using separate audio outputs
If you have layered instruments as described above but want to send them to separate mixer channels in your DAW
host, simply turn off sum “To Main” for their outputs on the I/O device. This will send each instrument device to a sep-
arate output, and your DAW will be able to receive them on individual channels.
Some instrument devices have multiple outputs themselves. For example, a drum module may have separate outputs
for each drum. To route these to individual channels in your DAW host:
1. Click the Flip Rack button to show the backside of the rack.
2. If the I/O device is folded, click its arrow button to unfold it so you can see the output jacks.
3. Scroll down to the instrument device.
4. Click on a separate output jack on the device, and drag up to the desired output jack on the I/O device.
A cable is shown when you drag.
5. Release the cable on the jack.
It is now connected. If the output was in stereo and you dragged the left channel, the right channel is automatically
connected as well.
6. Repeat the connection procedure for some more separate output pairs:

7. Flip the rack around again and make sure sum To Main is turned Off for the separate output pairs.
This sends the signals to the DAW on a separate output channels instead of summed to the main output:

Depending on the instrument, you may also need to make settings on the device itself to assign sounds to that sep-
arate output etc. See the documentation for the instrument device.

53 USING REASON RACK PLUGIN AS AN INSTRUMENT

 Adding Players
Players are special devices that transforms or generates MIDI notes and passes them on to an instrument device. A
Player can for example create chords or arpeggios from the notes you play, or generate notes without input, like a se-
quencer or drum machine.
Players are added from the Device Palette or from the Player Devices item on the instrument panel context menu.
They are listed on a separate palette or submenu.

54 USING REASON RACK PLUGIN AS AN INSTRUMENT

 A Player is always added to an instrument device - typically you first select the instrument and then add the Player. In
the rack, it sits on top of the instrument device, intercepting incoming MIDI notes and passing them on, transformed
in various ways. You can also chain/stack multiple Players:

A Scales & Chords Player in series with a Note Echo Player, controlling an NN-XT sampler instrument.
Most Players have On buttons. When turned off, they will bypass MIDI as if they were not connected at all. There is
also a Bypass All button at the top - this bypasses the whole chain of Players.
Read more about the included Player devices in “Working with Players”.
! Note that Reason Rack Plugin doesn't send MIDI back to the DAW host. You cannot use a Player as a general
MIDI effect for controlling another instrument plugin in the project.

55 USING REASON RACK PLUGIN AS AN INSTRUMENT

 Using Mixer devices
Although the sum To Main function makes it easy to layer several instrument devices, you may want more control
over how they are mixed. This is best achieved by adding a Mixer device in the rack before you add your instruments:
1. In an empty rack, click the "Add Other Device" button and select the Utilities submenu.
2. Under Reason Devices, you'll find the Mixer 14:2 and Line Mixer 6:2 - select one of these to add it in the rack.

The output of the Line Mixer 6:2 is automatically connected to the I/O device.
3. When you now add instrument devices, they will be routed to the Mixer inputs, instead of to the I/O device:

An ID8 instrument and a Monotone instrument added to the rack and auto-routed to the 6:2 Line Mixer.
A Mixer allows you to balance levels of the instruments, mute or solo them and use the Pan controls to place different
instruments in different parts of the stereo image. You can also add Send Effects, and the Mixer 14:2 has a basic EQ.
Read more about the mixer devices in “Mixer 14:2” and “The Line Mixer 6:2”.

56 USING REASON RACK PLUGIN AS AN INSTRUMENT

 Detailed control over MIDI note input
In Reason Rack Plugin, all instrument devices receive MIDI notes (as well as a few effect devices and utilities such as
the Combinator and RPG-8). However, sometimes you may not want all devices to receive MIDI notes, or you may
want to send MIDI notes to effect devices which normally don't receive them.
The solution is to select the devices and select "Combine" from the context menu. This puts the selected devices in-
side a Combinator. Devices inside a Combinator don't receive MIDI directly from the DAW host - instead, the Combi-
nator receives MIDI notes and distributes them to the devices within. You can use the "Receive Notes" setting in the
Combi Programmer section to control this for each device. This section also allows you to create keyboard and veloc-
ity splits. Read more about the Combinator in “The Combinator”!
The same is true for performance controls such as Mod Wheel, Pitch Bend, Aftertouch and Sustain Pedal. They are
normally only received by devices that receive notes, but there are separate settings in the Combi Programmer for
these controls as well.
q By creating a mixer device inside the Combinator and mixing instruments there, you create a self-contained
multi-instrument, which can be saved as a combi patch and opened in the standalone version of Reason.

57 USING REASON RACK PLUGIN AS AN INSTRUMENT

58 USING REASON RACK PLUGIN AS AN INSTRUMENT
Chapter 5
Using Reason Rack
Plugin as an Effect
 Creating an effect

When you add Reason Rack Plugin as an effect the plugin window opens. When the rack is empty, an overlay is
shown with icons of the most popular effect devices. Either:
D Double click an icon to add that effect,
D click "Browse Effects" to open the Browser with “Add Effects” focus,
D or click "Add other device" to add another device from the context menu that appears.
When you have added a device, the popular devices overlay goes away - it is only shown when the rack is empty.
Note that the list of popular devices will change over time to include the instrument devices you most often add!

60 USING REASON RACK PLUGIN AS AN EFFECT

 The first effect device you add will be automatically connected between the Main Input jacks on the I/O device and
the Main Output jacks. Audio sent from the DAW host will pass through the effect and be sent back to the DAW:

A The Echo effect device automatically connected to Main In (1-2) and Main Out (1-2) of the I/O device.
When nothing at all is connected to the I/O device, Reason Rack Plugin Effect will send any incoming audio back to
the DAW host (as if the rack was bypassed). This way you can still hear your audio when you add an empty rack.

Browsing for patches

Most effect devices come with patches for quickly changing sounds. There are two main ways to select a patch for a
device:
D Use the patch name display on the device panel.
Either click and select a patch from the menu or use the previous/next patch buttons next to the display:

D Click the Browse Patches button on the device (or select Browse Patches from the context menu).

This opens the Browser, where you can search and navigate the sound banks and folders on your hard drive. Read
more in “Sounds, Patches and the Browser”.
D Select a device and click the Browse icon on the global panel.

This opens the Browser with Browse Focus for the device, where you can search and navigate the sound banks
and folders on your hard drive. Read more in the “Sounds, Patches and the Browser” chapter.

61 USING REASON RACK PLUGIN AS AN EFFECT

 Creating effect chains
• Adding more effects will automatically connect them in series.
• If you press [Shift] and drag an effect up or down in the rack, it will be automatically re-routed, changing the or-
der of effects in the signal chain.
Note that this can change the sound drastically!
It is also possible to do more complex setups such as splits and parallel effect chains. For this you could use devices
such as the Spider Audio Merger & Splitter (see “Spider Audio Merger & Splitter”). This would require you to do some
manual routing by dragging cables on the back of the rack. Read more in “Manual routing”.
All effect devices in Reason Rack Plugin have a Bypass/On/Off switch:

This is normally set to On, but setting it to Bypass lets you temporarily disconnect the effect. Setting an effect to Off
will silence it completely (no sound will be passed through). This is mainly useful if you are using send effects or par-
allel effect chains.
Once you have an effect chain that you're happy with, you could Combine it (by selecting all devices and selecting
Combine from the context menu). This creates a Combinator with all devices. You can save this as a combi patch and
load it into other instances of Reason Rack Plugin or the standalone version of Reason, see “The Combinator”.

Using sidechain inputs

Reason Rack Plugin has four audio input channels: Main left + right and Sidechain left + right. This allows you to
send an additional stereo audio signal into the rack (provided that your DAW host supports this). How you use this is
really up to you, but a common usage would be sidechaining a compressor:
1. Open Reason Rack Plugin as an effect.
For this example, you should add it as an insert effect on a synth pad track or similar.
2. In the rack, add a compressor device with a sidechain input, e.g. the M-Class Compressor.
3. Flip the rack.
4. Click the Sidechain input on the Compressor and drag to connect it to the Sidechain In on the I/O device at the
top of the rack.
(If the I/O device is folded, just hold the cable over the folded device for a moment - it will unfold automatically to
allow you to connect it.)

Connecting the Sidechain inputs of the MClass Compressor.

62 USING REASON RACK PLUGIN AS AN EFFECT

 5. In your DAW host, route another channel or audio track to Reason Rack Plugin input 3-4 (Sidechain input).
Typically, you would use something like a kick drum loop for this.
6. Start playback of your synth pad and kick drum tracks.
The synth pad will be processed by the Compressor, but the Compressor will be triggered by the sidechain signal.
7. Adjust the settings on the M-Class Compressor to get a classic, pumping sidechain pad.

63 USING REASON RACK PLUGIN AS AN EFFECT

64 USING REASON RACK PLUGIN AS AN EFFECT
Chapter 6
Working in the Rack
 Creating devices
Devices can be created in a number of different ways:
D Double click an instrument icon to add that device,
D Click "Browse Instruments" or “Browse Effects” to open the Browser with “Add Instrument” or “Add Effect” fo-
cus.
Now a device will be automatically created when load a patch.
D or click "Add other device" to add another device from the context menu that appears.

D If there already are devices in the rack, click the Add Device button (or context-click in the rack) and select a
device from the menu:

66 WORKING IN THE RACK

 D Select a device from the context menu or select any of the “Add...” items and then select a patch in the Browser
and click OK (or double click a patch).
• Drag and drop a device from the Device Palette or drag and drop a patch from the Browser to the Rack.
As you drag a device or patch to the rack, a +-sign is shown together with an orange divider to indicate where the
device will be placed:

Adding a Polytone Dual-Layer Synthesizer device by dragging from the Instruments device palette and dropping in the rack.
! When using drag and drop, pay attention to where you drop the device:
• If you drop a device on top of an existing device in the rack, you will replace it.
• Dropping a patch on top of a device means loading the patch (and possibly replacing the device), while drop-
ping in the empty rack or below devices means creating a new device with that patch loaded.

The Device Palette

The Show/Hide Device Palette button at the top of the Reason song window.

67 WORKING IN THE RACK

 D Click the Devices button to bring up the Device Palette (or to hide it).
The devices are sorted in four different groups in the Device Palette: Instruments, Effects, Utilities and Players. The
Utilities group contains device types such as mixers, modulation devices, etc. In each device type group the devices
are listed in the following sections: Reason Devices, and then by Rack Extension manufacturer in alphabetical order.
The Players group is a little special; it contains a number of MIDI “generators” that can be used when recording In-
strument tracks in the sequencer, see “Working with Players”.
D To show all devices and device types in one single list, turn off all group buttons.

Searching and filtering in the Device Palette

It’s possible to search for devices, manufacturers and device descriptions, and to apply filters to specify what type of
devices you want to display in the Device Pallet groups.
D Set the focus to the search/text field at the top of the Device Palette.
D Type in (part of) the device name, or the device description, or the manufacturer you are searching for.
In this example we type in “kon” in the Search box and get the following results:

Typing in “kon” results in Kong and Konarie Chirp Synthesizer (3rd party Rack Extension device).
D To clear the text in the Search box, click the x button to the right (or press [Backspace] repeatedly).

68 WORKING IN THE RACK

 D Click the unfold button to the left of the Search field.
Then, select device types/characteristics by selecting one or several of the Categories:

Clicking the “Bass” Category displays devices suitable for bass sounds.
D You can also use the Search and Categories in combination, if you like.
The functions work exactly the same regardless of what device group (Instruments/Effects/Utilities/Players) you
have selected.

Selecting devices
D To select a single device, click on it in the rack.
The selected device is displayed with a colored border (based on the color scheme selected for your operating
system).

D To select several devices, hold down [Ctrl](Win) or [Cmd](Mac) and click on the desired devices.
D Hold down [Shift] and click to make a continuous (range) selection.
D To de-select all devices, click in the empty part of the rack.

69 WORKING IN THE RACK

 Moving devices
A device can be dragged freely up and down in the rack without affecting the routings.
In this example an RV-7 reverb device is moved to two different positions:

In this case, the line

indicates that the RV-7
reverb device will be
placed to the left of the
phaser.

This is the result. Note

that the filter device is
moved to the left, to fill
out the gap.

In this case, the line

indicates that the reverb
device will be placed to
the right of the chorus/
flanger.

This is the result. All three

devices are moved to the
left, to fill out the gap.

Re-routing devices
D If you hold [Shift] and drag a device to a new position in the rack (as described above), it will be re-routed (as
if you deleted it and created it in its new position).
This allows you to e.g. change the order of effect devices in a signal chain by Shift-dragging them.
See “Auto-routing” for more info on auto-routing.

70 WORKING IN THE RACK

 Deleting devices
D To delete a device, right-click it and select Delete.
If the deleted device was part of a chain, the signal chain will be kept.
• You can also select multiple devices and then select Delete to remove them all in one go.
• Deleting all devices in the rack will make the Popular Devices palette show.
! The I/O device is fixed to the top of the rack and cannot be deleted.

Replacing devices
D Drag and drop a device on top of an existing device in the rack, to replace it.
When dragging a device on top of another device in the rack, the panel of the existing device is shaded in orange:

Replacing a Subtractor device with a PX7 FM Synthesizer device by dragging from the Instruments device palette and dropping on
the Subtractor.
D You can also drag a patch from the Browser and drop on a device to replace the device and load the dropped
patch in it.

Cut, Copy, Paste and Duplicate devices

These functions on the context menu affect the currently selected devices.
q Note that you can also copy and paste devices between different Reason Rack Plugin instances in your DAW
project!

Naming devices
Each device has a "tape strip" which shows the name of the device. Normally, this is the name of the loaded patch (or
the device type if it doesn't support patches), but you can rename it by clicking the tape strip and typing. This is es-
pecially useful if your rack contains several devices of the same type and you need to separate them.

D To revert to the default patch name, double click the tape strip and delete your custom name.
The device name is also shown on the I/O device, if it's connected to one of the first 8 input pairs.

71 WORKING IN THE RACK

72 WORKING IN THE RACK
Chapter 7
Routing Audio and CV
 Working on the back of the rack
As you've seen in the previous chapters, instrument and effect devices are connected automatically when you create
them. This means you don't strictly have to do any manual signal routing to use Reason Rack Plugin, at least not for
standard instrument and effect functionality.
However, adding some manual routing vastly opens up the possibilities! Here are some of the things you can do:
• Connecting multiple outputs from instruments.
• Sidechaining effects.
• Splitting, merging and mixing signals, creating parallel signal chains, send effect structures and more.
• Using CV (Control Voltage - see “CV/Gate signals” below!) to modulate or control parameters on one device
from another.
• Playing instruments from CV sequencers and arpeggiators.
• Using instrument devices with audio input as effects.
All this routing is done on the back side of the rack:
D Click Flip Rack to get there (or select Flip Rack from the context menu):

All audio and CV connections are represented by cables.

74 ROUTING AUDIO AND CV

 Hiding cables
Sometimes, there can be quite a lot of cables in view, making the connections hard to follow. Then the Hide Cables
function is handy:
D Click Hide Cables on the Global Panel to toggle this on or off (or select Hide Cables from the context menu).

The result of Hide Cables depends on this setting in the Settings dialog:

Option Result
Hides auto-routed cables. Only cables you have connected manually will be fully shown. Cables that were
connected automatically are drawn semi-transparent.
Shows cables for selected devices only. Only cables connected to the currently selected device(s) are fully shown. Other
cables are drawn semi-transparent.
Hides all cables. All cable connections are indicated with colored dots in the jacks, and no cables
will be shown.

75 ROUTING AUDIO AND CV

 Signal types
The following signal types are used in Reason Rack Plugin:

Audio signals
Audio means sound being sent from one device to another (or to/from your DAW host).
• Audio connectors are shown as large quarter inch jacks and the cables are thick:

• Audio connectors can be either inputs or outputs, as indicated on the panel.

• You always route cables between inputs and outputs (it doesn't matter in which order you route).
The special case is the i/o device at the top of the rack, where the "Inputs" represent the inputs from the DAW
host and the "Outputs" represent the outputs to the DAW host. Thus, an instrument will have its own Output jacks
connected to the "Outputs" on the i/o device, to get the sound back into your DAW host.
• Many devices have audio jacks in stereo pairs, with one labeled e.g. "Left/Mono" and the other "Right".
When you connect such a Left output, the Right is typically automatically connected too - but you disconnect it to
use a mono signal from the device.

CV/Gate signals
In the early days of synthesizers, before the MIDI protocol was invented, analog synthesizers could be interconnected
using Control Voltage (CV) cables. For example, one cable would be used for controlling pitch while another would
send a Gate voltage, basically telling a synth when to play a note and when to stop. A third cable might send a mod-
ulating signal to some parameter, e.g. varying the filter frequency. Today, this system has become quite common
again, thanks to the rise of modular synthesizers.
The CV signal cables in Reason Rack Plugin emulate this analog control system. CV cables send a value, which may
be static or changing. They do not carry audio, but are used for modulating parameters and controlling devices.
• CV connectors are shown as smaller mini jacks, and the cables are thinner than the audio cables:

• CV connectors can be either inputs or outputs, as indicated on the panel.

You always route cables between inputs and outputs (it doesn't matter in which order you route).

76 ROUTING AUDIO AND CV

 • Connectors may be labelled "Gate".
A Gate signal is a CV signal that goes from zero to an "on" value and eventually back to zero again. They are used
for playing notes, triggering envelopes and more.
• Instrument devices often have a "Note" CV input, along with a "Gate" input.
These are used in tandem, to play the instrument from a CV source such as the Matrix rack sequencer. The Gate
signal will determine when notes start and stop, while the Note CV signal will set the pitch of the played notes.

About CV Trim knobs

Most CV inputs have an associated Trim knob. This is used to set the CV "sensitivity" when modulating a parameter.
The further clockwise a CV trim knob is set, the more pronounced the modulation effect.
• Turned fully clockwise, the modulation range will be 100% of the parameter range.
• Turned fully anti-clockwise, no CV modulation will be applied.

About MIDI routing

MIDI signals (notes and controllers such as pitch bend, mod wheel etc) are not represented by cables in Reason
Rack Plugin. Normally, all devices that use MIDI notes and performance controllers will automatically receive them
from the DAW host.
• To decide exactly which devices should receive MIDI notes, put them in a Combinator and use the "Receive
Notes" setting in the Combi Programmer section for each device. See “Configuring the Combinator panel”.
• If a Player is added to an Instrument device, MIDI notes will go to the Player instead of the instrument.
The Player may pass on the notes, transform them or generate new notes, depending on the Player type and set-
tings.
• You can also control individual parameters with MIDI controller messages (e.g. if you have a MIDI keyboard
with knobs and faders).

Manual routing
There are two ways to manually connect an output jack to an input jack (or vice versa):

Dragging cables
1. Click the jack, and keep the mouse button pressed.
A cable appears.
2. Drag the cable to the other jack.
When you're over a jack of the correct type, it lights up.
3. Release the mouse button.
The cable is connected (replacing the existing connection there, if any). If you dragged from the left jack in a ste-
reo pair, the right will automatically be connected as well.
D To change a connection, click and hold the jack to grab the cable. Then drag it to another jack.
D To disconnect a cable, click the jack at either end to grab the cable and drop it away from any jack.

77 ROUTING AUDIO AND CV

 Using the routing menu
1. Right-click a jack.
A context menu appears, listing all devices in the rack:

2. Move the mouse pointer to the device you want to connect to.
A submenu lists all outputs or inputs on that device. An asterisk (*) next to a jack means it's already connected.
3. Select the desired jack.
The two jacks are connected with a cable. If the jack was already in use, the old connection is replaced.
D To disconnect a jack, right-click it and select "Disconnect" from the context menu:

Checking and following cable connections

D If you point your mouse at a jack that's in use and wait a moment, a tool tip appears, showing both the device
name and connector that the cable is connected to:

D You can also right-click the jack and select "Scroll to Connected Device".
This will scroll to the device in the other end of the cable and highlight the connector briefly.

Auto-routing
Reason Rack Plugin will automatically route devices when you create them. If you don't want this, you can hold down
[Shift] when you create the device. This will add the device without connections, requiring that you route it manually
to use it.
It's also possible to invoke this automatic routing from the context menu, by right-clicking a non-connected device
and selecting "Auto-route Device".
To disconnect all cables going to and from a device, instead select "Disconnect Device" from the context menu.
When a device is auto-routed, either at creation or from the context menu, the following rules apply:
• An audio output will be connected to the first free and auto-routable input above it in the rack.
For example, if you auto-route an instrument and there's a rack mixer device above it in the rack, it will be con-
nected to the first free mixer channel input. If there are no such free, suitable inputs on devices above, it will be
routed to the I/O device instead.

78 ROUTING AUDIO AND CV

 • An audio input will be connected to the first auto-routable output above it in the rack.
If this was already connected, the device will insert itself into the signal chain but preserve the connection.

Auto-routing a non-connected The Echo device below a Thor instrument device.

• Devices inside a Combinator will not auto-route outside the Combinator device.

79 ROUTING AUDIO AND CV

 • A few CV devices will auto-route Note CV and Gate cables to the corresponding CV inputs on the first suitable
instrument device above in the rack.
For example, if you add a Matrix sequencer below a Thor synth, it will auto-route Note CV and Gate and be ready
for immediate playback:

80 ROUTING AUDIO AND CV

Chapter 8
Sounds, Patches and
the Browser
 About patches
A patch contains settings for a specific device. Patches can be either separate files on your hard disk or files embed-
ded in a ReFill (see “About ReFills” for info about ReFills). A Rack Extension often comes with patches embedded
within the Rack Extension itself.
• A patch most often includes all parameter settings on the front panel, but not cables and trim pot settings on
the back side.
However, a few Rack Extensions have additional parameters on the back panel which are included in the patches.
• A Combinator patch is special in that it contains both settings for the device itself (the Combinator settings)
and all settings for all contained devices.
This includes all routing within the combi, and all settings on the back panels as well.
• Patches for devices that use samples (samplers, loop players, drum machines) contain references to sample
files on disk.
The samples themselves are not part of the patches.
• Some devices don't have patch support at all.
Should you need to save the state of such a device, put it inside a Combinator device and save a combi patch!

About the “Load Default Sound in New Devices” setting

If this is activated in the Settings dialog, a default patch is loaded when a device is created. This way, the device is
ready for playing right away.
If you turn this setting off, new devices will be initialized - parameters are reset to their default values and no samples
are loaded in sample-based devices.

Loading patches
To select and load a patch for a device, use one of the following methods:
D Click the “Browse Patch” button in the Patch section on the device panel.
The Patch section has the same basic layout for all patch devices; a Patch Name display, two “Select Patch” but-
tons (up/down) for stepping through patches sequentially, a “Browse Patch” button to open the Browser, and a
“Save Patch” button to save patches.

Save Patch button

Browse Patch button

Patch Name display Select Patch buttons
The Patch section of the Subtractor device.
! On the panels of devices that can load samples/REX files, there are also Browse Sample/REX Loop buttons,
used for loading samples or REX files. Make sure you click on the button in the Patch section (next to the
patch name display) to load patches!

82 SOUNDS, PATCHES AND THE BROWSER

 When you click the Browse Patches button on a device, the device automatically gets “Browse Focus” and the
Browser opens. A button at the top of the Browser indicates that only patches for the highlighted device in the
rack are shown:

Browse focus for a device is indicated by orange side bars in the rack - and that the patch section on the device
is colored in orange:

The orange Browse Focus indicators in the rack.

D Alternatively, select a device in the rack and then click the Browse icon on the Global Panel:

This opens the Browser with Browse Focus on Patches for the selected device, as described above.
If several devices are selected in the rack, the Browser opens with Patch Browse Focus for the last selected de-
vice.
If no device is selected in the rack, the Browser opens without Browse Focus.
D Another way of loading patches is to select “Browse Patches...” from the context menu of a selected device.
Alternatively, hold down [Ctrl](Win) or [Cmd](Mac) and click [B].
q Once you have loaded a patch, you can step between all the patches in the same folder by using the “Select
Patch” buttons on the device panel:

D If you click on the Patch Name display on the device panel, a pop-up menu will appear, listing all patches in the
currently selected folder.
This allows you to quickly load another patch, without having to step through the patches one by one. You can also
choose to set browse focus to the device by selecting “Browse...” from the patch display pop-up menu.
When you load a patch in any of the ways described above, the device’s parameters will be set according to the val-
ues stored in the patch, and the name of the patch is shown in the Patch Name display.
! Any parameter adjustments you make on the device panel after loading a patch will not affect the actual patch
file on the computer. For this to happen, you need to save the patch - see “Saving patches”.
! For all the patch loading methods described above, you can automatically revert back to the original patch, by
clicking the Cancel button in the Browser.
For more details about browsing patches and searching for sounds, please refer to “Browsing patches for a device”
and “Filtering”.

83 SOUNDS, PATCHES AND THE BROWSER

 If referenced samples are missing
Patches for the Mimic, Grain, Europa (User Wave sample), Redrum, NN19, NN-XT and Dr. Octo Rex devices contain
references to samples or REX files. Just like patches, samples can be independent files on the hard disk or elements
within a ReFill. However, if sample files have been moved or renamed after a patch was saved, the sample file refer-
ences in the patch will not be accurate.
If this is the case when you select a patch, the program will tell you so. You can then choose to either manually locate
the missing files, to have the program search for them, or to proceed without the missing sounds. For details, see
“Handling Missing Sounds”.
! Proceeding without locating or replacing the missing samples will result in silent patches, drum channels, key
zones or loop slots (for the Redrum, NN19/NN-XT, Kong, Grain and Dr. Octo Rex).

84 SOUNDS, PATCHES AND THE BROWSER

 Saving patches
! All settings for all devices in your Reason Rack Plugin instances are automatically included when you save the
song or project in your DAW host - you don't have to save the patches separately!
Saving patches is useful if you have an instrument or effect setup that you want to use later, in other projects.
Saving is always done by clicking the Save Patch button on the device panel and specifying a name and location for
the file in the dialog that appears:

The Save Patch button on a device.

• If you have modified an existing patch and want to save it with the same name (overwriting the old version),
press [Alt](Win) or [Option](Mac) and click the Save Patch button.
This only works if the patch was loaded from a file on disk. If the patch was in a ReFill or a Rack Extension, you
cannot overwrite it and must save it elsewhere.
• The default folder for saving your patches is the User Library, see “User Library”.
You can of course save to other locations as well, if you like.

Copying and pasting patches between devices

A quick way to copy device parameter settings between devices of the same type is to use the “Copy Patch” and
“Paste Patch” functions. The result is exactly the same as if you had saved a patch from one device and opened it on
another device of the same type - this is just a quicker method.
! Copying and Pasting settings is possible with all device types that can use patches.
Proceed as follows:
1. Select a patch, and/or make the desired settings on the source device.
Make sure the device remains selected.
2. Select “Copy Patch” from the device context menu.
3. Select the destination device of the same type.
4. Select “Paste Patch” from the device context menu.
The settings of the source device (including Mimic, Europa, Grain, Redrum, NN19, NN-XT, and Kong sample ref-
erences and Dr. Octo Rex REX loop references) are automatically applied to the destination device.
! Note that this operation simply copies the settings from one device to another. Adjusting the settings on one
of the devices will not affect the other; neither will the edits affect any patch file on disk.

Initializing patches and resetting device parameters

Sometimes it is useful to start with a “clean slate” when creating a synth sound, a drum kit or a sampler patch. This is
achieved by selecting “Reset Device” from the device context menu. This resets all parameters to “default” values. Ini-
tializing Mimic, NN19, NN-XT, Dr. Octo Rex, Kong, Grain, Europa (if it uses a User Wave sample) or Redrum devices
will also remove all sample/REX file references, allowing you to start from scratch.

85 SOUNDS, PATCHES AND THE BROWSER

 Using the Browser
Browser overview
Fold/unfold buttons

Browse Focus field

(Type buttons
Filter text field

Category section

Tags section

Locations list

Item List

Sample/REX
Audition controls
Example of the Browser when using the Browse Patch button/function on a Subtractor device.
The Browser is used when you load patches, samples and REX files from regular file folders or from ReFills.
D Open and close the Browser by clicking the Browse icon on the global panel.
If no device is selected in the rack, the Browser will open without Browse Focus. Otherwise it will open with patch
Browse Focus for that device.

Browser open without Browse Focus, as indicated by the gray Browser header.

86 SOUNDS, PATCHES AND THE BROWSER

 D Right click a device panel in the rack and select “Browse Patches...” from the context menu.
This opens the Browser with Browse Focus, allowing you to browse for and load patches for the selected device.
D With a sample based device, right click the device panel (the sample section) in the rack and select “Browse
Samples” from the context menu (or click the “Browse Samples” button on the device panel).
This opens the Browser, where you can browse for, preview and load samples in the supported audio formats.
D With a Dr. Octo Rex Loop Player selected, right click the Programmer panel background and select “Browse
Loops” from the context menu (or click the “Browse Loops” button on the device panel).
This opens the Browser, allowing you to browse for, preview and load REX loops into the Dr. Octo Rex device.
D With a Kong Drum Designer device selected, select “Browse Drum Patches” from the context menu (or click
the “Browse Drum Patch” button on the Drum Control Panel on the device).
This opens the Browser, allowing you to browse for Kong Drum Patches and supported audio files to load into a
slot in the Kong Drum Designer device.
Besides standard file and folder browsing, the Browser offers several useful functions:
• Search for files by name, type, category, and tags - see “Filtering”.
• Use “cross-browsing” to search for patches belonging to any type of device.
For example, set browse focus from a Subtractor device by clicking its Browse Patch button. Instead of limiting the
Browser to show only Subtractor patches, you can choose to browse for any type of instrument patch. If you select
a patch with a different format than the device you are browsing from, the original device will be replaced by the
new device. See “Browsing patches for a device”.
• Audition audio samples and REX loops on the fly. See “Previewing samples and REX loops”.
• Save shortcuts to various locations on your local drive(s).
See “Shortcuts”.
• Create Favorite Lists containing shortcuts to your favorite files for instant access.
See “Favorite Lists”.

87 SOUNDS, PATCHES AND THE BROWSER

 Creating a device by browsing patches
If you want a certain sound or effect, the best way is often to add it directly by browsing (instead of first choosing a
device and then finding a patch for it).
D To add an instrument device, right-click anywhere in the rack and select "Add Instrument".
This opens the browser in Add Instrument mode, showing only instrument patches:

You can use the category and tag buttons to filter which patches are shown (or type directly into the filter text field
to filter by name, category, kind or tag). See “Filtering”.
1. Click a patch in the list to create a new device in the background.
You can audition it by playing your MIDI keyboard.
2. Select another patch with the mouse or the computer up/down arrow keys to load it instead.
3. Audition sounds this way until you find one you like.
4. Press [Return] or click OK to close the Browser, keeping the last loaded patch.
Double clicking a patch in the Item List is the same as selecting it and clicking OK, in one move.
D Alternatively, you can press [Esc] or click Cancel to close the browser and remove the created instrument.

Creating an Effect device by browsing patches

D To add an effect device, right-click anywhere in the rack select "Add Effect".
The insertion point of the new effect device is determined by where you right-click; often you want to right-click an
existing instrument or effect to add a new effect directly after that device.
This opens the browser in Add Effect mode, showing only effect patches.

88 SOUNDS, PATCHES AND THE BROWSER

 1. Select a patch to create its device in the background, then load other patches by clicking or using the computer
up/down arrow keys until you find the one you want.
You can audition the effect by playing the instrument it's connected to.
2. When you're done, close the browser by pressing [Return] or clicking OK.
Alternatively, press [Esc] or click Cancel to close the browser and remove the created effect.

Creating a Player device by browsing patches

D To add a Player to an instrument, right-click the instrument in the rack and select "Add Player".
This opens the browser in Add Player mode, which works the same way as the other Add modes (see above).

Browsing patches for a device

D To open the Browser for a specific device, click the “Browse patch” button on the device panel.

Browse Patch button

The Patch section of the Subtractor device.
D Alternatively, select the device in the rack and click the Browse icon on the Global Panel, or right-click the de-
vice panel and select "Browse Patches.,.".
This opens the browser in Patch Browse mode. The orange heading shows which device you're browsing for.

What is shown in the Browser depends on the device you started from:
• If the device had a patch loaded from the Browser (for example with the Add Instrument function), the browser
will show the same settings as when the patch was loaded.
This includes Location, Categories, Tags and other filtering settings.
• If the device had no patch loaded (or the default patch), the Browser will show all patches of that kind (e.g.
SubTractor Patches).
You can use the Patch Type buttons at the top to switch between showing patches for this device type only or for
all Instrument Patches - what we call cross-browsing.
• In Patch Browse mode you can click patches (or use the computer up/down arrow keys) to load them in the
background.
When you're happy with what you hear, click OK or press [Return} to close the Browser. Double clicking a patch is
the same as selecting it and clicking OK, in one move.
D To go back to the original patch instead, click Cancel or press [Esc].
q Tip: When you are browsing patches for a device, its "Home Folder" will be shown in the Locations list to the
left. This is a quick way to get to the patches that come with the device, shown in their original folder structure.

89 SOUNDS, PATCHES AND THE BROWSER

 Browsing samples for a device
D To load a sample, for example in a sample slot on a drum machine device, click its Browse Sample button.

Browse Sample button on Redrum

This opens the browser in Sample Browse mode. In this mode, the browser will only show samples.
It's often most useful to select a sample category (e.g. Drum Samples) here, since you might not want to go
through all instrument multisamples and similar:
D Click the Drum Samples button in the Category section to refine the list in the Item List to drum samples:

D Click samples in the Item List (or use the computer up/down arrow keys) to load the samples in the back-
ground.
When you're happy with what you hear, click OK or press [Return] to close the Browser. Double clicking a sample
is the same as selecting it and clicking OK, in one move.
D To go back to the original sample instead, click Cancel or press [Esc].

Previewing samples and REX loops

• The Audition controls below the Item List lets you preview the selected sample.
If Auto is on, selecting a sample will automatically play it back:

90 SOUNDS, PATCHES AND THE BROWSER

 Selecting multiple files
It is possible to select multiple files in the Browser, by using standard [Shift] or [Ctrl](Win) or [Cmd](Mac) selection
techniques. This, however, doesn’t necessarily mean that the selected files can be loaded. There are a number of in-
stances where selecting several files in the Browser file list is relevant:
• It is possible to load several drum patches (on consecutive pads) simultaneously into the Kong device.
See the “Kong Drum Designer” chapter for details.
• It is possible to load several samples simultaneously into the NN-XT, NN19 and Kong NN-Nano sampler de-
vices.
See “NN-19 Sampler”, “NN-XT Sampler” and “NN-Nano Sampler” for details.
• It is possible to load several REX files simultaneously into the Dr. Octo Rex device.
See the “Dr. Octo Rex Loop Player” chapter for details.
• You can select several files to add them to a Favorites List in one go - see “Favorite Lists”. Note that this is
only possible if a device does not have Browse Focus, see “Using the browser without Browse Focus”.
! In situations where several selected files (e.g. patches) cannot be loaded, the Create/Load button in the
Browser is grayed out.

About patch formats and sampler devices

As both the NN-XT and NN19 sampler devices can load patches in the NN19 (.smp) and REX (.rx2/.rcy/.rex) for-
mats, there must be certain rules regarding cross-browsing.
• The basic rule is that the Browser will load such patches into the original device type (the device you opened
the Browser from), whenever possible.
Thus, when the patch format is NN19 (.smp) or REX (.rx2/.rcy/.rex) and the NN19 device has browse focus, the
patch will be loaded into this device.
• If you are browsing from any other type of device, these patch types will be loaded into an NN-XT device.
• If you are using the “Add Instrument” function, an NN19 (.smp) patch will create a NN19 device and a REX
patch will create a Dr. Octo Rex device.

About browse lists

When you click the Browse Patch button to open a file from the Browser, the file and folder list shown at that time is
memorized for that device. This is called a “browse list”.
For patches (and to a certain extent samples) this list provides a specific functionality:
• The browse list is what applies when changing patches using the Next/Previous Patch buttons on the front
panel of a device (or from patch selectors on a control surface).
• The browse list is shown as the patch list when you click in the patch name field for a device.
• For samples, the browse list applies when changing samples using the Next/Previous Sample buttons on the
front panel of a sampler device.

91 SOUNDS, PATCHES AND THE BROWSER

 Other Browse Focus modes

Browse Focus means the Browser has a specific target (indicated by the orange heading background). The most
common examples are already described above, but there are some others:
• Browsing loops for a Dr Octo Rex or for a Nurse Rex module in Kong will open the Browser in Browse REX
Loops mode, showing only REX loops (.rx2 files).
• Browsing for a pad on Kong will open the Browser in a special Browse Drums mode, showing both samples
and Kong drum patches (.drum files).

Using the browser without Browse Focus

D With no device selected in the rack, click the Browse icon on the global panel at the top to open the Browser
without a specific Browse Focus (as shown by the gray heading):

Here you can see any type of item (as set with the Type buttons at the top). In this mode, patches and samples are
not automatically loaded in the background when you select them. Instead you have to explicitly load them.
Here are some examples:
D Select a patch and click the Create button (or double click the patch, or press [Return]) to create a new device
and close the Browser. Alternatively, drag and drop a patch to load/create the corresponding device.
D For samples, you can preview them as usual in the Browser. However to load samples you will have to drag
and drop the sample to an appropriate sampler device.

92 SOUNDS, PATCHES AND THE BROWSER

 • Alternatively, you can use drag and drop to drag an item from the Item List to the rack.
When you drag an item out of the Browser, it will automatically fold up to let you see background. This lets you
drop to devices or sample slots that were obscured by the Browser. When you drop the item, the browser reap-
pears.
If you change your mind before dropping, you can drag the item back to the browser heading to unfold it again.
q You can use drag and drop from the Browse Focus modes too. For example, if you are browsing piano patches
for a device and stumble upon a nice bass sound, you can just drag the bass patch into the rack to create a new
device. The Browser will remain in Patch Browse mode for your original piano instrument.

The Locations list

The Locations list to the left in the Browser provides different ways to get to files and folders.

All Locations
• All Locations shows all files that Reason knows about, in a flat list.
This means all patches, samples and loops in all folders listed in the left column (except the folders under the
"Computer" heading). All Locations is often the best starting point for browsing, since you can find all content
there, regardless of where the files actually are.

User Library
• User Library is a subfolder in the Music/Reason Studios folder on your computer.
When you save a patch in Reason, this is the default save folder. You can also move existing patches and sounds
there, to make sure they are indexed and included in "All Locations".

93 SOUNDS, PATCHES AND THE BROWSER

 Reason Library
• Reason Library shows the built-in sound banks, a subfolder listing all installed Rack Extensions and a ReFills
folder:

ReFills is a subfolder in the Music/Reason Studios folder on your computer. Reason Companion will install the op-
tional Drum Supply and Loop Supply ReFills in this folder, and you are also encouraged to move any ReFills you
have downloaded or purchased there. This will ensure that they are indexed and included in "All Locations".

Sound Packs
• If you are a Reason+ subscriber, you will continuously get new content in the form of Sound Packs.
You will find these in the Sound Packs location.

Shortcuts
• Shortcuts are aliases to folders and items that you need quick access to.
D To add an item as a Shortcut, drag it from the Item List and drop it under the Shortcuts heading.
D To delete a Shortcut, right click the folder icon and select “Delete Location” from the context menu.

Favorite Lists
• Favorite Lists are custom collections of patches, samples or loops.
There are two ways to add items to Favorite Lists:
D Drag files from the Item List and drop them on a Favorite List to the left.
D Right-click a file in the Item List and select a Favorite List from the context menu:

Adding a patch to the user created “Bells” Favorite List.

! The check mark next to a Favorite List means the item is already part of that Favorite List. Selecting it will re-
move it from the list.
D To rename or delete a Favorite List, right click the folder icon and select “Rename Favorite List” or “Delete Fa-
vorite List” from the context menu.
! We strongly recommend not to put ReFills in Favorite Lists. Instead, use the User Library or Shortcuts Location
for ReFills, see “User Library” and “Shortcuts”.

94 SOUNDS, PATCHES AND THE BROWSER

 Computer
• The Computer heading holds links to the User and Desktop folders on your computer.
These are not included in "All Locations" and therefore filtering cannot be used here (see below). Use these for
navigating to folders that you want to add as Shortcuts (see “Shortcuts”).

Filtering
The key to finding what you want in the Browser is using the filtering section above the item list. This is especially
powerful when “All Locations” is selected to the left:

The filtering section.

• You can set up the Browser to show only the Filter text field or the Categories and/or Tags buttons too, using
the fold buttons on each section:

The Filter text field, Categories and Tags sections unfolded.

The Type buttons

The Type buttons at the top of the Browser.

The Type buttons at the top specify which general type of files are listed. These can be set or limited automatically if
the browser is in a Browse Focus mode. For example, if you're browsing patches for an instrument device, you will
only be able to switch between Instrument Patches and patches for the current device type (e.g. Europa patches).

95 SOUNDS, PATCHES AND THE BROWSER

 The Text (search) field
The Filter text field can be the quickest way to filter the result:

• Typing something in the Filter text field shows a menu with filter property suggestions.
You can use the arrow keys to go down to a suggestion and press Return to select it. This adds that filter property,
shown to the right in the filter Filter text field:

The category “Keys: Organ” selected.

• If you just press Return in the text field (without going down the menu), the browser will show items that match
any of the filter properties on the menu.
This is indicated by the filter property "Any":

You can continue writing and adding other properties.

D To remove a filter property, click its X button (or use the left/right arrow keys to select it and press [Back-
space]).
The properties available on the menu are as follows:

• Name
The name of the item.
• Category
For example, whether an instrument is a piano or a synth pad, or whether a sample is a Snare Drum or a Sound FX
- different Types have different categories.

96 SOUNDS, PATCHES AND THE BROWSER

 • Tag
A more subjective description of a sound. All patches and sounds in the Reason Factory Sound library have been
manually tagged, to make them easier to search for and filter out.
• Kind
For patches, this is the device type (e.g. “Europa Patch”, “Polytone Patch” etc.).
For samples, Kind represents the file format: WAV, AIFF, mp3 etc.
! Adding multiple Names, Categories or Kinds will widen the result.
For example, if you filter on “Category: Keys” and “Category: Strings”, you will see both Keys and Strings patches.
! Adding multiple Tags will narrow down the result.
For example, if you filter on “Tag: Fat” and “Tag: Analog“ you will only see sounds that are tagged both as "Fat" and
"Analog".

The Categories buttons

• The Category buttons will be different depending on which Type button you have clicked.
If "All Types" is selected, no Category buttons are shown (but you can still filter on categories using the Filter text
field).
• In most cases, turning on a Category button will show its sub-categories:

Sub-categories for the “Bass” Category.

Sub-categories for the “Mallets” Category.

• You can choose whether to filter on the main category only ("Bass", "Mallets", etc) or to specify one or more
subcategories.
Multiple categories can be activated at the same time.
! Note: The "Uncategorized" Category shows all items that have no category (yet). That will initially include all
your own existing patches and samples as well as patches in 3rd party Rack Extensions and VSTs.

The Tags buttons

• The Tags buttons are the same for all Types.
They describe the character of a patch or sample and in some cases technical aspects ("Sampled", "Portamento",
etc). The "Staff Pick" tag lists sounds that we at Reason Studios think are extra nice!
• Multiple tags can be activated to narrow down the filtering result.

97 SOUNDS, PATCHES AND THE BROWSER

 Categories and Tags in the Filter text field
• Any activated Category or Tag buttons will be shown in the Filter text field.
You can clear them by clicking their individual X buttons (or clear all by clicking X for the whole text field):

98 SOUNDS, PATCHES AND THE BROWSER

 Editing Tags and Categories
You can change the categorization and tagging of items, or add categories and tags to your own sounds:
1. Set up the view to show the items you want to tag.
For example, if you want to categorize and tag all patches in a certain folder, navigate there. The items must be
part of the indexed locations; if you have custom patches in a folder on your hard drive you need to add that folder
as a Shortcut first, see “Shortcuts”.
2. Right click an item in the Item List and select "Edit Categories and Tags..."

The Browser switches to Edit mode:

Currently set Categories and Tags for the “DS8 Mk IV.thor” patch.
The Category and Tag buttons show the current categorization and tagging of the selected item.
D Turn buttons on or off to add or remove categories and tags.
D Select other items in the view to see and change their respective Categories and Tags.
• You can press [Shift] or [Cmd](Mac)/[Ctrl](Win) and click to select multiple items, and tag them all in one go.
If they have conflicting settings, those buttons will be shown with white text.
D To keep your changes and leave the edit mode, click Save at the bottom to the right in the Browser.
This takes you back to the regular browser.

99 SOUNDS, PATCHES AND THE BROWSER

 Other Browser functions

• Above the Item List, you will find back/forward buttons and a path menu for navigating up in the folder tree.
• To the right are buttons for selecting view mode: Flat list or Tree (showing folders).
Different locations have different default view modes when you go there. In some cases you cannot change the
view mode (“All Locations” is always a Flat list for example, while the folders under "Computer" will always be in
Tree mode).
D Right click a file in the Item List (Flat view) and select “Show in Folder” to show the file in its original folder (in
Tree view).
• You can sort by columns by clicking the column headings.
• You can resize the columns by dragging the heading dividers sideways.
• The entire Browser window can be resized and moved within the boundaries of the Reason Rack Plugin win-
dow.
! Note: Even though the Browser is shown, you can still edit settings in the rack.

Back/Forward buttons

These arrow buttons allow you to move back and forth between the Browser locations you have opened during the
browsing session, similar to when browsing pages in a web browser.

Path popup menu

This field displays the name of the currently selected root folder.
D Click in the name field to bring up a popup menu where you can move up in the folder hierarchy.
When you're in a top-level location such as All Locations or a Favorite list, the path popup menu cannot be used.

100 SOUNDS, PATCHES AND THE BROWSER

 D Click on the Up button to the right of the root folder list to move up in the hierarchy one level at a time:

! When you're in a top-level location such as All Locations or a Favorite list, the path popup menu cannot be
used.

Handling Missing Sounds

If you load a patch for a sample-based instrument, and not all samples can be found, the Global Panel will show a
Missing Sounds warning:

This may be because the samples have been moved or their folder renamed, etc.
1. Click the Missing Sounds warning to open the Missing Sounds dialog:

101 SOUNDS, PATCHES AND THE BROWSER

 This lists all samples and loops that couldn't be found.

| Column | Description
Device Shows the name of the device in which the missing sound is used, along with a device
type icon.
Sound Shows the name of the missing file, along with a file type icon.
Part of ReFill/R.E. If the missing file is part of a ReFill (or a SoundFont within a ReFill) or a Rack Extension
device, this column shows the name of the ReFill/SoundFont/Rack Extension.

Below the main display are three buttons:

D Select a file in the list and click the “Download ReFill/R.E.” button to open the Reason Companion app and
see if the ReFill can be found.
If it can’t be found, click the “Search for the ReFill in the Shop” button to direct your web Browser to the Reason
Studios Shop, where you can download (and purchase, if necessary) the ReFill/R.E.
Note that the “Download ReFill/R.E.” button is only enabled if the selected item comes from a ReFill or Rack Ex-
tension.
D If you have the missing samples somewhere on your computer, put them in a Browser Location (so the
Browser can find them), then click the “Search All Locations” button.
Reason will automatically search for and find the missing files in any of the Locations present in the Browser.
! If you have a lot of ReFills and folders in your Browser Locations, the search could take a while.
D Alternatively, close the Missing Sounds window.
The sampler device will still work, but with sounds missing. This means that sampler patches, drum machine
patches and/or loop players will not play back correctly.
On the panels of the concerned devices, missing samples are indicated with an asterisk (*) before the file names:

This sample is missing

About ReFills
A ReFill is a kind of component package for Reason and Reason Rack Plugin, which can contain sounds and effects
patches, samples, REX files and SoundFonts. On your computer, ReFills appear as (often large) files with the exten-
sion “.rfl”.
All sounds included in Reason Rack Plugin are embedded in two ReFills named “Reason Factory Sound Bank” and
“Orkester Sound Bank”, which were copied to the hard disk during installation.
Additional Reason Studios ReFills are available for purchase. You can also purchase ReFills from other sample man-
ufacturers, etc.
• Samples (Wave and AIFF files) are compressed to about half their original file size when stored in ReFills,
without loss of quality.
In Reason Rack Plugin you can use the browser to list and access the embedded sounds and other components
within the ReFills, just as if the ReFills were folders on your hard disk.
Furthermore, if a song makes use of components from ReFills, Reason Rack Plugin will tell you which ReFills are re-
quired.

102 SOUNDS, PATCHES AND THE BROWSER

Chapter 9
The I/O device
 Introduction

At the top of the rack is the I/O device (for "input/output"). This handles the audio communication between the de-
vices in the rack and the DAW host. It also includes a basic summing mixer for up to 8 stereo output channels and a
control for setting the Shuffle amount used by some pattern devices.

The back panel

This is where the rack devices are connected.

The input jacks deliver audio from the DAW host to devices in the rack. This is most often the case when Reason
Rack Plugin is used as an effect, and typically only the main (1-2) input jacks are used.
The output jacks deliver audio from devices in the rack to the DAW host. These are used both when Reason Rack
Plugin is an instrument and an effect.
• Devices you add will automatically be routed to free jacks on the I/O device.

The front panel

The first eight stereo outputs (output 1-16) have special settings on the front:

Level meters

These show the level of the signal received at the corresponding Output jack on the back panel.

104 THE I/O DEVICE

 Sum To Main (3-16 only)

When Sum to Main is on for an output pair, its signal is directed to the Main Out (1-2) instead, and summed with any
other signals there.
• If you want to layer several instrument devices on a single stereo channel in your DAW, leave "sum To Main"
on.
• If you want different devices in your rack to be routed to different channels in your DAW's mixer, turn off "sum
To Main" for these outputs.

Name labels

These show the names of the connected devices. By default, devices have the name of the loaded patch or the de-
vice type, but you can change this by double-clicking the tape labels on the device panels and typing in a new name.

Audio In/Out indicators

These light up whenever audio signals are received from the DAW host (input) or sent back to the DAW host (out-
put). Note that if you have connected an instrument to Output 3-4 and turned on sum To Main, the audio out indica-
tors for 3-4 will not light up when you play the instrument! Instead, the audio out indicators for 1-2 will light up
(because sum To Main sends the signal to the Main output 1-2).
The intensity of the lights indicate the audio signal levels.

Shuffle

Some devices features playback of patterns such as sequences and arpeggios. These can have a Shuffle mode,
where 1/16th notes are shuffled for a swing feel. Examples of such devices include the Dual Arpeggio Player (see
“Dual Arpeggio”) and the Redrum drum computer (see “Redrum Drum Computer”).
The amount of Shuffle for all such devices in a Reason Rack Plugin instance is set on the I/O device panel. Setting
Shuffle to 50% results in a "straight" beat, with no swing applied. Setting the Shuffle to a value of 66% results in a
perfect sixteenth-note triplet shuffle. Values between 50% and 66% have a less pronounced swing feel, and values
greater than 66% are more exaggerated.
• In the stand-alone version of Reason, this parameter is called "Global Shuffle" in the ReGroove Mixer.
If the reference manual for a device refers to Global Shuffle, it's the same as the Shuffle setting on the I/O device.

105 THE I/O DEVICE

106 THE I/O DEVICE
Chapter 10
Dr. Octo Rex
Loop Player
 Introduction

The Dr. Octo Rex Loop Player is the successor to the trusty Dr. Rex Loop Player, introduced in Reason Version 1. The
Dr. Octo Rex can hold up to eight different REX loops at once, in eight pattern memories, and allows you to switch
between loops and slices in very flexible ways.
The Dr. Octo Rex Loop Player is capable of playing back and manipulating files created in ReCycle, another product
created by Reason Studios, or bounced from open Single Take audio clips in the stand-alone version of Reason. Re-
Cycle is a program designed especially for working with sampled loops. By “slicing” an audio loop and making sepa-
rate samples of each beat, ReCycle makes it possible to change the tempo of loops without affecting the pitch and
to edit the loop as if it was built up of individual sounds.
! Please, note that this device is not available in Reason Lite Rack Plugin.

ReCycled Loops
To fully understand Dr. Octo Rex you need to understand what it means to ReCycle a drum loop. Imagine that you
have a sample of a drum loop that you want to use in a track you are working on. The loop is 144 BPM and your track
is 118 BPM. What do you do? You can of course lower the pitch of the loop, but that will make the loop sound very
different, and if the loop contains pitched elements they will no longer match your song. You can also time stretch it.
This won’t alter the pitch, but will make the loop sound different. Usually it means that you loose some “punch” in the
loop.
Instead of stretching the sample, ReCycle slices the loop into little pieces so that each drum hit (or whatever sound
you are working with) gets its own slice. These slices can be exported to an external hardware sampler or saved as a
REX file to be used in Reason. When the loop has been sliced you are free to change the tempo any way you want.

108 DR. OCTO REX LOOP PLAYER

 About REX file formats
Dr. Octo Rex can read REX files in the following formats:
• REX (.rex)
This is the file format generated by previous versions of ReCycle (Mac platform).
• RCY (.rcy)
This is the file format generated by previous versions of ReCycle (PC platform).
• REX 2 (.rx2)
This is the ReCycle file format for both Mac and PC platforms generated internally in the stand-alone version of
Reason or by ReCycle version 2.0 and later. One of the differences between the original REX format and REX2, is
that the REX2 format supports stereo files.
! Unlike the Dr. Rex device, Dr. Octo Rex can also load and save the device panel settings in a special Patch for-
mat (.drex). The REX file(s) and the Dr. Octo Rex panel settings are also saved in the Song file just like every
other patch in the song.

Loading and saving Dr. Octo Rex patches

Loading and saving Dr. Octo Rex patches is done in the same way as in other patch based devices - see “Loading
patches” and “Saving patches”.
Dr. Octo Rex patches can consist of up to eight separate REX loops. When you load an Dr. Octo Rex patch, the REX
loop(s) will be automatically loaded in the Loop Slots with their names shown in the display(s) below each button.
• You can also load Dr. Octo Rex patches by dragging them from the Browser and dropping on the device panel.

About the Dr. Octo Rex patch format

Dr. Octo Rex patches (.drex) contains all panel and synth parameter settings as well as references to all (up to eight)
REX loops. The actual REX files are not contained in the patch but must be available separately on the computer.

Playing Loops
1. Make sure the Enable Loop Playback button is on (lit).

2. Click the desired Loop Slot button.

3. Play back the loop by clicking the Run button.
The loop in the selected Loop Slot will play back repeatedly in the tempo set in the main sequencer. If you change
the tempo, the loop tempo will follow.

109 DR. OCTO REX LOOP PLAYER

 D You can also play the loop once via MIDI, by using the D0 key.
D To check out the loop(s) together with other device sequencer data and patterns already recorded, click the se-
quencer Play button.
The loops will automatically play back in perfect sync with the sequencer.

Switching playback between Loop Slots

Switching playback between loops in different Loop Slots is just like switching a Pattern in a Redrum device, for ex-
ample.
1. Activate the Enable Loop Playback button on the Dr. Octo Rex device.
2. Click the Run button - or start sequencer playback.
3. Click another Loop Slot button to switch loop.
Selecting Loop Slots that have no loops loaded will result in silence.

The Trig Next Loop function

The Trig Next Loop function determines how long Dr. Octo Rex waits after a Loop Slot button (or a key) is pressed
before it actually “gates in” or triggers the loop. This allows for different “precision” when switching between running
loops:

D Activate the Bar button to make the loops switch at the next bar of the current loop.
D Activate the Beat button to make the loops switch at the next beat of the current loop.
D Activate the 1/16 button to make the loops switch at the next 1/16th note of the current loop.

Switching Loops using Pattern Automation in the sequencer

Switching between Loop Slots can be automated, by using parameter automation in the main sequencer.
! When using parameter automation, the Trig Next Loop function described above is disregarded - the switching
between Loop Slots is instantaneous.

Triggering playback and selecting Loop Slots from a MIDI keyboard

It’s also possible to control playback, stop and Loop Slot selection in real-time by pressing different keys on a MIDI
keyboard. By pressing the keys E0 to B0 you select Loop Slot 1-8 and start playback of the corresponding loop. The
loop(s) will play back continuously, one loop at a time, until you press the D#0 key to stop playback, or click the Run
button or Stop in the main sequencer. The time between key press and Loop Slot switching is determined by the Trig
Next Loop function, see “The Trig Next Loop function”.
! Note that the Enable Loop Playback button must be on.

110 DR. OCTO REX LOOP PLAYER

 The picture shows what keys should be pressed to select and play Loop Slot and to stop loop playback:

Slot 3 play

Slot 5 play

Slot 7 play
Loop stop
Slot 1 play
Slot 2 play
Slot 4 play

Slot 6 play

Slot 8 play

C0 D0

• To maintain backwards compatibility with Dr. Rex, the D0 key can be used to play back the REX loop in the
Loop Slot that currently has Note To Slot focus (see “Note To Slot”).
The loop is played back once (single-shot) and cannot be stopped during this time.

Adding Loops
To add one or several (max 8) loops into the Dr. Octo Rex Loop Player, proceed as follows:
1. Unfold the Loop Editor panel.
2. Select the Loop Slot you wish to add the (first) REX loop into.

3. Open the REX Loop browser by clicking the folder button to the left of the Loop Slot buttons.
Alternatively, select “Browse Loops...” from the Edit menu or the device context menu.
4. In the Browser, locate and select the desired loop(s).
You can listen to the loops before loading by using the Preview function in the Browser.
D To select several loops, hold down [Ctrl](Win)/[Cmd](Mac) and click.
To select a range of loops, hold down [Shift] while clicking the last file.
5. Click the Load button in the Browser to load the selected file(s) in the Loop Slot(s).
! If you have selected and opened several loops, the first loop will load in the selected Loop Slot and the rest will
load in consecutive Loop Slots.
! Loading new REX files will replace any files currently in the slots.
D Alternatively, select the REX loop(s) in the Browser and drag and drop it/them on the Loop Editor panel sec-
tion, or on a Loop Slot button on the Controller Panel.
If you have selected several loops, the first loop will load in the selected Loop Slot and the rest will load in consec-
utive Loop Slots.
! If you drag a single REX loop from the Browser and drop on the Controller Panel (not on a Loop Slot button),
the REX loop will load into Slot 1 and all other Slots will be cleared.

Loading Loops “On the Fly”

Another practical method for checking out loops, is to load them “on the fly”, i.e. during playback. This is especially
useful if you want to check out a number of loops against other sequencer data and patterns previously recorded.
Proceed as follows:

111 DR. OCTO REX LOOP PLAYER

 1. Activate the Enable Loop Playback button on the Dr. Octo Rex device and start sequencer playback.
The REX loops and the sequencer are synced.
2. Now load a new REX file by using the Browser in one of the usual ways.
After a brief silence, the new file is loaded, and sync is maintained.
3. Repeat step 2 as necessary until you have found a suitable loop.
D If you are trying out loops within the same folder, the quickest ways to select and load a new loop is to use the
arrow buttons next to the loop name display.
Or, you can click in the loop name display and select a new loop from the pop-up menu that appears.

Removing Loops
D To remove a loop from a Loop Slot, select “Remove Loop” from the Edit menu or device panel context menu.

Cut/Copy and Paste Loops between Loop Slots

To cut or copy a loop from one Loop Slot and paste into another, proceed as follows:
1. Click the button of the Loop Slot that contains the loop you want to cut or copy.
2. Select “Cut Loop” or “Copy Loop” from the panel context menu.
! You have to context-click on the panel (not any button or knob) to access the correct context menu.
3. Click the destination Loop Slot button and select “Paste Loop” from the panel context menu.
Now, you can edit the slices of the pasted loop as desired, see “Slice handling”.

Playing individual Loop Slices

Besides playing back entire REX Loops using the Run function (or Play in the sequencer), it’s also possible to play in-
dividual slices of a loop from a MIDI master keyboard. This way you can use the Dr. Octo Rex almost like a traditional
sampler, playing separate slices from separate keys.
The slices are automatically distributed in semitone steps, with the first slice on MIDI note C1, the second slice on
C#1 and so on, with one note for each slice. The note range differs depending on how many slices the REX Loop
contains.
D Define which REX Loop to control from the MIDI master keyboard by selecting the desired Loop Slot with the
Note To Slot knob:

The range is 1-8 corresponding to Loop Slots 1-8. Selected Slot is indicated with a lit LED.

112 DR. OCTO REX LOOP PLAYER

 Slice handling
Selecting Slices

A selected slice is indicated by being highlighted in the waveform display. To select a slice, use one of the following
methods:
D By clicking in the waveform display.
If you hold down [Alt](Win) or [Option](Mac) and click on a slice in the waveform display, it will be played back. The
pointer takes on the shape of a speaker symbol to indicate this.
D By using the “Slice” knob below the waveform display.
D Via MIDI.
If you activate “Select Slice Via MIDI”, you can select and “play” slices using your MIDI keyboard. Slices are always
mapped to consecutive semitone steps, with the first slice always being on the “C1” key.

D If you play back a loop with “Select Slice via MIDI” option activated, each consecutive slice is selected in the
waveform display as you play the keys.
You can edit parameters during playback.

113 DR. OCTO REX LOOP PLAYER

 Editing Slices in the Waveform Display

Here you are able to edit several parameters for each slice, by first selecting the slice and then using the knobs be-
low the waveform display. If you want to edit a single parameter for several slices at once, a more convenient way
would be to use the Slice Edit Mode, see “The Slice Edit Mode”. The following slice parameters can be set:

| Parameter | Description
Pitch Allows you to transpose each individual slice in semitone steps, over a range of more than eight octaves.
Pan The stereo position of each slice.
Level The volume of each slice. The default level is 100.
Decay Allows you to shorten individual slices.
Rev Allows you to play back individual slices reversed (backwards)
F.Freq Allows you to modify the Filter (cutoff) Frequency of individual slices. This value is added to, or subtracted
(if negative) from the FREQ value of the synth panel, see “Filter Frequency”.
Alt Allows you to assign slices to an Alternate group (1-4). Slices assigned to any of these four Alt groups will
be played pack in a random fashion within each group, see “About the Alt parameter”.
Output Allows you to assign individual slices to separate audio output pairs (1-2, 3-4, 5-6 or 7-8). If you want the
slices routed to individual outputs (1-8), select the appropriate output pair and then pan the slice hard
Left/Right using the Pan parameter described above.

! If you have made settings to any of the parameters listed above, these will be lost if you load a new REX file
into that Loop Slot.

About the Alt parameter

The Alt parameter in the waveform display can be used to create a more “live” feel to your Rex loops by alternating
slices within each individual Alt group. For example, if you assign all snare hit slices in the loop to the Alt 1 group, the
snare slices will be selected and played back randomly each time these slices appear in the loop. Then, you could as-
sign all hi-hat slices to the Alt 2 group and so on. The result will be a loop that plays back differently every cycle.
In the example below, slices 3 and 6 have been assigned to the same Alt group. Here, we show the loop played back
five times, just so you can see the slice alternation. As you can see, slices 3 and 6 have been distributed randomly for
each loop cycle:

This randomization within each Alt group also occurs when you play back the REX loop using the Run function - and
when you use parameter automation in the main sequencer.

114 DR. OCTO REX LOOP PLAYER

 The Slice Edit Mode
A very convenient way of editing several slices at once is to work in Slice Edit Mode. In Slice Edit Mode, you can edit
one parameter at a time for all slices in the loop.
1. Click the Slice Edit Mode button.

The waveform display switches to show the REX loop in Slice Edit Mode.
2. Select the parameter you want to edit by clicking on its name below the REX loop.
The parameters that can be selected are: Pitch, Pan, Level, Decay, Reverse, Filter Frequency, Alt Group and Out-
put.

Here, we have selected the Pitch parameter.

3. Edit the Pitch value for each individual slice by clicking, or drawing across several slices, in the display.

Now, the Pitch parameter can be edited for all slices in a single sweep.
D To reset the selected parameter to its default value for one or several slices, hold down [Ctrl](Win) or
[Cmd](Mac) and click on the desired slice(s), or draw across the slices in the waveform display.
4. When you are finished with one parameter, select another parameter and repeat the procedure by drawing val-
ues for the slices in the waveform display.
! If you have made settings to any of the parameters listed above, these will be lost if you load a new REX file
into that Loop Slot.

115 DR. OCTO REX LOOP PLAYER

 Dr. Octo Rex panel parameters
Pitch and Mod wheels

The Pitch wheel to the left is used for “bending” the pitch up or down. The Mod wheel can be used to apply various
modulation while you are playing the loop(s). Virtually all MIDI keyboards have Pitch Bend and Modulation controls.
Dr. Octo Rex also has two “wheels” on the panel that could be used to apply real time modulation and pitch bend
should you not have these controllers on your keyboard, or if you aren’t using a keyboard at all. The wheels mirror the
movements of the corresponding MIDI keyboard controllers.
The Pitch bend range and Mod destination parameters are set on the synth parameter panel, see “Pitch Bend Range”
and “Mod. Wheel”.

Trig Next Loop

The Trig Next Loop parameter determines the timing when switching between Loop Slots See “The Trig Next Loop
function”.

Note To Slot

The Notes To Slot knob set to Slot 1

The Note To Slot knob controls which Loop Slot is currently controlled from the MIDI master keyboard, or by any re-
corded sequencer notes. The Loop Slot which currently has “note input” is indicated with a lit LED.
The Note To Slot parameter can also be automated in the main sequencer. This means you could switch between
Loop Slots for every single sequencer note if you like. This opens up for very interesting “beat mangling” experiments.

116 DR. OCTO REX LOOP PLAYER

 Loop Slot buttons

The eight Loop Slot buttons are located in the center of the front panel. You can load one REX loop per Slot. Loading
REX loops are done from the Loop Editor panel, see “Select Loop & Load Slot”.
D Click a Loop Slot button to select its REX loop for playback.
Play back the REX loop in the selected Loop Slot by clicking the Run button (or Play in the main sequencer).
! Note that selecting a Loop Slot only selects the corresponding REX loop for playback using the Run function
(see “Enable Loop Playback and Run”) or Play from the main sequencer. Which Loop Slot the master keyboard
or sequencer notes control is defined with the Note To Slot button, see “Note To Slot”.

Enable Loop Playback and Run

D Click the Enable Loop Playback button to make it possible to play back the REX loops using the Run button or
Play function in the main sequencer.
If the Enable Loop Playback button is off, clicking Run or Play in the sequencer won’t play back the loops. This can
be useful if you only want to control the individual slices of the REX loops from a master keyboard or from re-
corded notes in the main sequencer.

Volume

The Master Volume parameter acts as a general volume control for the loops in all Loop Slots.

Global Transpose

Set the global transposition of the loops in all Loop Slots by using the Global Transpose spin control. You can raise
or lower the pitch in 12 semitone steps (+/– 1 octave).

117 DR. OCTO REX LOOP PLAYER

 • The Global Transpose value can also be controlled via MIDI, by pressing a key between C-2 and C0 (with C-1
resetting the transpose value to zero).
This way you can also record transposition changes in the sequencer.

Dr. Octo Rex synth parameters

The Dr. Octo Rex synth parameters are used for shaping and modulating the sound of the REX loops. These param-
eters are familiar synth parameters, similar to the ones in the synthesizers; The Subtractor and the Malström, and in
the samplers; the NN-19 and the NN-XT. It is important to remember that these parameters do not alter the REX
files in any way, only the way they will play back.
! Most of the synth parameters are global, in the sense that they will affect all slices in the REX files as well as
all REX loops in all eight Loop Slots.
! All Dr. Octo Rex synth panel settings are stored in the Song (and in the Dr. Octo Rex patch file if you choose to
save the settings as a patch).

Select Loop & Load Slot

D Click any of the eight Select Loop & Load Slot buttons to select a loaded REX loop for editing, or to load a new
REX file to.
If no loop is already present in the selected Loop Slot, the Waveform Display will be blank. Otherwise, the display
shows a graphical readout of the REX loop and info (name, original loop tempo, number of bars and signature).
D Click the Follow Loop Playback button to “synchronize” the Select Loop & Load Slot buttons to the Loop Slot
buttons on the front panel.
This way, the currently playing loop will always be displayed in the Waveform Display. If you’re using Pattern Auto-
mation in the sequencer, where the Slots are switched during playback, you might want to deactivate the Follow
Loop Playback function to make it easier to edit a specific loop.
Refer to “Adding Loops” for info on how to load REX files and to “Editing Slices in the Waveform Display” for info
about editing the REX loop.

Copy MIDI Notes to a sequencer track

It’s possible to copy the MIDI Notes of the REX loop of the currently selected Slot to a track in your DAW’s se-
quencer:

118 DR. OCTO REX LOOP PLAYER

 1. Click to select the desired Loop Slot.
2. Click the “Drag MIDI Notes to track in sequencer” icon and drag to the destination track in the sequencer:

! Be sure to disable the Enable Loop Playback function on the Dr Octo Rex panel to avoid “doubled notes”.

Loop Transpose

D Set the transposition of individual loops in the Dr. Octo Rex by using the Loop Transpose knob to the bottom
left on the panel, or by clicking on the keyboard display below the knob.
You can raise or lower the pitch in 12 semitone steps (+/–1 octave).
q It’s also possible to set a global transpose value that affects all REX loops equally, see “Global Transpose”.

Loop Level
D Set the individual levels for the loops in the Loop Slots with the Loop Level knob.
This lets you match the levels of the loops in the 8 Loop Slots.

Oscillator section

For a REX file, the audio contained in the slices are what oscillators are for a synthesizer, the main sound source. The
following settings can be made in the Osc Pitch section of the Dr. Octo Rex:

119 DR. OCTO REX LOOP PLAYER

 Env. A
This parameter determines to what degree the overall pitch of all the REX files will be affected by the Filter Envelope
(see “Filter Envelope”). You can set negative or positive values here, which determines whether the envelope curve
should raise or lower the pitch.

Oct and Fine - Setting the overall Pitch

You can change the overall pitch of all REX files in the 8 Loop Slots in three ways:
D In octave steps.
This is done using the Oct knob. The range is 0 - 8, with “4” as default.
D In semitone steps.
This is done by using Global Transpose controls, see “Global Transpose”.
D In cents (hundredths of a semitone).
The range is -50 to 50 (down or up half a semitone).
q To transpose individual REX loops, use the Loop Transpose parameter, see “Loop Transpose”.
! To tune an individual slice in a REX loop, select it and use the Pitch parameter below the waveform display,
see “Editing Slices in the Waveform Display”.

Mod. Wheel

The Modulation wheel can be set to simultaneously control a number of parameters. You can set positive or negative
values, just like in the Velocity Control section. The following parameters can be affected by the modulation wheel:

| Parameter | Description
F. Freq This sets modulation wheel control of the filter frequency parameter. A positive value will raise the fre-
quency if the wheel is pushed forward. Negative values invert this relationship.
F. Res This sets modulation wheel control of the filter resonance parameter. A positive value will increase the
resonance if the wheel is pushed forward. Negative values invert this relationship.
F. Decay This sets modulation wheel control for the Filter Envelope Decay parameter. A positive value will increase
the decay if the wheel is pushed forward. Negative values invert this relationship.

Velocity section

Velocity is usually used to control various parameters according to how hard or soft you play notes on your keyboard.
A REX file does not contain velocity values on its own. As velocity information is meant to reflect variation, having
them all set to the same value is not meaningful if you wish to velocity control Dr. Octo Rex parameters.
There are basically two ways you can apply “meaningful” velocity values to REX files:

120 DR. OCTO REX LOOP PLAYER

 • After having recorded MIDI notes in the main sequencer, you can edit the velocity values in the in the se-
quencer.
• You can play slices in real time on your keyboard. The resulting data will have velocity values reflecting how
the notes were struck when you played.
When velocity values have been adjusted, you can control how much the various parameters will be affected by ve-
locity. The velocity sensitivity amount can be set to either positive or negative values, with the center position repre-
senting no velocity control.
The following parameters can be velocity controlled:

| Parameter | Description
F. Env This sets velocity control for the Filter Envelope Amount parameter. A positive value will increase the en-
velope amount with higher velocity values. Negative values invert this relationship.
F. Decay This sets velocity control for the Filter Envelope Decay parameter. A positive value will increase the De-
cay time with higher velocity values. Negative values invert this relationship.
Amp This let’s you velocity control the overall volume of the file. If a positive value is set, the volume will in-
crease with higher velocity values.

The Filter Section

Filters are used for shaping the overall timbre of all REX files in all 8 Loop Slots. The filter in Dr. Octo Rex is a multi-
mode filter with five filter modes.
D Activate or deactivate the filter completely by clicking the Filter On button.
The filter is active when the button is lit.

Mode
With this selector you can set the filter to operate as one of five different types of filter. These are as follows:
• Notch
A notch filter (or band reject filter) could be described as the opposite of a bandpass filter. It cuts off frequencies
in a narrow midrange band, letting the frequencies below and above through.
• High-Pass (HP12)
A highpass filter is the opposite of a lowpass filter, cutting out lower frequencies and letting high frequencies pass.
The HP filter slope has a 12 dB/Octave roll-off.
• Bandpass (BP 12)
A bandpass filter cuts both high and low frequencies, while midrange frequencies are not affected. Each slope in
this filter type has a 12 dB/Octave roll-off.
• 12 dB Lowpass (LP 12)
This type of lowpass filter is also widely used in classic analog synthesizers (Oberheim, early Korg synths, etc.). It
has a gentler slope (12 dB/Octave), leaving more of the harmonics in the filtered sound compared to the LP 24 fil-
ter.

121 DR. OCTO REX LOOP PLAYER

 • 24 dB Lowpass (LP 24)
Lowpass filters lets low frequencies pass and cuts out the high frequencies. This filter type has a fairly steep roll-
off curve (24dB/Octave). Many classic synthesizers (Minimoog/Prophet 5 etc.) used this filter type.

Filter Frequency
The Filter Frequency parameter (often referred to as “cutoff”) determines which area of the frequency spectrum the
filter will operate in. For a lowpass filter, the frequency parameter could be described as governing the “opening” and
“closing” of the filter. If the Filter Freq is set to zero, none or only the very lowest frequencies are heard, if set to max-
imum, all frequencies in the waveform are heard. Gradually changing the Filter Frequency produces the classic syn-
thesizer filter “sweep” sound.
! Note that the Filter Frequency parameter is usually controlled by the Filter Envelope (see “Filter Envelope”) as
well. Changing the Filter Frequency with the Freq slider may therefore not produce the expected result.

Resonance
The filter resonance parameter affects the character of the filter sound. For lowpass filters, raising the resonance will
emphasize the frequencies around the set filter frequency. This produces a generally thinner sound, but with a
sharper, more pronounced filter frequency “sweep”. The higher the resonance value, the more resonant the sound be-
comes until it produces a whistling or ringing sound. If you set a high value for the resonance parameter and then vary
the filter frequency, this will produce a very distinct sweep, with the ringing sound being very evident at certain fre-
quencies.
• For the highpass filter, the resonance parameter operates just like for the lowpass filters.
• When you use the Bandpass or Notch filter, the resonance setting adjusts the width of the band.
When you raise the resonance, the band where frequencies are let through (Bandpass), or cut (Notch) will become
narrower. Generally, the Notch filter produces more musical results using low resonance settings.

Envelope section
Envelope generators are used to control several important sound parameters in analog synthesizers, such as pitch,
volume, filter frequency etc. In a conventional synthesizer, envelopes govern how these parameters should respond
over time - from the moment a note is struck to the moment it is released. In the Dr. Octo Rex device however, the en-
velopes are triggered each time a slice is played back.
There are two envelope generators in the Dr. Octo Rex, one for volume, and one for the filter frequency (and/or
pitch). Both have the standard four parameters; Attack, Decay, Sustain and Release
! Please refer to “Envelopes - General” in the Subtractor chapter for a description of the basic envelope param-
eters.

Amplitude Envelope

122 DR. OCTO REX LOOP PLAYER

 The Amp Envelope governs how the volume of each slice should change over time, from the time it is triggered (the
slice note starts) until the slice note ends. This can be used to make a loop more distinct (by having a snappy attack
and a short decay time) or more spaced-out (by raising the attack time).

Filter Envelope

The Filter Envelope can be used to control two parameters for all REX loops in the 8 Loop Slots; filter frequency and
overall loop pitch. By setting up a filter envelope you control how the filter frequency and/or the pitch should change
over time for each slice.
The Amount parameter determines to what degree the filter frequency will be affected by the Filter Envelope. The
higher the Amount setting, the more pronounced the effect of the envelope on the filter.
q Try lowering the Frequency slider and raising Resonance and Envelope Amount to get the most effect of the
filter envelope!

LFO section

LFO stands for Low Frequency Oscillator. LFOs are oscillators in the sense that they generate a waveform and a fre-
quency. However, there are two significant differences compared to normal sound generating oscillators:
• LFOs only generate waveforms with low frequencies.
• The output of the two LFOs are never actually heard. Instead they are used for modulating various parameters.
The most typical application of an LFO is to modulate the pitch of a (sound generating) oscillator or sample, to pro-
duce vibrato. In the Dr. Octo Rex device, you can also use the LFO to modulate the filter frequency or panning.

Waveform
LFO 1 allows you to select different waveforms for modulating parameters. These are, from top to bottom:

| Waveform | Description
Triangle This is a smooth waveform, suitable for normal vibrato.
Inverted This produces a “ramp up” cycle. If set to control pitch (frequency), the pitch would sweep up to a set point (governed by
Sawtooth the Amount setting), after which the cycle immediately starts over.

123 DR. OCTO REX LOOP PLAYER

 | Waveform | Description
Sawtooth This produces a “ramp down” cycle, the same as above but inverted.
Square This produces cycles that abruptly changes between two values, usable for trills etc.
Random Produces random stepped modulation to the destination. Some vintage analog synths called this feature “sample & hold”.
Soft Random The same as above, but with smooth modulation.

Destination
The available LFO Destinations are as follows:

| Destination | Description
Osc Selecting this makes LFO control the pitch (frequency) of the REX file.
Filter Selecting this makes the LFO control the filter frequency.
Pan Selecting this makes the LFO modulate the pan position of the REX file, i.e. it will move the sound from
left to right in the stereo field.

Sync
By clicking the SYNC button you activate/deactivate LFO sync. The frequency of the LFO will then be synchronized
to the main sequencer tempo, in one of 16 possible time divisions. When sync is activated, the Rate knob (see below)
is used for setting the desired time division.
Turn the knob and check the tooltip for an indication of the time division.

Rate
The Rate knob controls the LFO’s frequency. Turn clockwise for a faster modulation rate.

Amount
This parameter determines to what degree the selected parameter destination will be affected by the LFO 1, i.e. the
amount of vibrato, filter wah or auto-panning.

Pitch Bend Range

The Pitch Bend Range parameter sets the amount of pitch bend when the wheel is turned fully up or down. The max-
imum range is 24 semitones (=up/down 2 Octaves).

124 DR. OCTO REX LOOP PLAYER

 Setting number of voices - polyphony

This determines the polyphony, i.e. the number of voices, or slices, Dr. Octo Rex can play simultaneously. For normal
loop playback, it is worth noting that slices sometimes “overlap”. Therefore, it is recommended that you use a polyph-
ony setting of about 3-4 voices when playing REX files. If you are “playing” slices via MIDI, the polyphony setting
should be set according to how many overlapping slices you want to have.
! Note that the Polyphony setting does not “hog” voices. For example, if you are playing a file that has a polyph-
ony setting of ten voices, but the file only uses four voices, this doesn’t mean that you are “wasting” six voices.
In other words, the polyphony setting is not something you need to consider if you want to conserve CPU
power - it is only the number of voices actually used that counts.

Audio Quality settings

Dr. Octo Rex features two parameters that provide ways of balancing audio quality vs. conservation of computer
power. The parameters are called “High Quality Interpolation” and “Low Bandwidth” and are located to the right on
the rear panel:

High Quality Interpolation

When High Quality Interpolation is active, the loop file playback is calculated using a more advanced interpolation al-
gorithm. This results in better audio quality, especially for loops with a lot of high frequency content.
• High Quality Interpolation uses more computer power - if you don’t need it, it’s a good idea to turn it off.
Listen to the loop in a context and determine whether you think this setting makes any difference.

Low Bandwidth
This will remove some high frequency content from the sound, but often this is not noticeable (this is especially true
if you have “filtered down” your loop). Activating this mode will save you some extra computer power, if needed.

Connections

125 DR. OCTO REX LOOP PLAYER

 On the rear panel of Dr. Octo Rex you will find the connectors. The left part of the panel houses a number of CV/Gate
inputs and outputs. Using CV/Gate is described in “Routing Audio and CV”.

Modulation Inputs
These control voltage (CV) inputs (with trim pots), allow you to modulate various Dr. Octo Rex parameters from other
devices (or from the modulation outputs of the Dr. Octo Rex device itself). The following CV inputs are available:
• Master Volume
• Mod Wheel
• Pitch Wheel
• Filter Cutoff
• Filter Resonance
• Osc Pitch

Modulation Outputs
The Modulation outputs can be used to voltage control other devices, or other parameters in the Dr. Octo Rex device
itself. The Modulation Outputs are:
• Filter Envelope
The Filter Envelopes in Dr. Octo Rex are polyphonic (one per voice). Only the filter envelope of voice 1 is output
here.
• LFO

Gate Inputs
These inputs can receive a CV/gate signal to trigger the two envelopes. Note that connecting to these inputs will
override the “normal” triggering of the envelopes. For example, if you connected an LFO CV output on another device
to the Gate Amp input on the Dr. Octo Rex, the amplitude envelope would not be triggered by the incoming MIDI
notes to the Dr. Octo Rex device, but by the LFO CV signal. In addition you would only hear the LFO triggering the en-
velope for the slices that were playing at the moment of the trigger.
• Amp Envelope
• Filter Envelope

Gate Output
This outputs a gate signal for each triggered slice in the loop.

Slice Outputs
To the right of the modulation inputs and outputs are the eight individual slice audio outputs. You can assign individ-
ual slices to any of these outputs as described in “Editing Slices in the Waveform Display”.

Main Outputs
To the right are the main left and right audio outputs. When you create a new Dr. Octo Rex device, these are auto-
routed to the first available outputs in the I/O device.

126 DR. OCTO REX LOOP PLAYER

Chapter 11
Europa
Shapeshifting
Synthesizer
 Introduction

The Europa Shapeshifting Synthesizer is the most advanced and sonically “wide” synthesizer in Reason. Despite
being a very advanced synthesizer, it’s really easy to create great sounds from scratch. Just a few mouse clicks and
knob twists in a Sound Engine section will generate truly impressive and inspiring sounds!
The three powerful and flexible sound engines offer a unique combination of analog/wavetable/spectral/physical
modeling/FM synthesis techniques. If you like, you could also draw your own waveforms and filter curves to design
your own completely unique sounds. In addition to this you can also load your own sample into Europa and use as a
wavetable and/or filter! Each sound engine also has its own Unison module for generating really wide multi-voice
stereo chorus effects.
The extensive Envelopes section and Modulation Bus section allow for very detailed and flexible modulation and con-
trol. Europa also features a top-notch semi-modular multi-effect section so you could put that final touch on your
sounds.
Don’t forget to check out the Europa videos here!
! Please, note that this device is not available in Reason Lite Rack Plugin.

128 EUROPA SHAPESHIFTING SYNTHESIZER

 Panel overview
The Europa front panel contains the following sections:

1 2
3

4 5 6 7

8 9 10

11 12

The Europa front panel sections.

• 1. MIDI Note On LED.
• 2. Patch Selector (for browsing, loading and saving patches).
• 3. Sound Engines section.
• 4. User Wave and Mixer section.
• 5. Filter section.
• 6. Amplifier section.
• 7. Global output controls.
• 8. Global performance and “play” controls.
• 9. Envelopes section.
• 10. LFO section.
• 11. Modulation Bus section.
• 12. Effects section.

129 EUROPA SHAPESHIFTING SYNTHESIZER

 Signal flow
The picture below shows the basic signal flow in Europa:
Engine III
Engine II
Amt Mod Amt Mod Engine I

KBD Mod Vel Count Blend

Modifier 1 Modifier 2
Pan

Oscillator Spectral Filter Harmonics Unison

Level

Pitch KBD Shape Freq Reso Modify Amt Detune Width

KBD Mod Vel Pan

Amp
Filter Multi FX
Envelope Out
Volume

: audio signal
Per Voice Drive Freq Reso Gain Vel : control signal

Europa signal flowchart.

• The “hearts” of Europa are the three Sound Engines I, II and III.
• First in each Sound Engine is an Oscillator, which generates the basic audio signal.
The oscillators in Europa are extremely powerful and flexible. Besides all the basic “analog” waveforms, the oscil-
lators can also generate a vast variety of wavetable waveforms, physical modeling signals and other types of
unique signals - and also your own samples! The signals can also be continuously transformed into various shapes.
• The Oscillator signal can be modified by the two Modifiers.
The Modifiers feature a huge amount of algorithms that can modify the Oscillator signal in various ways.
• The signal from the Oscillator is routed to the Spectral Filter.
The Spectral Filter affects the partials of the signal. The algorithms could be various types of filters - or special
purpose signal processors.
• The signal from the Spectral Filter can then be routed to the Harmonics processor.
The Harmonics processor modulates the harmonics in the signal, for example introducing ensemble or stretch ef-
fects.
• The signal from the Harmonics processor can then be routed to the Unison module.
This module can generate various types of unison chorus effects, to make the sound really fat and wide.
• The signals from the Sound Engines are then routed to the Mixer, where you can set the mix between the three
Sound Engine output signals and also pan the signals individually.
• The mixed signal is then routed to the Filter, Amp Envelope and then, via the Multi FX section, to the stereo
outputs.
• The remaining sections in Europa (Envelopes 1-4 and LFOs 1-3) can be freely assigned to modulate destina-
tion parameters via the Modulation Bus section.

130 EUROPA SHAPESHIFTING SYNTHESIZER

 Playing and using Europa
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Global output controls

Master Volume
This is the main stereo output volume control.

Voices
Here you set the desired maximum polyphony of your patch, from 1 to 16 voices. This control is mainly intended for
deliberately restricting the polyphony of a sound. If you just want to play a patch polyphonically you can leave the
Voices setting at 16 at all times. The DSP Load won’t increase with higher voice number settings - only if you play a
lot of notes simultaneously.
q If you want monophonic playback you could use the “Retrig” and “Legato” modes instead of lowering the
Voices parameter to 1.

Global performance and “play” controls

131 EUROPA SHAPESHIFTING SYNTHESIZER

 Key Mode
Here you choose how Europa should respond to MIDI Note data:
• Poly
Select this if you want to play Europa polyphonically. The maximum number of voices is 16. The number of voices
is set in the Voices control at the center right of the Europa panel, see “Voices”.
• Retrig
Select this if you want to play Europa in monophonic mode and always retrigger the envelopes as soon as you play
a new note.
• Legato
The Mono Legato mode is also monophonic. However, if you play a new note without having released the previous
one, the envelopes won’t start over.

Porta
Portamento makes note pitches glide from previous notes to new ones, at the time set with the Time knob. Porta-
mento can be used in all Key modes (see above).
• When On in Poly Key Mode (see above), the pitches will glide from any of the available voices.
The results will be unpredictable since there is no way of controlling from which note(s) the glide(s) will com-
mence. The effect is very nice, though.
• When On in Retrig or Legato Key Mode (see above), the pitch will glide between consecutive notes.
• In Auto mode, the pitch will glide between consecutive monophonic notes only when you play legato. If you
have selected Poly Key Mode (see above), Auto will have no effect at all.
If you release the previous key before hitting the new key, there will be no portamento effect.

P.Range
D Set the desired Pitch Bend range for the “Pitch” wheel with the up/down buttons, or by click-holding on the
display and dragging up/down.
Range: +/-24 semitones (+/-2 octaves) in steps of +/-1 semitone.

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Europa also responds to Pitch Bend MIDI
data from a connected MIDI master keyboard. You set the desired Pitch bend Range with the “P.Range” control
above the Pitch bend wheel.

Mod
The Mod wheel can be used for controlling almost any parameter in Europa. Use the Mod wheel as a Source param-
eter in any of the Modulation Source boxes in the different sections. Or use it as Source parameter in the Modulation
Bus section and then route to the desired Destination parameter(s), see “The Modulation Bus section”.

132 EUROPA SHAPESHIFTING SYNTHESIZER

 Panel reference
Sound Engines On/Off and Edit Focus section

Engine Select
D Click the On LED buttons to activate the corresponding Sound Engine.
D Click the I, II or III LED radio buttons to select the corresponding Sound Engine for editing.

The Oscillator section

Here is where you choose oscillator waveform and set the wave shape and pitch for the selected Sound Engine.

On
D Click the red rectangular LED button to switch the selected Sound Engine on/off.

Oct
D Set the pitch in octave steps.
Range: 5 octaves.

Semi
D Set the pitch in semitone steps.
Range: 12 semitones (one octave).

Tune
D Change the pitch in steps of 1 cent.
Range: +/- 50 cents (down or up half a semitone).

133 EUROPA SHAPESHIFTING SYNTHESIZER

 Kbd
D Set how much the pitch should track incoming MIDI Notes.
Range: 0% (no tracking (constant pitch)) to 100% (1 semitone per note).

Waveform display
The interactive Waveform display shows the waveform shape in real-time.
• Clicking and dragging vertically in the display changes the Shape parameter, see “Shape”.
• Clicking and dragging horizontally in the display changes the Modifier 1 Amount parameter, see “Amount”.
q See “Recording display movements in the sequencer” for tips about automating display movements.

Waveform selector
D Click the Waveform name box to bring up a menu of the available waveforms.
The wave shapes are shown in the display above and are updated in real-time according to the current settings
and modulations. A great way to understand how the sound actually “looks”.
The waveforms are:
• Basic Analog
A pure sinewave at Shape=0%, gradually transformed via triangle and square towards a sawtooth wave at
Shape=100%.
• Square-Ramp
A square wave at Shape=0%, gradually transformed towards a sawtooth wave at Shape=100%.
• Saw-Triangle
A negative ramp sawtooth wave at Shape=0%, gradually transformed via triangle towards a positive ramp saw-
tooth wave at Shape=100%.
• Pulse Width
A 0% duty cycle pulse wave (silence) at Shape=0 gradually transformed via a 50% duty cycle square wave to-
wards a 100% duty cycle pule wave (silence) at Shape=100%.
q Modulate the Shape parameter from an LFO to achieve PWM, see “Shape Modulation” below.
• Game
A lo-fi “early computer game” type of signal. Turn the Shape knob to change the overtone contents and the octave
transposition.
• Synced Sine
A pure sinewave at Shape=0%. As the Shape is increased, the pitch of the synced sinewave oscillator is raised.
• Formant Sweep
A cosine window modulated by a sinewave. Turn the Shape knob to change the sinewave frequency and thus
sweep through the generated formants.
• Electro Mechanical
This is a simulation of an electric piano. A soft/mellow tone at Shape=0% gradually transformed towards an
agitated signal at Shape=100%, with natural sound at the 12 o’clock position (50%).
• Vocal Cord
A simulation of a vocal cord with a bit of noise modulation. Change the overtone content with the Shape knob.
q Try this together with the Vocal Formants algorithm in the Spectral Filter section to generate “vocal” sounds,
see “Vocal Formant”.
• Karplus-Strong
A physical model of a “string”, generated by sending a short pulse through pitched delay lines. At Shape=0% there
is no damping and at Shape=100% there is full damping, which results in just a short clicking sound.
q Try this together with the “Stretch” algorithm in the Harmonics section to create realistic metallic sounds.

134 EUROPA SHAPESHIFTING SYNTHESIZER

 • Envelope 3-4
This is a special mode where you can manually draw your waveforms in the Envelope 3 and Envelope 4 windows
and then gradually crossfade between the drawn waveforms using the Shape knob. See “Using the Envelope 3
and Envelope 4 curves as Sound Engine waveforms” for information on how to draw your own waveforms.
• FM > FM Ratio (1:1, 1:2, 1:8, 2:1)
These are frequency modulated sine waves with different frequency ratios between the carrier (C: ) and modulator
( :M) signals. Set the frequency modulation amount with the Shape knob.
• FM > FM Feedback
A pure sinewave signal at Shape=0% gradually fed back internally at an 1:1 ratio. The feedback signal is filtered
before fed back to the carrier signal. If you modulate the Shape parameter from e.g. an LFO you will get a similar
result as when using the FM FB Noise waveform without Shape modulation, see “Noise > FM FB Noise” below.
• Noise > S/H Noise
A sample & hold modulated noise. Change the sample & hold rate with the Shape knob. If you play high up on the
keyboard at high Shape values, you get a kind of “pitched noise” sound.
q To get white noise, set Shape to max, set Oct to -1 and turn Kbd to 0.
• Noise > Perlin Noise
A pure sinewave signal modulated by low frequency noise. At Shape=0% the noise has its lowest frequency and
at Shape=100% the noise frequency is higher (but still low-frequent). The character of the signal is similar to the
Band Noise in the Thor synthesizer.
• Noise > Bit Noise
This generates a random lo-fi “digital” bit noise. At Shape=0% the signal is completely silent and with increasing
Shape values the signal is modulated faster and in a wider frequency range.
• Noise > FM FB Noise
A pure sinewave signal at Shape=0% gradually fed back internally at an 1:1 ratio. The feedback signal unfiltered
before fed back to the carrier signal which gives the signal a noisy character.
q To get a cleaner FM signal, use the FM Feedback waveform, see “FM > FM Feedback” above.
• Noise > Freeze Noise
This signal produces a range of noises, from tonal noise up to almost white noise, by amplitude modulating the sig-
nal’s partials with noise.
• Wave Tables >
The Wave Tables sub-menu contains a selection of very useful wave tables. Each wave table features eight wave-
forms that you could crossfade between with the Shape knob.
• User Wave/User Wave Smooth
The User Wave options let you use the external sample you have loaded in the User Wave section (see “The User
Wave and Mixer section”). The oscillator then generates and plays back wavetables (grains) of that sample. The
“User Wave Smooth” algorithm uses a crossfaded loop within each grain, which produces a smoother character to
the sound. Set the playback position in the sample with the Shape knob. Modulate the Shape parameter, for exam-
ple from a negative Envelope ramp, for continuous movement in the sample.

Shape
D Turn the Shape knob to change the shape of the currently selected waveform.
The wave shapes are shown in the display above and are updated in real-time according to the current Shape set-
tings.

Shape Modulation
D Click the Shape Modulation Source box to bring up a menu of the available modulation sources.
The “Inverted” sub-menu contains inverted variations of all modulation sources.
D Set the modulation amount with the Shape Modulation Amount knob.

135 EUROPA SHAPESHIFTING SYNTHESIZER

 D Turn the Velo knob to control the Shape Modulation Amount from Keyboard Velocity.
q If you want other modulation sources or scaling options, use the Mod Bus, see “The Modulation Bus section”.

Phase Sync
D Click the Phase Sync button to force the waveform cycle to always start at the same phase (0 degrees).
When active, the sound character will be the same each time you play the same note. When inactive, the sound
character will vary more or less each time you play the same note.

The Modifiers section

The two Modifiers can be used for modifying the currently selected waveform in various ways. The two Modifiers are
identical in functionality and can be used alone or together (or not at all).

Modifier On/Off
D Click the On/Off LED buttons to activate/deactivate the corresponding Modifier.

Modifier selector
D Click the Modifier name box to bring up a menu of the available Modifier types.
The available Modifier types are:
• Faded Sync
This is oscillator sync but with a crossfade at the sync positions. This makes the effect a little smoother (less
bright) than with regular hard sync, see below.
• Hard Sync
Oscillator hard sync is when one oscillator restarts the period of another oscillator, so that they will have the same
base frequency. If you change or modulate the frequency of the synced oscillator you get the characteristic sound
associated with oscillator sync. Control the frequency of the synced oscillator (and thereby the overtone spectrum)
with the Amount knob.
• Invert
This inverts the waveform phase at a variable position within the waveform cycle. Set the phase angle with the
Amount knob.
• Mirror
This mirrors the waveform cycle (in the time line) at a variable position in the waveform cycle. Set the mirroring
position in the waveform cycle with the Amount knob. At Amount=50% the waveform is completely symmetric.
• DownSample
This lets you quantize the waveform in time, i.e. reduce the sample rate. Set the sample rate reduction amount with
the Amount knob.
• Quantize
This lets you truncate the signal’s bit depth, thus making it possible to achieve that noisy, characteristic “8-bit
sound” for example. Set the bit-reduction amount with the Amount knob.

136 EUROPA SHAPESHIFTING SYNTHESIZER

 • Phase Distort
This distorts the waveform by modulating the start phase of the waveform cycle. This generally creates a brighter
tone towards the extremes of the Amount range (0% and 100%). At Amount=50% the signal is unaffected.
• Self Multiply
This multiplies a copy of the waveform with the original waveform. Set the phase angle of the copied waveform
with the Amount knob.
• Noise Mod
This modulates the waveform with low frequency noise. Perfect for adding e.g. “breath noise” to a signal. Set the
noise modulation amount with the Amount knob.
• Shaping > Wrap
This amplifies the signal above the available headroom and then wraps the peaks down into the available head-
room. This adds quite an aggressive distortion to the sound.
• Shaping > Fold
This amplifies the signal above the available headroom and then “mirrors” the peaks down into the available head-
room. Fold is similar to the Wrap shaping but is generally less “aggressive”.
• Shaping > Hard Clip
This amplifies the signal above the available headroom and then clips the peaks that are above the headroom.
Generally, a signal that is clipped to the maximum would result in a pulse/square shaped waveform.
• Shaping > Soft Clip
Soft clip is similar to hard clip described above, but has a smoother shape at the clipping points and thus generates
less overtones.
• Shaping > Sine Shape
This generates sine shaping distortion to the signal.
• Shaping > Glitch 1
This distorts the waveform by introducing a short low-frequency noise glitch in the waveform cycle, but only in
parts of the waveform that go from zero to positive level.
• Shaping > Glitch 2
This is similar to Glitch 1 described above, but introduces a more high-frequent noise glitch in the waveform cycle.
• Harmonics > Octave
This makes it possible to gradually crossfade between the original signal and a copy of the signal one octave
above. Set the crossfade amount with the Amount knob.
• Harmonics > Fifth
This makes it possible to gradually crossfade between “one octave up” and “one octave+one fifth up”. As soon as
you turn on the Fifth modifier you automatically raise the pitch by one octave. The reason for this is that this is
done by multiplying frequencies, i.e. you crossfade between the double and triple of the original frequency. Set the
crossfade amount with the Amount knob.
• Harmonics > 16 Harmonics
This makes it possible to gradually crossfade between the original signal and copies of the signal at the 16 first
harmonics above the original frequency. Set the position in the harmonic spectrum with the Amount knob.
• Harmonics > Fund. + 16 Harm.
This is the same as “16 Harmonics” described above, except this always keeps the original signal mixed in with the
selected harmonic.
• Harmonics > All 16 Harm.
This gradually adds copies of the original signal at the first 16 harmonics. Turning up the Amount knob will add on
the harmonics one by one until all 16 harmonics are present in the signal.
• Harmonics > Ring Harm.
This multiplies the waveform with a sinewave signal to generate a ring modulator effect. Set the modulator fre-
quency with the Amount knob.

137 EUROPA SHAPESHIFTING SYNTHESIZER

 • FM > FM Ratio (1:1, 1:2, 1:8, 2:1)
These modifiers let you frequency modulate the currently selected Waveform at various ratios. The carrier signal is
the currently selected Waveform (C: ) and the modulator ( :M) is the modifier signal. Set the frequency modulation
amount with the Amount knob.
• FM > FM Feedback
Here, an internally fed back sinewave signal at an 1:1 ratio modulates the waveform (same signal type as in the
“FM > FM Feedback” Waveform).
• Detuning > Unison3
This simulates 2 copies of the original signal. The Amount knob controls the detuning amount and rate.
• Detuning > Unison7
This simulates 6 copies of the original signal. The Amount knob controls the detuning amount and rate.
• Detuning > Ensemble
This simulates a variable number of copies of the original signal. The Amount knob controls the number of copies,
the detuning amount and rate.
• Detuning > Unison 3Oct
This simulates 2 copies of the original signal at +1 and +2 octaves relative to the original signal. The Amount knob
controls the detuning amount and rate.
• Formant
This simulates a formant (body) filter, which produces multiple peaks and notches in the frequency spectrum of
the signal. The Amount knob controls the formant transposition in the frequency spectrum. At Amount=50% the
signal is unaffected. Below 50% the formant is transposed down and above 50% it’s transposed up.
To make the formant static in the frequency spectrum, regardless of which note you play, modulate the Amount
parameter using the “-KEY” (inverted) modulation source with a fairly high value (see “Amount Modulation” below).
This is especially useful if you are using an acoustic instrument sample as User Wave in the Oscillator section.

Amount
D Turn the Amount knob to change the modification amount of the currently selected Modifier.
The wave shapes are updated in real-time and shown in the Waveform display.

Amount Modulation
D Click the Modulation Source box to bring up a menu of the available modulation sources.
The “Inverted” sub-menu contains inverted variations of all modulation sources.
D Set the modulation amount with the Amount Modulation knob.
q If you want other modulation sources or scaling options, use the Mod Bus, see “The Modulation Bus section”.

The Spectral Filter

The signal from the Oscillator section can then be processed by the Spectral Filter. The Spectral Filter features a
wide variety of algorithms that affect the partials of the signal.

138 EUROPA SHAPESHIFTING SYNTHESIZER

 Spectral Filter On/Off
D Click the On/Off LED button to activate/deactivate the Spectral Filter.

Spectral Filter display

The interactive Spectral Filter display shows the filter shape in real-time.
• Clicking and dragging vertically in the display changes the Freq parameter, see “Freq”.
• Clicking and dragging horizontally in the display changes the Resonance parameter, see “Reso”.
q See “Recording display movements in the sequencer” for tips about automating display movements.

Spectral Filter selector

D Click the Spectral Filter name box to bring up a menu of the available filter types.
The available filter types are:
• LP 12
This simulates a standard 12dB/octave lowpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
• LP 24
This simulates a standard 24dB/octave lowpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
• HP 24
This simulates a standard 24dB/octave highpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
• BP 12
This simulates a standard 12dB/octave bandpass filter. Set the center frequency with the Freq knob and the
resonance amount with the Reso knob.
• Par EQ
This simulates a standard single band parametric equalizer with a fixed bandwidth. Set the center frequency with
the Freq knob and the gain/attenuation with the Reso knob.
• Dual Peak
This simulates two 12dB/octave bandpass filter routed in parallel. Set the center frequency of the first bandpass
filter with the Freq knob and the peak separation with the Reso knob.
• Vocal Formant
This simulates the formants of the vocal tract by using multi-peak+notch filters. Change the formant with the Freq
and Reso knobs.
• LP Variable Slope
This simulates a non-resonant lowpass filter with a variable attenuation slope. Set the cutoff frequency with the
Freq knob and the attenuation slope with the Reso knob.
• HP Variable Slope
This simulates a non-resonant highpass filter with a variable attenuation slope. Set the cutoff frequency with the
Freq knob and the attenuation slope with the Reso knob.
• Comb +
This simulates a multi notch filter, great for phaser types of effects. Set the cutoff frequency of the first notch with
the Freq knob and the attenuation amount - and consequently the bandwidth - of the notches with the Reso knob.
The difference between “Comb +” and “Comb –” (see below) is in the position of the peaks in the spectrum. The
main audible difference is that the “Comb –” version causes a bass cut.

139 EUROPA SHAPESHIFTING SYNTHESIZER

 • Comb -
This simulates a comb filter with a positive feedback loop - but without feed forward - ideal for flanger and phaser
types of effects. Set the cutoff frequency of the second peak with the Freq knob and the resonance amount with
the Reso knob. The difference between “Comb +” (see above) and “Comb –” is in the position of the peaks in the
spectrum. The main audible difference is that the “Comb –” version causes a bass cut.
• Resonator 1,2 and 3
The three Resonator algorithms contain formant filter tables that simulate various body resonances (multi-
peak+notch filters). Set the position in the formant tables with the Freq knob and the resonance with the Reso
knob.
• Envelope 4
This is a special mode where you can manually draw your own filter curve in the Envelope 4 window. You then con-
trol the cutoff/center frequency with the Freq knob and the resonance with the Reso knob. See “Using the Enve-
lope 4 curve as a Spectral Filter curve” for information on how to draw your own filter curves.
• User Wave
This utilizes a filter generated from FFT analyses of the external sample you have loaded in the User Wave section
(see “The User Wave and Mixer section”). Transpose the formant up/down in the frequency spectrum with the
Freq knob and change the filter’s position in the sample with the Reso knob.
q To create a classic “vocoder” effect, load a vocal/speech sample in the User Wave section. Then, set the Freq
knob to 50%, the Reso knob to 0% and the KBD knob to 0%. Then, have one of the Envelopes modulate the
Reso parameter using the Modulation Bus (see “The Modulation Bus section”). Create a positive ramp (in-
verted sawtooth) envelope and set the Reso modulation amount to 100%.

Freq
D Set the cutoff/center frequency of the currently selected Spectral Filter type.

Reso
D Set the resonance amount of the currently selected Spectral Filter type.

Frequency Modulation
D Turn the Kbd knob to set the keyboard tracking amount.
At 0% the filter is static and doesn’t track the keyboard at all. At 100% the filter tracks the keyboard 1 semitone
per note. At values above 0% you can also see the filter curve move sideways in the Spectral Filter Display
depending on where on the keyboard you play.
D Click the Frequency Modulation Source box to bring up a menu of the available modulation sources.
The “Inverted” sub-menu contains inverted variations of all modulation sources.
D Set the modulation amount with the Frequency Modulation Amount knob.
D Turn the Velo knob to control the Frequency Modulation Amount from Keyboard Velocity.
q If you want other modulation sources or scaling options, use the Mod Bus, see “The Modulation Bus section”.

The Harmonics section

The Harmonics section offers extensive modulation possibilities of the partials of the signal. For most algorithms the
partials’ characteristics is displayed in the Spectral Filter display, see “Spectral Filter display”.

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 Harmonics On/Off
D Click the On/Off LED buttons to activate/deactivate the Harmonics section.

Harmonics selector
D Click the Harmonics name box to bring up a menu of the available harmonic algorithms.
The available Harmonics types are:
• Random Gain
This alters the gain for each of the partials in the signal in a random fashion. Turn the Pos knob to change the
randomization “pattern” and the Amount knob to change the partial gain levels in the “pattern”.
• Harmonic 1-8 Mix
This alters the gain/attenuation for the first eight partials in the signal. Turn the Pos knob to crossfade between
the partials and the Amount knob to change the partial gain/attenuation level. Amount levels below 50% attenu-
ate all partials but the one selected with the Pos knob. Amount levels above 50% attenuate the partial selected
with the Pos knob.
• Odd-Even
This alters the gain/attenuation of the partials in the signal. At Pos=0% the Amount knob controls the mix strictly
between the odd and even partials in the signal. At other Pos values, the gain/attenuation is not strictly on odd and
even partials. At Amount=50% the Pos value has no effect.
• Stretch
This stretches or squeezes all partials (overtones) - except for the fundamental - in the signal, up or down in the
frequency range. Perfect for turning a harmonic signal into a more inharmonic one. Change the stretch amount
with the Amount knob. The Pos knob controls the start phase of all the overtones. At Pos=0% all partials start at
the same phase. When Amount is set fairly high the Pos parameter have little or no influence on the sound.
q Try this with the Karplus-Strong waveform to create really realistic metallic sounds, see “Karplus-Strong”!
• Ensemble
This is the perfect algorithm for really dense pad sounds. The Ensemble algorithm simulates a type of chorus
effect by utilizing noise modulation of the partials. Set the noise frequency with the Pos knob and the mix level with
the Amount knob.
• Ensemble Sparse
The Ensemble Sparse algorithm also utilizes noise modulation of the partials, but here a lot of noise frequency
bands are cut out. This makes Ensemble Sparse sound more animated and less smooth than the Ensemble algo-
rithm described above. Set the noise frequency with the Pos knob and the mix level with the Amount knob.
• HF Noise
This amplitude modulates the high frequencies in the signal with (high-frequency) noise, perfect for adding “breath
noise” to the signal, for example. Set the noise frequency with the Pos knob and the noise mix level with the
Amount knob.
• Harmonic Lag A-R
The Harmonic Lag A-R algorithm is designed especially for use with the User Wave algorithm in the Spectral Filter
(see “User Wave”) to create vocoder effects. The Harmonic Lag A-R algorithm controls the Spectral Filter - so the
Spectral Filter has to be on for this to work!
Note that the Harmonic Lag A-R algorithm works on the filter partials - not the oscillator’s signal partials. Set the
Attack time of the filter partials with the Pos knob and the Release time with the Amount knob. These controls
work similarly to the Attack and Decay parameters on the BV512 Vocoder device.

Pos
D Turn the Pos knob to change the frequency of the currently selected Harmonics algorithm.
The frequency spectrum is updated in real-time and shown in the Spectral Filter display.

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 Amount
D Turn the Amount knob to change the intensity of the currently selected Harmonics algorithm.
The frequency spectrum is updated in real-time and shown in the Spectral Filter display.

The Unison section

The Unison function generates detuned duplicates of the signal in pairs on either side of the original signal’s pitch.

Unison On/Off
D Click the On/Off LED button to activate/deactivate the Unison section.

Unison display
The Unison display shows the unison characteristics, as set with the controls in the Unison section. Note, though, that
this display is not interactive like the Waveform and Spectral Filter displays.

Unison Type selector

D Click the Unison name box to bring up a menu of the available Unison types.
The available Unison types are:
• Normal
This generates duplicates of the signal on either side of the original signal’s pitch.
• Fourth
This generates duplicates of the signal on either side of the fourth above the original signal’s pitch.
• Fifth
This generates duplicates of the signal on either side of the fifth above the original signal’s pitch.
• Octave Down
This generates duplicates of the signal on either side of the original signal’s pitch - one octave down.
• Phase Only
This generates duplicates of the signal on either side of the original signal’s pitch. The Detune parameter (see be-
low) now controls the phases of the signal copies, instead of the detuning. This is great for creating wide stereo
sounds without lots of detuning.
If Phase Sync is active in the Oscillator (see “Phase Sync”), all signals in the Unison function get the same relative
start phases each time you play the same note. The original signal always has the phase 0 degrees.

Count
D Set the number of desired signals in the unison effect.
Range: 1-7.
Note that for even numbers, the original signal is represented by two duplicates.

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 Blend
D Set the mix between the original signal and the duplicates.
For even “Count” numbers (see above), one of the duplicates represents the original signal in the mix.

Detune
D Set the pitch detuning of the signal duplicates.
If the “Phase Only” Unison type is selected (see above), the Detune knob controls the phases of the signal dupli-
cates instead of the pitch detuning.
q In the “Phase Only” scenario it could also be a good idea to modulate the Detune parameter from an LFO
using the Modulation Bus (see “The Modulation Bus section”), to create nice phasing effects.

Spread
D Set the stereo spread amount of the signal duplicates.

The User Wave and Mixer section

The User Wave and Mixer section is where you can load an external sample to use in the Oscillators (see “User
Wave/User Wave Smooth”) and/or in the Spectral Filter (see “User Wave”). In the Mixer you can mix and pan the sig-
nals from the three Sound Engines before sending them to the Filter, Amp and Multi FX sections.

Sample Select/Load/Edit buttons

D Load a sample using drag & and drop, or by clicking the Browse sample button, or by using the Up/Down but-
tons to scroll and load a sample from the currently selected folder.
• It’s possible to load stereo samples. However, the sample will be automatically converted to mono in Europa.
! Like with the other “sampler” devices in Reason, the Europa patch does not include the actual sample - only a
reference to it. Therefore, the sample has to be stored separately on disk or in a ReFill on your computer.
! If the sample length is a multiple of 2048 samples (“Serum compatible”), no pitch detection is being made.
Then, Europa automatically assumes that 2048 samples is one complete waveform cycle (period). If the sample
is not an exact multiple of 2048 samples, a pitch detection is being performed by Europa to determine the
pitch. Longer samples (with a stable pitch) will render better pitch detection results, so don’t use very short
samples.

Level
D Set the volume of the corresponding Sound Engine signal with the Level slider.
Range: -Inf to +6.0dB.

Pan
D Set the panning of the corresponding Sound Engine signal in the stereo panorama.

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 The Filter section

Routing buttons
D Click the red LED buttons to route the corresponding Sound Engine signals to the Filter section.
If deactivated, the signal bypasses the Filter and goes straight to the Amp section, see “The Amplifier section”.

Drive
D Turn the Drive knob to amplify and introduce an overdrive type of distortion to the Sound Engine signal(s)
in the filter.

Filter Type selector

D Click the Filter Type name box to select one of the following filter types:
• SVF HP 12dB
A state variable (SVF) highpass filter with a 12dB/octave slope. This filter is similar to the State Variable Filter in
the Thor synthesizer.
• SVF BP 12dB
A state variable (SVF) bandpass filter with 12dB/octave slopes. This filter is similar to the State Variable Filter in
the Thor synthesizer.
• SVF LP 12dB
A state variable (SVF) lowpass filter with a 12dB/octave slope. This filter is similar to the State Variable Filter in
the Thor synthesizer.
• SVF Notch
A state variable (SVF) notch filter. This filter is similar to the State Variable Filter in the Thor synthesizer.
• Ladder LP 24dB
A ladder-type lowpass filter with a 24dB/octave slope. The resonance peak more narrow in this filter type than in
the MFB LP 24dB filter (see below). The filter can be driven to self-oscillate.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!
• MFB LP 12dB
A multiple feedback (MFB) lowpass filter with a 12dB/octave slope.
• MFB LP 24dB
A multiple feedback (MFB) lowpass filter with a 24dB/octave slope. The resonance peak is wider in this filter type
that in the Ladder filter (see above). The filter can be driven to self-oscillate.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!

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 • MFB HP 24dB
A multiple feedback (MFB) highpass filter with a 24dB/octave slope.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!
• K35 LP 12dB
An “early MS-20 type” of lowpass filter with a 12dB/octave slope. The filter can be driven to self-oscillate.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!

Reso
D Set the resonance amount.
In the SVF Notch filter, the Reso knob controls the width of the notch - from wide to narrow.

Freq
D Set the cutoff frequency (for the HP and LP filter types) or the center frequency (for the BP filter type).

Frequency Modulation
D Turn the Kbd knob to set the keyboard tracking amount.
At 0% the filter is static and doesn’t track the keyboard at all. At 100% the filter tracks the keyboard 1 semitone
per note.
D Click the Frequency Modulation Source box to bring up a menu of the available modulation sources.
The “Inverted” sub-menu contains inverted variations of all modulation sources.
q Use one of the Envelopes (see “The Envelopes section”) as modulation source to create a Filter envelope.
D Set the modulation amount with the Frequency Modulation Amount knob.
D Turn the Velo knob to control the Frequency Modulation Amount from Keyboard Velocity.
q If you want other modulation sources or scaling options, use the Mod Bus, see “The Modulation Bus section”.

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 The Amplifier section

The Amplifier section contains a standard ADSR envelope, which controls the amplitude of the signals from all three
Sound Engines equally.
q To create an “amp envelope” for a separate Sound Engine, have a look at “Creating an individual “pre amp en-
velope” for a Sound Engine”.
The picture below shows the various stages of the ADSR envelope:

Level

Gain
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value
set with the Gain knob (see below). How long this should take, depends on the Attack setting. If the Attack is set to
“0”, the Gain level is reached instantly. If the Attack value is raised, it will take longer time before the Gain level is
reached.

D(ecay)
After the Gain level has been reached, the level starts to drop. How long this should take is governed by the Decay
parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to “0”,
the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the volume
gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to drop to
some other value than zero, you raise the Sustain parameter.

S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain to
full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain param-
eter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays
there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).

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 But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Gain level, then grad-
ually decreases to finally land to rest on a level somewhere in-between zero and the Gain level. Note that Sustain
represents a level, whereas the other envelope parameters represent times.

R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume to
drop back to zero after you release the key.

Pan
D Set the panning of the output signal from the Amplifier in the stereo panorama.
q Since Pan works individually per voice, you can assign e.g. Keyboard Velocity or an Envelope in the Modula-
tion Bus to control the Pan effect, see “The Modulation Bus section”.

Gain
D Set the desired maximum level for the Amplifier with the Gain knob.
This is the maximum level the envelope will reach after the Attack stage is completed (see above).
q If you want to create a tremolo effect, assign “Gain” as Destination and an LFO as Source in the Modulation
Bus section, see “The Modulation Bus section”.

Velo
D If you want the Gain level to be controlled from keyboard velocity, turn up the Velo knob.

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 The Envelopes section

The Envelopes section features four separate polyphonic (one per voice) general purpose envelope generators, that
can be assigned to control selectable parameter(s) in the Modulation Bus section.
The Envelopes are extremely flexible, and you can draw your own custom modulation shapes by clicking and drawing
in the display area. There are also a number of preset shapes that you can use as starting points (or use as is). If you
use Loop mode, you could turn the envelope into a kind of LFO.
See “The Modulation Bus section” for details on how to assign the Envelopes to the desired destination(s).

Envelope 1, 2, 3 and 4
D Click one of the Envelope 1, 2, 3 or 4 buttons to select which envelope to edit:

Preset
1. Click the Preset button to bring up a palette of envelope preset curves:

2. Click the desired envelope preset curve to place it on the display.

Let’s select a standard ADSR style of envelope curve:

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 Adding a Sustain stage
D Click the Sustain button to add a sustain stage to the envelope:

The vertical red marker that appears indicates at what level (and where) the envelope will stay sustained until you
release the key.
D Drag the sustain marker sideways to move the sustain stage to the desired position:

D To remove the sustain stage, click the Sustain button.

Adding and removing envelope points

D Double click, or hold down [Ctrl](win) or [Cmd](Mac) and click in the envelope display to add points to the
envelope curve:

D To remove a point, double click, or hold down [Ctrl](win) or [Cmd](Mac) and click, on an existing point on the
envelope curve.

Changing the envelope curve shape

D Click a line segment (between two points) and drag up/down to change the curve shape:

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 Looping the envelope
D Click the Loop button to turn the envelope into a kind of LFO.
If there was previously a sustain stage in the envelope, this will automatically be disabled when you click the Loop
button.

Here we have edited a stepped curve from the Presets. We have also enabled Beat Sync and set the length/rate
to 4/4. This means that each step in the curve now represents an 1/8th note.
• Key Trig means the envelope restarts when you play a note.
• You can choose whether the envelope should send out a bipolar value or unipolar one (0-100%).
• If Global is on, the envelope will be common for all voices.

Editing levels only

D To restrict the editing to levels only, without affecting the time positions, click the Edit Y-Pos button:

In this mode you cannot change the time positions of the envelope points, only their levels (height). This is extra
useful with a stepped Preset curve, because dragging up or down will change the value of an entire segment, turn-
ing the Envelope into a pseudo-sequencer.
! To be able to adjust the level of a segment, the two points on either side of the segment have to be on the ex-
act same time positions. Otherwise, only the closest point will be changed. Also, any inclining/declining seg-
ment will automatically turn horizontal when edited:

Adjusting the level of a segment.

Creating “free form” envelope curves

In the Edit Y-Pos mode, you can also draw “free form” curves:
D To continuously add new consecutive points, hold down [Ctrl](win) or [Cmd](Mac) and drag in the envelope
display:

D To erase points, hold down [Shift] and [Ctrl](win) or [Cmd](Mac) and drag in the envelope display.

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 Using the Envelope 3 and Envelope 4 curves as Sound Engine waveforms
As a special feature you can use the Envelope 3 and Envelope 4 curves as waveforms for the Sound Engines:
1. Select Envelope 3 and create/modify a curve in the envelope display:

2. Select Envelope 4 and create/modify another curve:

3. In the Waveform selector for a Sound Engine, select the “Envelope 3-4” waveform:

4. Turn the Shape knob to crossfade between the curves/waveforms of Envelopes 3 and 4:

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 Using the Envelope 4 curve as a Spectral Filter curve
Another special feature is that you could use the Envelope 4 curve as a filter curve in the Spectral Filter:
1. Select Envelope 4 and create/modify a curve in the envelope display:

2. In the Filter selector in the Spectral Filter section, select “Envelope 4”:

3. Turn the Freq knob to change the curve’s “cutoff” frequency and the Reso knob to change the curve’s “reso-
nance”.
At Reso=0% the curve is completely flat (no gain or attenuation) and at Reso=100% the resonance corresponds
exactly to the Envelope 4 curve.

The LFO section

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO
modulate the pitch of a signal to produce vibrato, but there are countless other applications for LFOs.
The LFO section features three separate general purpose LFOs, that can be assigned to control selectable parame-
ter(s) in the Modulation Bus section, see “The Modulation Bus section”.
1. Select which of the three LFOs you want to edit by clicking one of the LFO 1, LFO 2 and LFO 3 buttons.
2. Select an LFO waveform by clicking the spin controls to the right of the waveform display, or by click-holding
in the display and dragging up or down.
Besides the standard waveforms (sine, triangle, pulse, etc.) there are random, slope and stepped waveforms. The
shape of the waveforms are shown in the display.
3. Set the LFO frequency with the Rate knob.
D Click the Beat Sync button to sync the LFO to the main sequencer tempo.
The Rate parameter now controls the time divisions.
D Click the Key Sync button to restart the LFO at every new Note On.
D Click the Global button to make the LFO common for all voices (monophonic).
D Turn the Delay knob to introduce a delay before the LFO modulation kicks in after a note is played.
Turn clockwise for longer delay times.

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 The Effects section

The Effects section features six different effect modules that can be freely reordered by dragging & dropping. Most
of the effect parameters are also available as destinations in the Modulation Bus, see “The Modulation Bus section”.
At the top of the Effects section are six Effect buttons. Click any of these to bring up the control panel for the corre-
sponding effect. Below the Effect buttons are the On/Off buttons for the individual effects. Click these to activate the
effects.

Reordering the effects

D To define the order of the effects in the serial chain, click and hold on the desired Effect button and drag side-
ways to the desired position:

Moving the Reverb effect to another position in the effects chain.

You can reorder the effects at any time.

Reverb

This is a stereo reverb, routed as a send effect.

• Decay
This governs the length of the reverb effect.
• Size
Sets the emulated room size, from small room to large hall. Middle position is the default room size.
Lowering this parameter results in a closer and gradually more “canned” sound. Raising the parameter results in a
more spacey sound, with longer pre-delay.
• Damp
Raising the Damp value cuts off the high frequencies of the reverb, thereby creating a smoother, warmer effect.
• Amount
Use this parameter to adjust the send level to the Reverb effect.
If you play a note, have a long delay Decay time and turn down Amount, the reverberation will continue.

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 Delay

This is a stereo delay, routed as a send effect.

• Sync
Activate Sync to sync the delay time to the main sequencer tempo.
• Time
This sets the time between the delay repeats. If Sync is active (see above), the Time parameter now controls the
time divisions.
• Ping Pong
Activate Ping Pong to have the delay repeats alternating between left and right in the stereo panorama. The effect
is also dependent on the Pan parameter (see below).
• Pan
Sets the panning of the delay repeats in the stereo panorama. If Ping Pong is active (see above) the Pan knob
controls the panning of the initial delay repeat as well as the total stereo spread of the remaining repeats.
• FB
The FB (feedback) parameter determines the number of delay repeats.
• Amount
Use this parameter to adjust the send level to the Delay effect.
q If you play a note, have a long delay feedback and turn down Amount, the echoes will continue. This allows for
automated “triggered delay” fx.

Distortion

The Distortion effect features six different types of distortion.

D Select distortion type with the switch.
“Dist” produces a dense, rich analog type of distortion.
“Scream” produces a less bright type of distortion.
“Tube” emulates a tube type of distortion.
“Sine” is a sine shaping distortion.
“S/H” gives the effect of sample rate reduction.
“Ring” is a ring modulator effect.
• Drive
Sets the overdrive/feedback level of the selected distortion.
• Tone
This is a lowpass filter and sets the tone of the selected distortion.
• Amount
Sets the Dry/Wet amount of the distortion.

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 Compressor

This is a stereo compressor.

• Attack
This governs how quickly the compressor will apply its effect when signals rise above the set threshold. If you raise
this value, the response will be slower, allowing more of the signal to pass through the compressor unaffected.
Typically, this is used for preserving the attacks of the sounds.
• Release
When the signal level drops below the set threshold, this determines how long it takes before the compressor lets
the sound through unaffected. Set this to short values for intense, “pumping” compressor effects, or to longer val-
ues for a smoother change of the dynamics.
• Thres
This is the threshold level above which the compression sets in. Signals with levels above the threshold will be af-
fected, signals below it will not. In practice, this means that the lower the Threshold setting, the more the compres-
sion effect.
• Ratio
This specifies the amount of gain reduction applied to the signals above the set threshold.

Phaser/Flanger/Chorus

This is a stereo Phaser/Flanger/Chorus.

D Select effect type with the Phaser/Flanger/Chorus switch.
The selected effect type is displayed on the Effect button.
• Depth
Sets the depth of the selected effect. To get a static sound, set Depth to zero.
• Rate
Sets the rate/speed of the modulation.
• Spread
Sets the stereo width of the effect.
• Amount
Sets the Dry/Wet amount of the effect.
q It’s also possible to modulate the start/center frequency of the Phaser/Flanger/Chorus using the “Chorus/
Flanger/Phaser > Frequency” destination in the Modulation Bus, see “The Modulation Bus section”.

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 EQ

The EQ effect is a single band parametric equalizer with adjustable Q-value and Gain.
• Freq
Sets the center frequency of the EQ band.
• Q
Sets the bandwidth of the EQ band, from wide to narrow.
• Gain
Sets the gain/attenuation of the EQ band, from -18dB to +18dB.

The Modulation Bus section

The Modulation Bus section is used for routing a modulation Source to one or two modulation Destinations each.
This creates a very flexible routing system that complements the “pre-wired” routing in Europa.
The Modulation Bus section in Europa is derived from the one in the Reason Thor Polysonic Synthesizer device, so if
you are familiar with Thor, you will quickly find your way around in Europa’s modulation bus.
There are eight “Source –> Destination 1 –> Destination 2 –> Scale” busses, of which the first four have pre-as-
signed sources. However, these four pre-assigned sources can be easily changed if you like.
A Source parameter can modulate two different Destination parameters per bus (with variable Amount settings).
Each bus also has a Scale parameter that affects the relative modulation Amount for both Destinations.
q Note that it is possible to assign the same source parameter as Source in several busses. This allows you to
control more than two Destination parameters from the same Source.

1. Select the desired Source parameter by clicking in the corresponding Source box and selecting from the list.
The following parameters can be used as modulation Sources:

| Parameter | Description
Velocity This applies modulation according to the Keyboard Velocity values (how hard or soft you strike the MIDI
keyboard keys).
LFOs (LFO 1, LFO 2 and LFO 3) This allows you to modulate parameters from LFO 1, LFO 2 and LFO 3 respectively.
Envelopes (Amp Envelope, Envelope 1, This allows you to modulate parameters from any of the Envelopes.
Envelope 2, Envelope 3, Envelope 4, As a special feature you can also modulate parameters from the multiplied signal of Envelope 3 and
Envelope 3 * Envelope 4, Envelope 3 * Envelope 4, as well as from the multiplied signal of Envelope 3 and LFO 3.
LFO 3)
Mod Wheel This allows you to modulate parameters from the Mod Wheel.

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 | Parameter | Description
MW Latched This allows you to modulate parameters based on the current Mod Wheel value at a given Note On.
Pitch Wheel This allows you to modulate parameters from the Pitch Bend control.
Breath This allows you to modulate parameters from the Breath performance controller
Expression This allows you to modulate parameters from the Expression performance controller
Aftertouch This allows you to modulate parameters from Keyboard Aftertouch (channel aftertouch)
Sustain This allows you to modulate parameters from a connected sustain pedal. Note that continuous sustain
data (0-127) is supported - not just on/off.
Key This is keyboard tracking. If a positive Amount value is used and the destination is filter frequency, the
filter frequency will track the keyboard, i.e. increase with higher notes.
Random This sends out a random value each time a new note is played.
Key In Octave This allows you to modulate parameters based on 12 separate note values (within each octave).
Noise This allows you to modulate parameters from white noise.
Polyphony This allows you to modulate parameters based on the number of playing voices at a given time.
Last Velocity This applies modulation according to the latest Keyboard Velocity value (how hard or soft you hit the
latest MIDI keyboard key).
CV Input 1/2/3/4 This takes the current value on the CV 1/CV 2/CV 3/CV 4 inputs on the rear panel and sends to the
desired destination.
CV Input 1/2/3/4 Latched This allows you to modulate parameters based on the current CV 1/CV 2/CV 3/CV 4 value at a given
Note On.

Modulation Bus Source parameters.

2. Set the Amount for the first Destination by turning the corresponding Amount knob, or by clicking and drag-
ging vertically in the corresponding Amount box.
! Note that the Amount range is +/-100. This means that the Amount value can exceed the modulated parame-
ter’s range. When this happens, the modulated parameter simply stays at its extreme value until the Amount
value gets within the parameter’s range again.
3. Select the first Destination parameter by click-holding the red arrow symbol to the right of the corresponding
Destination box.
4. While click-holding, drag to the desired destination parameter on the panel:

Assigning LFO 1 Rate as a Destination for Envelope 1.

As you hover over a valid destination control on the panel, the parameter name is automatically displayed in the
Destination box in the Modulation Bus.
5. To assign the currently selected Destination control, release the mouse button.
D Alternatively, click the desired Destination box and select the Destination parameter from the list.

157 EUROPA SHAPESHIFTING SYNTHESIZER

 The following parameters can be used as modulation Destinations:

| Parameter | Description
Engine: Pitch This affects the (full range) pitch of the Oscillator.
Engine: Shape This affects the Shape parameter in the Oscillator section.
Engine: Mod 1 Amount This affects the Modifier 1 Amount parameter in the Sound Engine.
Engine: Mod 2 Amount This affects the Modifier 2 Amount parameter in the Sound Engine.
Engine: Filter Freq This affects the Spectral Filter Frequency parameter in the Sound Engine.
Engine: Filter Res This affects the Spectral Filter Resonance parameter in the Sound Engine.
Engine: Harmonics Pos This affects the Harmonics Pos parameter in the Sound Engine.
Engine: Harmonics Amount This affects the Harmonics Amount parameter in the Sound Engine.
Engine: Unison Count This affects the Unison Count parameter in the Sound Engine.
Engine: Unison Detune This affects the Unison Detune parameter in the Sound Engine.
Engine: Unison Blend This affects the Unison Blend parameter in the Sound Engine.
Engine: Unison Spread This affects the Unison Spread parameter in the Sound Engine.
Mixer: Level This affects the Sound Engine Level in the Mixer section.
Mixer: Pan This affects the Sound Engine Pan in the Mixer section
Filter: Drive This affects the Drive parameter in the Filter section.
Filter: Freq This affects the Frequency parameter in the Filter section.
Filter: Reso This affects the Resonance parameter in the Filter section.
Amplifier: Gain This affects the Gain parameter of the Amplifier section.
Amplifier: Pan This affects the Pan parameter of the Amplifier section.
Amp Envelope: Attack This affects the Attack time of the Envelope in the Amplifier section.
Amp Envelope: Decay This affects the Decay time of the Envelope in the Amplifier section.
Amp Envelope: Sustain This affects the Sustain level of the Envelope in the Amplifier section.
Amp Envelope: Release This affects the Release time of the Envelope in the Amplifier section.
LFOs: Delay This affects the LFO Delay parameters.
LFOs: LFO Rate This affects the LFO Rate parameters.
Envelopes: Env Rate This affects the Envelope Rate parameters.
Portamento This affects the Portamento Time parameter.
CV Outputs: CV1/2/3/4 Out This sends out the source modulation value(s) on the CV1/2/3/4 Output on the rear panel.
Reverb: Decay This affects the Decay parameter in the Reverb effect.
Reverb: Amount This affects the Amount parameter in the Reverb effect.
Delay: Time This affects the Time parameter in the Delay effect.
Delay: Feedback This affects the FB parameter in the Delay effect.
Delay: Amount This affects the Amount parameter in the Delay effect.
Delay: Pan This affects the Pan parameter in the Delay effect.
Dist: Drive This affects the Drive parameter in the Dist effect.
Dist: Tone This affects the Tone parameter in the Dist effect.
Dist: Amount This affects the Amount parameter in the Dist effect.
Compressor: Release This affects the Release parameter in the Compressor effect.
Compressor: Ratio This affects the Ratio parameter in the Compressor effect.
Chorus/Flanger/Phaser: Frequency This affects the center frequency of the Chorus/Flanger/Phaser effects.
Chorus/Flanger/Phaser: Amount This affects the Amount parameter of the Chorus/Flanger/Phaser effects.
Par EQ: Frequency This affects the Freq parameter in the EQ effect.
Par EQ: Gain This affects the Gain parameter in the EQ effect.

Modulation Bus Destination parameters.

158 EUROPA SHAPESHIFTING SYNTHESIZER

 6. Set the Amount for the second Destination (if desired) by turning the corresponding Amount knob, or by click-
ing and dragging vertically in the Amount box for the second destination.
7. If desired, select a second Destination parameter by click-holding the blue arrow symbol to the right of the
corresponding Destination box, and dragging to the desired control on the panel.
8. If desired, click the Scale box and select a Scale parameter.
The available Scale parameters are the same as the Source parameters, see “Modulation Bus Source parame-
ters.”.
9. Turn the Scale Amount knob, or click the Amount box to the left of the Scale box and move the mouse pointer
up or down to set a Scale Amount value.
Both positive and negative Scale Amount values can be set (+/- 100%). If you, for example, are using the Mod
Wheel as Scale parameter and don’t want any modulation when the Mod Wheel is set to zero, set the Scale
Amount parameter to 100%. Then, there will be no effect when the Mod wheel is set to zero, and full modulation
when the Mod Wheel is all the way up.
• How much modulation will be applied when the Scale parameter is set to maximum is governed by the to Des-
tination Amount parameter(s).
• How much the Scale parameter controls the modulation is set with the Scale Amount parameter.
D To clear an assigned Source, Destination or Scale parameter, hold down [Ctrl](Win) or [Cmd](Mac) and click
the Source/Destination/Scale box. Alternatively, click the Source/Destination/Scale box and select “Off” from
the list.
D To reset an Amount value to 0, hold down [Ctrl](Win) or [Cmd](Mac) and click the desired Amount box or knob.
D To clear an entire modulation assignment (a whole row), click the circular X button to the right of the corre-
sponding Scale box.

159 EUROPA SHAPESHIFTING SYNTHESIZER

 Connections

! Remember that CV connections are NOT stored in the Europa patches! If you want to store CV connections
between devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Europa from another CV/Gate device (typically a Matrix
or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/
off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

CV Modulation inputs and outputs

These assignable control voltage (CV) inputs and outputs can be used for modulation of and from assigned Source
and Destination parameters in the Modulation Bus section, see “The Modulation Bus section”.

Audio Output
These are the main audio outputs. When you create a new Europa device, these outputs are auto-routed to the first
available outputs in the I/O device.

160 EUROPA SHAPESHIFTING SYNTHESIZER

 Tips and Tricks
Creating an individual “pre amp envelope” for a Sound Engine
There might be situations where you want to control the amplitude envelopes separately for each Sound Engine.
Let’s say you have a plucked sound with a fairly long release in Sound Engine 2 and then you want to slowly fade in
a pad sound from Sound Engine 1. Since the built-in Amp Envelope controls all three Sound Engines together, you
could use the following “workaround”:
1. Assign “Envelope 1” as Source and “Mixer: Eng1 Level” as Destination in the Modulation Bus.
Turn the Amount knob to 100 in the Modulation Bus.

2. Create your “fading” envelope curve in the Envelope 1 display.

Click the Sustain button in the Envelopes display if you want to have a sustain stage in your envelope.

3. Set the Sound Engine 1 Mixer Level slider to zero.

4. As you play the keyboard, Envelope 1 will now fade in the signal from Sound Engine 1, while the signal from
Sound Engine 2 is only controlled by the built-in Amp Envelope.

! Note that the built-in Amp Envelope’s settings will also affect the “fading pad” sound from Sound Engine 1,
since all Sound Engine signals eventually pass through the Amp Envelope.

161 EUROPA SHAPESHIFTING SYNTHESIZER

 Recording display movements in the sequencer
If you are in “experimentation mode” and want to try out some wild waveform and/or Spectral Filter tweaking, you can
record automation of your interactive display movements in the main sequencer:
1. Record some notes on the sequencer track in the main sequencer and then hit stop.
2. Hit Record again in the main sequencer and click and drag in the Waveform display during recording:

3. Hit Stop in the sequencer when you are done recording.

Any vertical movements have now been recorded as Shape parameter automation and any horizontal movements
have been recorded as Modifier 1 Amount parameter automation.
4. If you like, hit Record again in the Reason sequencer and click and drag in the Spectral Filter display during
recording:

5. Hit Stop in the main sequencer when you are done recording.
Any vertical movements have now been recorded as Spectral Filter Resonance parameter automation and any
horizontal movements have been recorded as Spectral Filter Frequency parameter automation.

162 EUROPA SHAPESHIFTING SYNTHESIZER

Chapter 12
Grain Sample
Manipulator
 Introduction

The Grain Sample Manipulator is a very advanced sampler and granular synthesizer, which offers sonic possibilities
far beyond the ordinary. Despite its vast sonic capabilities, Grain has a straight-forward user interface, designed for
experimentation.
Grain uses samples as base for sound generation. You could load a sample from your computer and then select var-
ious types of sample playback modes and algorithms to manipulate and process the audio. You could also use Grain
as a traditional sample player and just play back samples in a regular fashion.
A number of filter and modifier algorithms make it possible to modulate and control the audio further. The extensive
Envelopes section and Modulation Bus section allow for detailed and flexible modulation and control. Grain also fea-
tures a flexible and great-sounding multi-effect to spice up your sounds even more.
Don’t forget to check out the Grain videos here!
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

164 GRAIN SAMPLE MANIPULATOR

 A few words about granular synthesis
Grain utilizes “granular synthesis” to generate sounds. This synthesis method results in playback of a series of snip-
pets of audio data - grains - “extracted” from an audio sample. The grains could be of a selectable length and spac-
ing, and could be from anywhere in the original sample. The grains could also be played back in a number different
ways - with or without crossfades between the grains.
The picture below shows the basic principle of granular synthesis:

Level

Original sample

Time

5 “extracted” grains

Level
The resulting signal is generated by
appending and crossfading the grains.

Time

An example of a signal generated from 5 grains of a sample.

Here is what happens in the example above:
• The original sample at the top is used as base for the granular synthesis.
• 5 grains (of the same lengths and the same distances between them) are “extracted” from the original sample.
The distance between the grains is determined by the current sample playback speed. The grains could contain
common audio data in some parts (like in the beginnings and ends in the example above).
• The 5 grains are then placed after one another, partly overlapping each other.
The distance between the grains is determined by the playback rate.
• When the grains are played back, big parts of the grains are played back together (since they are overlapping).
In the example above, there are also crossfades between the grains to make the overlaps smoother.
Note that the picture above only describes one basic example of granular synthesis - the “Long Grains” playback
algorithm in Grain. Grain uses a number of different granular synthesis and spectral synthesis techniques, with
different functionality and characteristics.

165 GRAIN SAMPLE MANIPULATOR

 Panel overview
The Grain front panel contains the following sections:

1 2
3

4 5

6 7 8

9 10

The Grain front panel sections.

• 1. MIDI Note On LED.
• 2. Patch Selector (for browsing, loading and saving patches), Polyphony and Master Volume controls.
• 3. Sample section (for sample loading and sample playback functions).
• 4. Playback Algorithm and Oscillator section.
• 5. Filter and Amplifier sections.
• 6. Global performance and “play” controls.
• 7. Envelopes section.
• 8. LFO section.
• 9. Modulation Bus section.
• 10. Effects section.

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 Playing and using Grain
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.
! Like with the other sampler devices in Reason, the patch does not include the actual sample - only a
reference to it. Therefore, the sample has to be stored separately, or already be on disk or in a ReFill on your
computer).

Global performance and “play” controls

Key Mode
Here you choose how Grain should respond to MIDI Note data:
• Poly
Select this if you want to play Grain polyphonically. The maximum number of voices is 12. The number of voices is
set in the Voices control at the top right of the Grain panel, see “Voices”.
• Retrig
Select this if you want to play Grain in monophonic mode and always retrigger the envelopes as soon as you play
a new note.
• Legato
The Mono Legato mode is also monophonic. However, if you play a new note without having released the previous
one, the envelopes and sample playback position won’t start over.
q Also see the description of the “Global Position” parameter. This describes how to play through a sample in a
“non-legato” fashion - or polyphonically - in a “sample playback legato” fashion, where new notes will con-
tinue at the current sample playback position (and not restart playback).

167 GRAIN SAMPLE MANIPULATOR

 Porta
Portamento makes note pitches glide from previous notes to new ones, at the time set with the Time knob. Porta-
mento can be used in all Key modes (see above).
• When On in Poly Key Mode (see above), the pitches will glide from any of the available voices.
The results will be unpredictable since there is no way of controlling from which note(s) the glide(s) will com-
mence. The effect is very nice, though.
• When On in Retrig or Legato Key Mode (see above), the pitch will glide between consecutive notes.
• In Auto mode, the pitch will glide between consecutive monophonic notes only when you play legato. If you
have selected Poly Key Mode (see above), Auto will have no effect at all.
If you release the previous key before hitting the new key, there will be no portamento effect.

Range
D Set the desired Pitch Bend range for the “Pitch” wheel with the up/down buttons, or by click-holding on the
display and dragging up/down.
Range: +/-24 semitones (+/-2 octaves) in steps of +/-1 semitone.

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Grain also responds to Pitch Bend MIDI
data from a connected MIDI master keyboard. You set the desired Pitch bend Range with the “Range” control above
the Pitch bend wheel.
! Note that with some playback algorithms, such as Spectral Grains, the audible pitch depends on the formant
rather than the pitch settings (see “Spectral Grains”). For pitch bend to have an effect here, you need to add a
Pitch Wheel -> Formant routing in the Modulation Bus, see “The Modulation Bus section”.

Mod
The Mod wheel can be used for controlling almost any parameter in Grain. Use the Mod wheel as a Source parameter
in the Modulation Bus section and then route to the desired Destination parameter(s), see “The Modulation Bus sec-
tion”.

Global output controls

Voices
Here you set the polyphony of your patch, from 1 to 12 voices.
q If you want monophonic playback you can use the “Retrig” and “Legato” modes instead of lowering the Voices
parameter to 1.

Master Volume
This is the main stereo output volume control.

168 GRAIN SAMPLE MANIPULATOR

 Panel reference
The Sample section
Here is where you load/sample and configure the audio that should serve as the base for the granular synthesis.

Sample Load section Sample Preview button

Sample Overview

Sample range markers

Position marker

Sample Start Sample End Waveform display

marker marker

Loading
D Load a sample using drag & and drop, or by clicking the Browse sample button, or by using the Up/Down but-
tons to scroll and load a sample from the currently selected folder.
! It’s possible to load stereo samples. However, the waveform will always be displayed as a mono signal, re-
gardless if it’s mono or stereo.

Setting the sample range

First you could decide how much of the original sample you want to use - and where in the sample you want to work:
D Zoom and/or scroll in the Sample Overview to define the Sample range you want to work in.
To scroll, click and drag between the orange sample range markers. To zoom, click and drag any of the sample
range markers sideways. The set Sample range is automatically updated and displayed in the waveform display.
D To work in the entire Sample range, drag the left Sample range marker all the way to the left, and the right
Sample range marker all the way to the right, in the Sample Overview.
q It’s possible to automate the sample range settings, see “Automating sample playback parameters from the
sequencer”.

169 GRAIN SAMPLE MANIPULATOR

 Setting the sample start and end
D Drag the green Sample Start marker to where you want the sample to begin playing back.
D Drag the red Sample End marker to where you want the sample playback to end.
The green triangular “flag” on the Sample Start marker shows the current playback direction. If the Sample Start
marker should be to the right of the Sample End marker, the sample will play back in the opposite direction.
! Note that the Freeze playback motion mode don’t have any Sample End Marker, see “Freeze” below.
q It’s possible to automate the sample start and end settings, see “Automating sample playback parameters
from the sequencer”.

Motion
Motion controls the way the Position marker (“playhead”) is played back in the original sample. The Motion modes
work in conjunction with the Sample Start/End markers in the waveform.
D Click the Motion selector to choose one of the following playback motion modes:
• Freeze
In Freeze mode, the sample is played back at (and around) the Sample Start marker position. There is no Sample
End marker in this mode. Note that if you have selected the Tape algorithm (see “Tape”), there will be no sound.
• One Shot
In One Shot mode, the sample is played back (from the Sample Start marker to the Sample End marker) in its en-
tirety each time you press a key.
• FW Loop
In FW Loop mode, the sample is looped forward (from the Sample End marker to the Sample Start marker) for as
long as you hold down a key.
• FW-BW Loop
In FW-BW Loop mode, the sample is looped back and forth between the Sample End marker and the Sample Start
marker for as long as you hold down a key.
• End Freeze
In End Freeze mode, the sample is played back once from the Sample Start marker to the Sample End marker and
then played back at (and around) the Sample End marker position. Note that if you have selected the Tape algo-
rithm (see “Tape”), there will be no sound after you reached the Sample End marker.
• Envelope 1
In Envelope 1 mode, the sample is played back between the Sample Start marker and the Sample End marker ac-
cording to the Envelope 1 curve (see “The Envelopes section”). The Sample Start position is represented by the
minimum Y value and the Sample End position is represented by the maximum Y value in the Envelope display.
The Envelope 1 mode is also the mode to use if you want to play back and loop the sample in sync with the Rea-
son sequencer. Use a straight ramp (up) in Envelope 1, activate Beat Sync and set the sync to a suitable bar
length, see “Looping the envelope”.

Speed
The Speed control determines how fast the play position moves in the waveform.
D Set the sample playback speed with the Speed knob.
Depending on which Motion mode and Playback Algorithm is currently selected, the sonic result may vary heavily.
If you have selected the Tape algorithm (see “Tape”), the Speed knob also affects the pitch. Note that the Speed
can be set all the way down to 0%, i.e. “stop”. Great for Tape Stop effects, for example.
! Note that the Speed control doesn’t have any effect when you use the Envelope 1 motion mode, see “Envelope
1” above.

170 GRAIN SAMPLE MANIPULATOR

 Jitter
The Jitter function modulates the sample playback position minutely and randomly. The Jitter function can be great
for generating “chorus”-like effects and to make a sound more “alive”, depending on the other settings in the sound.
D Set the playback position deviation with the Jitter knob.
At 0%, the timing and playback position is completely accurate and at 100% it is completely random.

Global Position
D Click the Global Position button to start playback of new voices from the global position, i.e. from where the
blue Position marker is currently positioned in the waveform display:

This function is great for rhythmic and vocoder-like sounds etc.

If not active, new voices will always start playing back from the Sample Start marker.

Root Key
A sample is automatically analyzed for its original pitch at the Sample Start position. The analyzed pitch is displayed
in the Analyzed display in the Root Key section. If you move the Sample Start marker, the sample is automatically re-
analyzed.
D Click the “SET” button to use the analyzed Root Key.
This will automatically place the analyzed Root Key on the correct note in the keyboard range.
D Alternatively, define the Root Key manually by dragging up/down in the “Semitone” and “Cents” boxes.

The Playback Algorithms section

Here is where you select which Playback Algorithm to use for manipulating the sample. Each of the Playback Algo-
rithms produce very different sonic results and have different unique parameters.
! Note that the common Pitch controls to the upper right could affect the sound differently, depending on the se-
lected algorithm.

Playback Algorithm selector Pitch controls

D Click the Playback Algorithm selector and choose one of the following four algorithms:

171 GRAIN SAMPLE MANIPULATOR

 Spectral Grains

The Spectral Grains playback algorithm uses FFT analysis to analyze the frequency content (partials) of the original
sample. You can then stretch the generated signal by pitch-shifting the partials, and also filter out inharmonic partials.
This way you could continuously transform inharmonic signals into harmonic signals, for example. You can also draw
your own formant curves in the spectrum display to give the sounds different pitches/characters.
• Snap
This pitch-shifts inharmonic partials towards the closest harmonic partials. At 0% the sound is almost unaffected
and at 100% the sound contains only harmonic partials.
• Filter
Instead of pitch-shifting inharmonic partials towards harmonic ones, as the Snap control above does, the Filter
control filters out the inharmonic partials and keeps the harmonic ones. Since the filter slopes are not brickwall
shaped some of the inharmonic partials (if any) will remain audible even at 100%.
• FFT Size
This sets the accuracy (and speed) of the frequency analysis. “0” is the fastest detection, but this also leaves out
detection of low frequencies. “3” is the most accurate detection. However, it’s also slower since it also detects low-
frequency material (which takes longer to detect).
• Curve
With the Curve tool you can draw your own formant curves in the frequency spectrum. Drawing above the pink
area means the partials are amplified, and drawing below the pink area means the partials are attenuated.
• Amount
Set the gain/attenuation amount of the drawn formant curve (see “Curve” above). At 0% the curve is completely
flat.
• Formant
Sets the initial pitch of the sample, together with the Root Key setting (see “Root Key”). If Snap and Filter are both
set to 0%, the Root Key and Formant controls the pitch of the signal. This also means that the Pitch parameters
(see “Pitch controls”) and Pitch wheel (see “Pitch”) have no effect. To have the Formant track the keyboard in a
musical way, make sure the Formant Kbd parameter (see below) is set to 100%.
When you raise the Snap or Filter parameters towards 100% the sound gradually adapts to the Pitch settings in-
stead, and the Root Key and Formant parameters now affect the tone color instead.
• Formant Tune
Here you can fine-tune the formant curve to adjust the pitch to the Oscillator pitch (see “The Oscillator section”).
• Formant Kbd
Here you set how much you want the formant to track the keyboard. 0% means no keyboard tracking and 100%
means full 1:1 keyboard tracking. If the Snap and Filter parameters (see above) are both set to 0%, make sure the
Formant Kbd is set to 100% to make the audible pitch track the keyboard one semitone per note.

172 GRAIN SAMPLE MANIPULATOR

 Grain Oscillator

The Grain Oscillator plays back a mix of two very short grains of the original sample. The grain playback rate corre-
sponds to the oscillator pitch. This means the original pitch (Root Key/Formant) of the sample doesn’t affect the
pitch of the sound, but the timbre.
• Pan Spread
Here you set how much you want the grains to be panned in the stereo panorama. 0% means the signal will be un-
affected and 100% means every other grain will be panned hard left and hard right. Great for nice stereo effects
and for the impression of an added stereo sub-oscillator, depending on the settings. Note that the pitch of the
panned signal becomes 1 octave lower than the original signal due to the fact that every other grain is panned.
• Pitch Jitter
Changes the pitch of every grain. The pitch modulation character is “smooth random”.
• Grain Length
Sets the lengths of the grains and also the crossfade amount. At 0% you get the shortest grains and almost no
crossfade at all. This means the sound could be a little gritty at this setting. At 100% you get longer grains, that
also overlaps each other with a smooth crossfade.
• Grain Spacing
Sets the spacing in the original sample between the two played back grains. High Spacing values render more
even sound character throughout the played notes - almost like a wavetable synth - since a lot of audio data in the
original sample is skipped. Less spacing normally creates more varying sound character between each played
note.
• Formant
Sets the formant’s initial frequency. Turn this knob to change the tone color of the sound. At high Grain Spacing
values (see above) the effect of changing the Formant could be similar to the classic “oscillator sync” sound. To
have the Formant fully track the keyboard, make sure the Formant KBD parameter (see below) is set to 100%.
• Formant Tune
Here you can fine-tune the formant curve.
• Formant Kbd
Here you set how much you want the formant to track the keyboard. 0% means no keyboard tracking and 100%
means full 1:1 keyboard tracking.

173 GRAIN SAMPLE MANIPULATOR

 Long Grains

The Long Grains playback algorithm plays back fairly long grains of the original sample. This means that it’s the orig-
inal pitch of the sound (Root Key) that affects the pitch, along with the Pitch settings (see “Pitch controls”).
The display shows the effects of the Grain Length, Rate and X-Fade settings.
• Pan Spread
Here you set how much you want every other grain to be panned in the stereo panorama. 0% means the signal will
be unaffected and 100% means every other grain will be panned hard left and hard right. Great for nice stereo ef-
fects!
• Pitch Jitter
Changes the pitch of every grain. The pitch modulation character is “smooth random”.
• Grain Length
Sets the lengths of the grains. At 0% you get the shortest grains and towards 100% you get longer grains.
• Rate
This controls the playback rate of the grains.
• X-Fade
Here you set the crossfade between the grains. At 0% there is minimal crossfade, which will give the signal a gritty
or “popping” character at the playback start and end of each grain.

Tape

The Tape playback algorithm plays back the sample the old-fashioned “tape-style” way, where playback speed and
pitch are linked. If playback speed is zero (in Freeze and End Freeze Motion modes for example), no sound will be
heard - but you can drag, modulate or automate the playback position for scrubbing and tape stop effects.
• Loop X-Fade
Sets the crossfade amount if you have selected FW Loop or FW-BW Loop as Motion type (see “Motion”)
! Note that the Loop X-Fade control has no effect if you have selected “Envelope 1” as Motion type.
! If you have selected “Envelope 1” as Motion type (see “Envelope 1”), the Speed (see “Speed”) and Pitch set-
tings (see “Pitch controls”) have no effect. The sample will play back at the same pitch regardless of which
note you play.

174 GRAIN SAMPLE MANIPULATOR

 Pitch controls

• OCT
Sets the pitch in octave steps.
Range: 5 octaves.
• SEMI
Sets the pitch in semitone steps.
Range: 12 semitones (one octave).
• TUNE
Changes the pitch in steps of 1 cent.
Range: +/- 50 cents (down or up half a semitone).
• KBD
Sets how much the pitch should track incoming MIDI Notes.
Range: 0% (no tracking (constant pitch)) to 100% (1 semitone per key).
! In the Spectral Grains playback algorithm (see “Spectral Grains”), the Pitch controls have no effect if Snap and
Filter are set to 0%. To get full effect of the Pitch controls, set Snap or Filter to 100%.

175 GRAIN SAMPLE MANIPULATOR

 The Oscillator section

The Oscillator can be used in addition to the sample playback. The oscillator features a number of selectable wave-
forms and a modulation control, which affect the signals differently depending on selected waveform. The oscillator
pitch always tracks the keyboard to 100%. This makes it perfect as a pitch reference for the sample signal.

On/Off
D Click the On/Off LED button to switch on/off the oscillator.

Oct
D Turn the OCT knob to change the pitch in octave steps.
Range: 5 octaves.

Waveform and Mod

D Select the desired waveform by dragging up/down in the waveform display, or by clicking the up/down but-
tons.
The waveforms are:
• Sine
A pure sinewave at Mod=0%, gradually transformed towards a sawtooth type signal at Mod=100%.
• Triangle
A triangle wave at Mod=0%, gradually transformed towards a sawtooth type signal at Mod=100%.
• Sawtooth
A lowpass-filtered sawtooth wave at Mod=0%, gradually transformed to a pure sawtooth signal at Mod=100%.
• Square/Pulse
A 50% duty cycle square wave at Mod=0%, gradually pulsewidth-modulated to a 0% duty cycle pulse wave
(silence) at Mod=100%.
q Modulate the Mod parameter from an LFO (using the Modulation Bus) to achieve PWM.
• Noise
A level-compensated lowpass-filtered noise at Mod=0%, gradually transformed to white noise at Mod=100%.
• Band Noise
A noise-modulated sinewave. The Mod knob controls noise bandwidth. At Mod=100%, the oscillator produces
pure noise. Turning the knob counter-clockwise towards 0% gradually narrows the noise bandwidth until a slightly
modulated sinewave is produced.

176 GRAIN SAMPLE MANIPULATOR

 The Filter section

The signals from the Playback Algorithms section and the Oscillator section can be individually mixed and routed
through the Filter section. The Filter section features four different filter types.

Routing buttons
D Click the red buttons with a triangle pointing to the right, to route the desired signal to the Filter section.
To bypass the signals from the Filter section, click the buttons with the triangle pointing upwards or downwards.

Filter type
D Click the Filter type selector to choose any of the following filter types:
• HP 12dB
A highpass filter with a 12dB/octave slope.
• BP 12dB
A bandpass filter with 12dB/octave slopes.
• LP 12dB
A lowpass filter with a 12dB/octave slope.
• LP Ladder 24dB
A ladder-type lowpass filter with a 24dB/octave slope.

Freq
D Set the cutoff frequency (for the HP and LP filter types) or the center frequency (for the BP filter type).

Reso
D Set the resonance amount.

Env 2
D Set the cutoff/center frequency modulation amount from the Envelope 2.
Since this is a “hardwired” connection from Envelope 2 you don’t need to use the Modulation Bus for envelope
modulating the cutoff/center frequency.

Vel
D If you want the Envelope 2 amount to be controlled from keyboard velocity, turn up the Vel knob.

Kbd
D Set how much you want the filter cutoff/center frequency to track the keyboard.
At 0%, the filter frequency is static regardless where on the keyboard you play. At 100% the filter tracks the key-
board 1:1, i.e. one semitone per note.

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 The Amplifier section

The Amplifier section contains a standard ADSR envelope which controls the amplitude of the signals from the Play-
back Algorithms and Oscillator sections equally. The picture below shows the various stages of the ADSR envelope:

Level

Gain
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value
set with the Gain knob (see below). How long this should take, depends on the Attack setting. If the Attack is set to
“0”, the Gain level is reached instantly. If the Attack value is raised, it will take longer time before the Gain level is
reached.

D(ecay)
After the Gain level has been reached, the level starts to drop. How long this should take is governed by the Decay
parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to “0”,
the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the volume
gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to drop to
some other value than zero, you raise the Sustain parameter.

S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain to
full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain param-
eter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays
there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).

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 But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Gain level, then grad-
ually decreases to finally land to rest on a level somewhere in-between zero and the Gain level. Note that Sustain rep-
resents a level, whereas the other envelope parameters represent times.

R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume to
drop back to zero after you release the key.

Gain
D Set the desired maximum level for the Amplifier with the Gain knob.
This is the maximum level the envelope will reach after the Attack stage is completed (see above).
q If you want to create a tremolo effect, assign “Gain” as Destination and an LFO as Source in the Modulation
Bus section, see “The Modulation Bus section”.

Vel
D If you want the Gain level to be controlled from keyboard velocity, turn up the Vel knob.

Pan
D Set the panning of the output signal from the Amplifier in the stereo panorama.
q Since Pan works individually per voice, you can assign e.g. Keyboard Velocity or an Envelope in the Modula-
tion Bus to control the Pan effect, see “The Modulation Bus section”.

The Envelopes section

The Envelopes section features four separate polyphonic (one per voice) general purpose envelope generators, that
can be assigned to control selectable parameter(s) in the Modulation Bus section. The first two envelopes (Envelope
1 and Envelope 2) are also hardwired to the Motion and Filter Frequency destinations respectively.
The Envelopes are extremely flexible, and you can draw your own custom modulation shapes by clicking and drawing
in the display area. There are also a number of preset shapes that you can use as starting points (or use as is). If you
use Loop mode, you could turn the envelope into a kind of LFO.
See “The Modulation Bus section” for details on how to assign the Envelopes to the desired destination(s).

Envelope 1, 2, 3 and 4
D Click one of the Envelope 1, 2, 3 or 4 buttons to select which envelope to edit:

Envelope 1 and Envelope 2 are also hardwired to the Motion and Filter Frequency destinations respectively.

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 Preset
1. Click the Preset button to bring up a palette of envelope preset curves:

2. Click the desired envelope preset curve to place it on the display.

Let’s select a standard ADSR style of envelope curve:

Adding a Sustain stage

D Click the Sustain button to add a sustain stage to the envelope:

The vertical blue marker that appears indicates where the envelope will stay sustained until you release the key.
D Drag the sustain marker sideways to move the sustain stage to the desired position:

D To remove the sustain stage, click the Sustain button.

180 GRAIN SAMPLE MANIPULATOR

 Adding and removing envelope points
D Double click, or hold down [Ctrl](win) or [Cmd](Mac) and click, in the envelope display to add points to the en-
velope curve:

D To remove a point, double click, or hold down [Ctrl](win) or [Cmd](Mac) and click, on an existing point on the
envelope curve.

Changing the envelope curve shape

D Click a line segment (between two points) and drag up/down to change the curve shape:

Looping the envelope

D Click the Loop button to turn the envelope into a kind of LFO.
If there was previously a sustain stage in the envelope, this will automatically be disabled when you click the Loop
button.

Here we have edited a stepped curve from the Presets. We have also enabled Beat Sync and set the length/rate
to 4/4. This means that each step in the curve now represents an 1/8th note.
• Key Trig means the envelope restarts when you play a note.
• You can choose whether the envelope should send out a bipolar value or unipolar one (0-100%).
• If Global is on, the envelope will be common for all voices.
Another useful application for looped envelopes is to sync the sample playback to the Reason sequencer when using
the Envelope 1 Motion mode (see “Envelope 1”):
1. Select Envelope 1 (since it is hardwired to the sample playback Motion parameter).
2. Select the “Ramp Up” Preset, enable Loop and set the Beat Sync to the desired value:

Playing back Reason’s sequencer now plays back the sample synced to the sequencer Tempo.

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 Editing levels only
D To restrict the editing to levels only, without affecting the time positions, click the Edit Y-Pos button:

In this mode you cannot change the time positions of the envelope points, only their levels (height). This is extra
useful with a stepped Preset curve, because dragging up or down will change the value of an entire segment, turn-
ing the Envelope into a pseudo-sequencer.
! To be able to adjust the level of a segment, the two points on either side of the segment have to be on the ex-
act same time positions. Otherwise, only the closest point will be changed. Also, any inclining/declining seg-
ment will automatically turn horizontal when edited:

Adjusting the level of a segment.

Creating “free form” envelope curves

As a special feature in the Edit Y-Pos mode, you can also draw “free form” curves:
D To continuously add new consecutive points, hold down [Ctrl](win) or [Cmd](Mac) and drag in the envelope
display:

D To erase points, hold down [Shift] and [Ctrl](win) or [Cmd](Mac) and drag in the envelope display.

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 The LFO section

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO
modulate the pitch of a signal to produce vibrato, but there are countless other applications for LFOs.
The LFO section features three separate general purpose LFOs, that can be assigned to control selectable parame-
ter(s) in the Modulation Bus section.
D Select which of the three LFOs you want to edit by clicking one of the LFO 1, LFO 2 and LFO 3 buttons.
D Select an LFO waveform by clicking the spin controls to the right of the waveform display, or by click-holding
in the display and dragging up or down.
Besides the standard waveforms (sine, triangle, pulse, etc.) there are random, slope and stepped waveforms. The
shape of the waveforms are shown in the display.
D Set the LFO frequency with the Rate knob.
D Click the Beat Sync button to sync the LFO to the main sequencer tempo.
The Rate parameter now controls the time divisions.
D Click the Key Sync button to restart the LFO at every new Note On.
D Click the Global button to make the LFO common for all voices (monophonic).
D Turn the Delay knob to introduce a delay before the LFO modulation kicks in after a note is played.
Turn clockwise for longer delay times.

The Effects section

The Effects section features six different effect modules that can be freely reordered by dragging & dropping. Most
of the effect parameters are also available as destinations in the Modulation Bus, see “The Modulation Bus section”.
At the top of the Effects section are six Effect buttons. Click any of these to bring up the control panel for the corre-
sponding effect. Below the Effect buttons are the On/Off buttons for the individual effects. Click these to activate the
effects.

Reordering the effects

D To define the order of the effects in the serial connection, click and hold on the desired Effect button and drag
to the desired position:

Moving the Chorus effect to another position in the effects chain.

You can reorder the effects at any time.

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 Phaser/Flanger/Chorus

This is a stereo Phaser/Flanger/Chorus.

D Select effect type with the Phaser/Flanger/Chorus switch.
The selected effect type is displayed on the Effect button.
• Depth
Sets the depth of the selected effect. To get a static sound, set Depth to zero.
• Rate
Sets the rate/speed of the modulation.
• Spread
Sets the stereo width of the effect.
• Amount
Sets the Dry/Wet amount of the effect.
q It’s also possible to modulate the start/center frequency of the Phaser/Flanger/Chorus using the “Chorus/
Flanger/Phaser > Frequency” destination in the Modulation Bus, see “The Modulation Bus section”.

Distortion

The Distortion effect features six different types of distortion.

D Select distortion type with the switch.
“Dist” produces a dense, rich analog type of distortion.
“Scream” produces a less bright type of distortion.
“Tube” emulates a tube type of distortion.
“Sine” is a sine shaping distortion.
“S/H” gives the effect of sample rate reduction.
“Ring” is a ring modulator effect.
• Drive
Sets the overdrive/feedback level of the selected distortion.
• Tone
This is a lowpass filter and sets the tone of the selected distortion.
• Amount
Sets the Dry/Wet amount of the distortion.

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 EQ

The EQ effect is a single band parametric equalizer with adjustable Q-value and Gain.
• Freq
Sets the center frequency of the EQ band.
• Q
Sets the bandwidth of the EQ band, from wide to narrow.
• Gain
Sets the gain/attenuation of the EQ band, from -18dB to +18dB.

Delay

This is a stereo delay, routed as a send effect.

• Sync
Activate Sync to sync the delay time to the main sequencer tempo.
• Time
This sets the time between the delay repeats. If Sync is active (see above), the Time parameter now controls the
time divisions.
• Ping Pong
Activate Ping Pong to have the delay repeats alternating between left and right in the stereo panorama. The effect
is also dependent on the Pan parameter (see below).
• Pan
Sets the panning of the delay repeats in the stereo panorama. If Ping Pong is active (see above) the Pan knob
controls the panning of the initial delay repeat as well as the total stereo spread of the remaining repeats.
• FB
The FB (feedback) parameter determines the number of delay repeats.
• Amount
Use this parameter to adjust the send level to the Delay effect.
q If you play a note, have a long delay feedback and turn down Amount, the echoes will continue. This allows for
automated “triggered delay” fx.

185 GRAIN SAMPLE MANIPULATOR

 Compressor

This is a stereo compressor.

• Attack
This governs how quickly the compressor will apply its effect when signals rise above the set threshold. If you raise
this value, the response will be slower, allowing more of the signal to pass through the compressor unaffected.
Typically, this is used for preserving the attacks of the sounds.
• Release
When the signal level drops below the set threshold, this determines how long it takes before the compressor lets
the sound through unaffected. Set this to short values for intense, “pumping” compressor effects, or to longer val-
ues for a smoother change of the dynamics.
• Thres
This is the threshold level above which the compression sets in. Signals with levels above the threshold will be af-
fected, signals below it will not. In practice, this means that the lower the Threshold setting, the more the compres-
sion effect.
• Ratio
This specifies the amount of gain reduction applied to the signals above the set threshold.

Reverb

This is a stereo reverb, routed as a send effect.

• Decay
This governs the length of the reverb effect.
• Size
Sets the emulated room size, from small room to large hall. Middle position is the default room size.
Lowering this parameter results in a closer and gradually more “canned” sound. Raising the parameter results in a
more spacey sound, with longer pre-delay.
• Damp
Raising the Damp value cuts off the high frequencies of the reverb, thereby creating a smoother, warmer effect.
• Amount
Use this parameter to adjust the send level to the Reverb effect.
If you play a note, have a long delay Decay time and turn down Amount, the reverberation will continue.

186 GRAIN SAMPLE MANIPULATOR

 The Modulation Bus section

The Modulation Bus section is used for routing a modulation Source to one or two modulation Destinations each.
This creates a very flexible routing system that complements the pre-wired routing in Grain.
The Modulation Bus section in Grain is derived from the one in the Reason Thor Polysonic Synthesizer device, so if
you are familiar with Thor, you will quickly find your way around in Grain’s modulation bus.
There are eight “Source –> Destination 1 –> Destination 2 –> Scale” busses, of which the first four have pre-as-
signed sources. However, these four pre-assigned sources can be easily changed if you like.
A Source parameter can modulate two different Destination parameters per bus (with variable Amount settings).
Each bus also has a Scale parameter that affects the relative modulation Amount for both Destinations.
q Note that it is possible to assign the same source parameter as Source in several busses. This allows you to
control more than two Destination parameters from the same Source.

1. Select the desired Source parameter by clicking in the corresponding Source box and selecting from the list.
The following parameters can be used as modulation Sources:

| Parameter | Description
Velocity This applies modulation according to the Keyboard Velocity values (how hard or soft you strike the MIDI
keyboard keys).
LFOs (LFO 1, LFO 2 and LFO 3) This allows you to modulate parameters from LFO 1, LFO 2 and LFO 3 respectively.
Envelopes (Amp Envelope, Envelope 1, This allows you to modulate parameters from any of the Envelopes.
Envelope 2, Envelope 3, Envelope 4, As a special feature you can also modulate parameters from the multiplied signal of Envelope 3 and
Envelope 3 * Envelope 4, Envelope 3 * Envelope 4, as well as from the multiplied signal of Envelope 3 and LFO 3.
LFO 3)
Mod Wheel This allows you to modulate parameters from the Mod Wheel.
MW Latched This allows you to modulate parameters based on the current Mod Wheel value at a given Note On.
Pitch Wheel This allows you to modulate parameters from the Pitch Bend control.
Breath This allows you to modulate parameters from the Breath performance controller
Expression This allows you to modulate parameters from the Expression performance controller
Aftertouch This allows you to modulate parameters from Keyboard Aftertouch (channel aftertouch)
Sustain This allows you to modulate parameters from a connected sustain pedal. Note that continuous sustain
data (0-127) is supported - not just on/off.
Key This is keyboard tracking. If a positive Amount value is used and the destination is filter frequency, the
filter frequency will track the keyboard, i.e. increase with higher notes.
Random This sends out a random value each time a new note is played.
Key In Octave This allows you to modulate parameters based on 12 separate note values (within each octave).
Noise This allows you to modulate parameters from white noise.
Polyphony This allows you to modulate parameters based on the number of playing voices at a given time.
Last Velocity This applies modulation according to the latest Keyboard Velocity value (how hard or soft you hit the
latest MIDI keyboard key).
Sample Pitch Curve As soon as you load a sample in Grain the pitches throughout the entire sample is automatically analyzed
and saved as a “Pitch Curve”. This allows you to modulate parameters based on the analyzed pitch value
at the Position marker’s current position in the original sample.

187 GRAIN SAMPLE MANIPULATOR

 | Parameter | Description
Display Y Position This allows you to modulate parameters based on the mouse pointer’s Y position in the sample window.
See “Automating sample playback parameters from the sequencer” for an example.
Display Mouse Gate This allows you to modulate parameters based on the clicking/holding the mouse in the sample window.
See “Automating sample playback parameters from the sequencer” for an example.
CV Input 1/2/3/4 This takes the current value on the CV 1/CV 2/CV 3/CV 4 inputs on the rear panel and sends to the
desired destination.
CV Input 1/2/3/4 Latched This allows you to modulate parameters based on the current CV 1/CV 2/CV 3/CV 4 value at a given
Note On.

Modulation Bus Source parameters.

2. Set the Amount for the first Destination by turning the corresponding Amount knob, or by clicking and drag-
ging vertically in the corresponding Amount box.
! Note that the Amount range is +/-100. This means that the Amount value can exceed the modulated parame-
ter’s range. When this happens, the modulated parameter simply stays at its extreme value until the Amount
value gets within the parameter’s range again.
3. Select the first Destination parameter by click-holding the grey arrow symbol to the right of the corresponding
Destination box.
The arrow symbol turns blue.
4. While click-holding, drag to the desired destination parameter on the panel:

Assigning LFO 1 Rate as a Destination for Envelope 1.

As you hover over a valid destination control on the panel, the parameter name is automatically displayed in the
Destination box in the Modulation Bus.
5. To assign the currently selected Destination control, release the mouse button.
! Dragging to the Waveform Display (see “The Sample section”) will always assign the playback Position
parameter. To assign the sample Start Position or End Position, select the parameter from the list (see below).
D Alternatively, click the desired Destination box and select the Destination parameter from the list.
The following parameters can be used as modulation Destinations:

| Parameter | Description
Position This affects the sample “playhead” position in the original sample.
Speed This affects the Speed control in the sample window.
Jitter This affects the Jitter control in the sample window.
Start Position This affects the Sample Start marker position in the original sample.
End Position This affects the Sample End marker position in the original sample.
Pitch This affects the pitch of the original sample.
Octave This affects the Oct control in the Playback Algorithm section.
Formant This affects the Formant Form control in the Playback Algorithm section (if applicable).

188 GRAIN SAMPLE MANIPULATOR

 | Parameter | Description
Sample Level This affects the Sample Level control of the Playback Algorithm section.
Grain: Length This affects the Grain Length parameter on the Grain Oscillator and Long Grains playback algorithms.
Grain: Rate/Spacing This affects the Grain Spacing parameter on the Grain Oscillator and the Rate parameter on the Long
Grains playback algorithms.
Grain: X-Fade This affects the X-Fade parameter on the Long Grains playback algorithm.
Grain: Pan Spread This affects the Pan Spread parameter on the Grain Oscillator and Long Grains playback algorithms
Grain: Pitch Jitter This affects the Pitch Jitter parameter on the Grain Oscillator and Long Grains playback algorithms
Spectral Grain: Harm Snap This affects the Snap parameter on the Spectral Grains playback algorithm.
Spectral Grain: Harm Filter This affects the Filter parameter on the Spectral Grains playback algorithm.
Spectral Grain: Curve Amount This affects the Amount parameter on the Spectral Grains playback algorithm.
Oscillator: Modulation This affects the Mod parameter on the Oscillator.
Oscillator: Octave This affects the Oct parameter on the Oscillator.
Oscillator: Pitch This affects the (full range) pitch of the Oscillator.
Oscillator: Level This affects the Osc Level parameter of the Oscillator section.
Filter: Freq This affects the Frequency parameter in the Filter section.
Filter: Reso This affects the Resonance parameter in the Filter section.
Amplifier: Gain This affects the Gain parameter of the Amplifier section.
Amplifier: Pan This affects the Pan parameter of the Amplifier section.
Amp Envelope: Attack This affects the Attack time of the Envelope in the Amplifier section.
Amp Envelope: Decay This affects the Decay time of the Envelope in the Amplifier section.
Amp Envelope: Sustain This affects the Sustain level of the Envelope in the Amplifier section.
Amp Envelope: Release This affects the Release time of the Envelope in the Amplifier section.
LFO’s: LFO1/2/3 Delay This affects the LFO1/2/3 Delay parameter.
LFO’s: LFO1/2/3 Rate This affects the LFO1/2/3 Rate parameter.
Envelopes: Env1/2/3/4 Rate This affects the Envelope 1/2/3/4 Rate parameter.
Portamento This affects the Portamento Time parameter.
CV Outputs: CV1/2/3/4 Out This sends out the source modulation value(s) on the CV1/2/3/4 Output on the rear panel.
Reverb: Decay This affects the Decay parameter in the Reverb effect.
Reverb: Amount This affects the Amount parameter in the Reverb effect.
Delay: Time This affects the Time parameter in the Delay effect.
Delay: Feedback This affects the FB parameter in the Delay effect.
Delay: Amount This affects the Amount parameter in the Delay effect.
Delay: Pan This affects the Pan parameter in the Delay effect.
Dist: Drive This affects the Drive parameter in the Dist effect.
Dist: Tone This affects the Tone parameter in the Dist effect.
Dist: Amount This affects the Amount parameter in the Dist effect.
Compressor: Release This affects the Release parameter in the Compressor effect.
Compressor: Ratio This affects the Ratio parameter in the Compressor effect.
Chorus/Flanger/Phaser: Frequency This affects the center frequency of the Chorus/Flanger/Phaser effects.
Chorus/Flanger/Phaser: Amount This affects the Amount parameter of the Chorus/Flanger/Phaser effects.
Par EQ: Frequency This affects the Freq parameter in the EQ effect.
Par EQ: Gain This affects the Gain parameter in the EQ effect.
Note Trig This is mainly intended for use with the Display > Mouse Gate or Display > Y-Position source parame-
ters, e.g. with the Freeze motion mode, to trig/gate the sample without having to play the keyboard. This
way you can play back the Root Key of the sample by just clicking/dragging in the Waveform display.

Modulation Bus Destination parameters.

189 GRAIN SAMPLE MANIPULATOR

 6. Set the Amount for the second Destination (if desired) by turning the corresponding Amount knob, or by click-
ing and dragging vertically in the Amount box for the second destination.
7. If desired, select a second Destination parameter by click-holding the blue arrow symbol to the right of the cor-
responding Destination box, and dragging to the desired control on the panel.
8. If desired, click the Scale box and select a Scale parameter.
The available Scale parameters are the same as the Source parameters, see “Modulation Bus Source parame-
ters.”.
9. Turn the Scale Amount knob, or click the Amount box to the left of the Scale box and move the mouse pointer
up or down to set a Scale Amount value.
Both positive and negative Scale Amount values can be set (+/- 100%). If you, for example, are using the Mod
Wheel as Scale parameter and don’t want any modulation when the Mod Wheel is set to zero, set the Scale
Amount parameter to 100%. Then, there will be no effect when the Mod wheel is set to zero, and full modulation
when the Mod Wheel is all the way up.
• How much modulation will be applied when the Scale parameter is set to maximum is governed by the to Des-
tination Amount parameter(s).
• How much the Scale parameter controls the modulation is set with the Scale Amount parameter.
D To clear an assigned Source, Destination or Scale parameter, hold down [Ctrl](Win) or [Cmd](Mac) and click
the Source/Destination/Scale box. Alternatively, click the Source/Destination/Scale box and select “Off” from
the list.
D To reset an Amount value to 0, hold down [Ctrl](Win) or [Cmd](Mac) and click the desired Amount box or knob.
D To clear an entire modulation assignment (a whole row), click the circular X button to the right of the corre-
sponding Scale box.

Modulation example patches

The factory patches for Grain features a folder named “Templates”. In this folder you will find a number of patches
that show typical modulation examples, to make it easier to get the hang of how to create your own patches.

190 GRAIN SAMPLE MANIPULATOR

 Connections

! Remember that CV connections are NOT stored in the Grain patches! If you want to store CV connections
between devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Grain from another CV/Gate device (typically a Matrix
or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/
off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

CV Modulation inputs and outputs

These assignable control voltage (CV) inputs and outputs can be used for modulation of and from assigned Source
and Destination parameters in the Modulation Bus section, see “The Modulation Bus section”.

Audio Output
These are the main audio outputs. When you create a new Grain device, these outputs are auto-routed to the first
available outputs in the I/O device.

191 GRAIN SAMPLE MANIPULATOR

 Tips and Tricks
Automating sample playback parameters from the sequencer
Besides the extensive modulation capabilities of the Modulation Bus, the sample playback parameters can also be
automated in the main sequencer. For example, you could automate the Sample Start and/or Sample End markers in
the Sample section to have the markers reposition in real-time during playback of the main sequencer. Below is a ba-
sic example of how you could automate various sample playback parameters:
1. Record some notes on the Grain sequencer track in the main sequencer and then hit Stop twice.
2. Hit Record again in the main sequencer and drag the Sample Start and Sample End markers during recording:

3. Hit Stop in the sequencer twice when you are done recording.
If you moved both the Sample Start marker and the Sample End marker you will now have four parameter automa-
tion lanes/tracks in the sequencer.
• The Display Y lane/track and the Display Gate lane/track always appear as soon as you have recorded any
marker movements in the Sample section display.
• The Display Y lane/track represents the vertical movements you made with the mouse during the recording.
This automation doesn’t affect the sample playback in any way but can instead be used as a modulation source in
the Modulation Bus, for modulating the desired destination parameter(s).
• The Display Gate lane/track reflects when you clicked (and held) the mouse button during the recording.
This automation doesn’t affect the sample playback either but can be used as a modulation source in the Modula-
tion Bus, for modulating/gating the desired destination parameter(s).
• The Position and End Pos lanes/track represent the movements of the Sample Start and Sample End markers
respectively.
D If you like you could also record automation of the Sample Range Zoom and Scroll parameters by dragging the
markers in the Sample Overview area:

After recording the movements of the Sample Range markers, two new Parameter Automation lanes/tracks ap-
pear:
• The Scroll lane/track represents the movement of the leftmost Sample Range marker, and thus the movement
of the entire sample range.
• The Zoom lane/track represents the movement of the rightmost Sample Range marker, and thus the zooming
(in or/and out) of the sample range.
q It’s also possible to automate the Motion, Speed, Jitter and Global Position parameters.
q Also, don’t forget to check out the modulation example patches, see “Modulation example patches”.

192 GRAIN SAMPLE MANIPULATOR

Chapter 13
Humana
Vocal Ensemble
 Introduction

Humana Vocal Ensemble features a great selection of male and female vocal samples - perfect for any music style.
The multi-sampled vocal sound sets can also be tailored and processed in the high-quality filter, amp, delay and re-
verb sections.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel overview
The Humana front panel contains the following sections:

1 2

3 4 5 6 7 8

The Humana front panel sections.

• 1. Patch Selector (for browsing, loading and saving patches).
• 2. Master Volume.
• 3. Global performance and “play” controls.
• 4. Instruments section.
• 5. Filter section.
• 6. Amp Envelope section.
• 7. Delay effect section.
• 8. Reverb effect section.

194 HUMANA VOCAL ENSEMBLE

 Using Humana
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Global performance and “play” controls

Note
The Note LED lights up each time Humana receives a MIDI Note On.

Range
D Set the desired Pitch Bend range for the “Pitch” wheel with the up/down buttons, or by click-holding on the
display and dragging up/down.
Range: +/-12 semitones (+/-1 octave) in steps of +/-1 semitone.

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Humana also responds to Pitch Bend MIDI
data from a connected MIDI master keyboard. You set the desired Pitch bend Range with the “Range” control to the
right of the Mod wheel.

Mod
The Mod wheel can be used for controlling the three predefined parameters to the right of the Mod wheel:
• S. Start
Here you set how/if the Mod wheel should affect the Sample Start position of the currently selected instrument.
The parameter is bi-polar, with zero modulation at the 12 o’clock position. A negative value moves the sample start
back and a positive value moves it forward.
! Note that if the S.Start parameter in the Instruments section (see “S. Start (Sample Start)”) is at 0 ms, the sam-
ple start cannot be moved back any further. Similarly, if the S.Start parameter in the Instruments section is at
150 ms, the sample start cannot be moved forward any further.
• F. Freq
Here you set how/if the Mod wheel should affect the Filter Cutoff parameter. The parameter is bi-polar, with zero
modulation at the 12 o’clock position.

195 HUMANA VOCAL ENSEMBLE

 ! Note that if the Cutoff parameter in the Filter section (see “Cutoff”) is at 20 Hz, the frequency cannot be low-
ered any further. Similarly, if the Cutoff parameter in the Filter section is at 25 kHz, the frequency cannot be
raised any further.
• Level
Here you set how/if the Mod wheel should affect the Master Volume parameter. The parameter is bi-polar, with
zero modulation at the 12 o’clock position.
! The volume can never be modulated louder than the maximum +12 dB limit of the Master Volume control.

Panel controls
The Instruments section

Instrument selector

D Click the Instrument name selector to bring up a menu of the available instruments - and then select the de-
sired instrument from the menu.
Alternatively, click and drag up/down in the display above the selector to scroll through the instruments.
! Depending on the instrument size (in MB), it could take a short moment before the entire instrument is loaded
into RAM.
! Also note that the note ranges of the instruments extend outside their “natural” ranges, which could produce
nice artificial effects.
The available instruments are:

196 HUMANA VOCAL ENSEMBLE

 • Mars ah

A male vocal ensemble singing sustained “Ah’s” (forte). Recorded with stage mics (stereo) in a large hall, wet. Con-
ducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Mars oo
A male vocal ensemble singing sustained “Oo’s” (piano). Recorded with stage mics (stereo) in a large hall, wet.
Conducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Mars ah Staccato
A male vocal ensemble singing staccato “Ah’s” (forte). Recorded with stage mics (stereo) in a large hall, wet. Con-
ducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Mars oo Staccato
A male vocal ensemble singing staccato “Oo’s” (piano). Recorded with stage mics (stereo) in a large hall, wet. Con-
ducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Venus ah

A female vocal ensemble singing sustained “Ah’s” (forte). Recorded with stage mics (stereo) in a large hall, wet.
Conducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Venus oo
A female vocal ensemble singing sustained “Oo’s” (piano). Recorded with stage mics (stereo) in a large hall, wet.
Conducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Venus ah Staccato
A female vocal ensemble singing staccato “Ah’s” (forte). Recorded with stage mics (stereo) in a large hall, wet.
Conducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Venus oo Staccato
A female vocal ensemble singing staccato “Oo’s” (piano). Recorded with stage mics (stereo) in a large hall, wet.
Conducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.

197 HUMANA VOCAL ENSEMBLE

 • Mercury ah

A boys’ choir singing sustained “Ah’s” (forte). Recorded with stage mics (stereo) in a large hall, wet. Conducted by
Robert Geary. The samples are from Soundiron's Mercury Symphonic Boys’ Choir.
• Female Soprano ah

Female soprano Nichole Dechaine singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio,
dry. The samples are from Soundiron's Voices Of Rapture.

198 HUMANA VOCAL ENSEMBLE

 • Female Alto ah

Female alto Kindra Scharich singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio, dry. The
samples are from Soundiron's Voices Of Rapture.
• Male Tenor ah

Male tenor Brian Thorsett singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio, dry. The
samples are from Soundiron's Voices Of Rapture.

199 HUMANA VOCAL ENSEMBLE

 • Male Bass ah

Male bass Joseph Trumbo singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio, dry. The
samples are from Soundiron's Voices Of Rapture.
• Female ah

Female alto soloist Francesca Genco singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio,
dry. The samples are from Soundiron's Voices Of Gaia.

200 HUMANA VOCAL ENSEMBLE

 • Female ah 2

Female alto soloist Linda Strawberry singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio,
dry. The samples are from Soundiron's Voices Of Gaia.
• Male ah

Male tenor soloist Brian Lane singing sustained “Ah’s” (forte). Recorded with close mic (mono) in a studio, dry. The
samples are from Soundiron's Voices Of Gaia.

201 HUMANA VOCAL ENSEMBLE

 S. Start (Sample Start)

D Turn the S.Start knob to set where in the sample the playback should start.
Note that the effect could be different depending on the selected instrument.

Oct

D Set the octave transposition for the instrument.

Range: 5 (+/-2) octaves.
! Note that the note ranges of the instruments can be transposed far outside their “natural” ranges, which could
produce nice artificial effects.

Semi

D Set the pitch in semitone steps.

Range: +/-12 semitones (two octaves).

Fine

D Set the pitch in steps of 1 cent.

Range: +/- 50 cents (down or up half a semitone).

The Filter section

202 HUMANA VOCAL ENSEMBLE

 Filter On/Off

D Click the On/Off LED button to switch on/off the Filter section.

Filter Type selector

D Click and drag up/down on the Filter Type selector to select one of the available filter types - or step through
the filter types by clicking the Up/Down arrow buttons.
The available filter types are:
• LP
This is a lowpass filter with 12db/octave slope.
• HP
This is a highpass filter with 12db/octave slope.
• BP
This is a bandpass filter with 6db/octave slopes.
• Comb
This is a comb filter for phaser/flanger type of effects.

Cutoff

D Set the cutoff/center frequency with the Cutoff knob.

The cutoff parameter sets where in the frequency range you want the resonance and attenuation to appear.
Range: 20 Hz to 25 kHz.

Reso

D Set the resonance amount with the Reso knob.

The resonance parameter amplifies the frequencies at, and around the cutoff/center frequency.

Env

D With the Env knob you set how much you want the Filter Envelope (see below) to affect the Cutoff frequency.
Range: 0% to 100%.

203 HUMANA VOCAL ENSEMBLE

 Filter Envelope

The standard ADSR type envelope controls the filter cutoff frequency modulation over time. The ADSR envelope
characteristics are described in detail in “Amp Envelope”.

Vel

D Turn the Vel knob to set how much the cutoff/center frequency should be modulated by Keyboard Velocity.
Range: 0% to 100%.

Kbd

D Turn the KBD (Keyboard Track) knob to set how much the cutoff/center frequency should track incoming MIDI
Notes.
Range: 0% (no tracking (constant frequency)) to 100% (1 semitone per key).

The Amp section

Vel

D Turn the Vel knob to set how much the amplitude should be modulated by Keyboard Velocity.
Range: 0% to 100%.

204 HUMANA VOCAL ENSEMBLE

 Amp Envelope

The Amp Envelope is a standard ADSR envelope which controls the amplitude of the signal over time. The picture
below shows the various stages of the ADSR envelope:

Level

Volume
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

• A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value
set with the Master Volume knob. How long this should take, depends on the Attack setting. If the Attack is set to
“0”, the Volume value is reached instantly. If the Attack value is raised, it will take longer time before the Master
Volume value is reached.
• D(ecay)
After the Master Volume value has been reached, the level starts to drop. How long this should take is governed
by the Decay parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to
“0”, the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the
volume gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to
drop to some other value than zero, you raise the Sustain parameter.
• S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain
to full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain pa-
rameter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and
stays there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Volume value, then
gradually decreases to finally land to rest on a level somewhere in-between zero and the Master Volume value.
Note that Sustain represents a level, whereas the other envelope parameters represent times.

205 HUMANA VOCAL ENSEMBLE

 • R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume
to drop back to zero after you release the key.

The Delay section

This is a stereo delay which affects all voices globally.

Delay On/Off

D Click the On/Off LED button to switch on/off the Delay section.

Time

D Set the time between the delay repeats.

If Sync is active (see below), the Time parameter controls the time divisions.

Feedback

D Set the number of delay repeats with the Feedback knob.

Sync

D Click the Sync button to sync the delay time to the main sequencer Tempo.
When active, the Time knob (see above) controls the time divisions.

206 HUMANA VOCAL ENSEMBLE

 Ping Pong

D Activate this to get the delay repeats alternating between the left and right channels.
Note that this also doubles the delay tempo.

Damp

D Raise the Damp value to gradually cut off the high frequencies of the delay repeats.

Amount

D Set the delay amount with the Amount knob.

Note that the Delay effect is routed as a “send effect”, with the Amount knob working as an “effect return” control.
This means that if you play a short note with the Amount knob set to zero, you might still get a delay effect if you
turn up the Amount afterwards (depending on the Feedback setting, see “Feedback”).

The Reverb section

This is a stereo reverb which affects all voices globally.

Reverb On/Off

D Click the On/Off LED button to switch on/off the Reverb section.

207 HUMANA VOCAL ENSEMBLE

 Time

D Set the reverberation duration time.

In practice this sets the “size” of the reverberation “chamber/room/hall”.

Pre-Delay

D Set the pre-delay time of the reverb.

Range: 0-200 ms.

Hi Damp

D Raise the Hi Damp value to cut off the high frequencies of the reverb and thereby create a smoother, warmer
effect.

Lo Damp

D Raise the Lo Damp value to cut off the low frequencies of the reverb signal, to make the reverb effect less
“muddy”.

Amount

D Set the reverb amount with the Amount knob.

Note that the Reverb effect is routed as a “send effect”, with the Amount knob working as an “effect return” con-
trol. This means that if you play a short note with the Amount knob set to zero, you might still get a reverb effect if
you turn up the Amount afterwards (depending on the Time setting, see “Time”).

208 HUMANA VOCAL ENSEMBLE

 Connections

! Remember that CV connections are NOT stored in the Humana patches! If you want to store CV connections
between devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Humana from another CV/Gate device (typically a Ma-
trix or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note
on/off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

Modulation Inputs
These control voltage (CV) inputs can be used for modulating the Filter Cutoff and Resonance parameters, as well as
the Master Volume level.

Audio Out
These are the main audio outputs. When you create a new Humana device, these outputs are auto-routed to the first
available outputs in the I/O device.

209 HUMANA VOCAL ENSEMBLE

210 HUMANA VOCAL ENSEMBLE
Chapter 14
ID8 Instrument Device
 Introduction

The ID8 Instrument device is a synth module packed with great sounds - ideal for quickly creating nice complete ar-
rangements. The sounds have been extracted from various Reason devices and ReFills to guarantee supreme audio
quality.

The Sounds
The ID8 contains 36 presets divided into nine categories, with four sounds in each category. The categories are
these:
• Piano
The Piano category features a grand piano, an upright piano, a dance oriented piano sound and vibes.
• Electric Piano
The Electric Piano category holds two classic electric piano sounds plus a digital FM type piano and a Clav.
• Organ
The Organ category contains two classic tone-wheel organ sounds, one transistor organ sound and a pump organ.
• Guitar
The Guitar category sports an acoustic steel string guitar, a clean electric guitar, a half-acoustic jazz guitar and a
dulcimer.
• Bass
The Bass category features one fingered and one picked electric bass, an acoustic upright bass and a synth bass.
• Strings
The Strings category holds orchestral strings, arco strings, a small string section and a choir sound.
• Brass-Wind
The Brass-Wind category features Fat Brass, Brass Section, French Horns and Flute.
• Synth
The Synth category contains two classic monophonic synth lead sounds and two characteristic polyphonic pad
sounds, one with fast attack and one with slow.
• Drums
The Drums category sports four extensive combinations of drums and percussion instruments aimed at different
musical styles. Each “drum kit” contains between 53 and 65 different instruments, so there is plenty to choose
from!
See “Velocity mapping” for information about the velocity mapping of some of the sounds.

212 ID8 INSTRUMENT DEVICE

 Using the ID8
Selecting Sounds
D Select Category by clicking the Up/Down buttons to the left of the Display.

D Select Sound in the selected Category by clicking on any of the A-D buttons, or by clicking on the Sound name
in the Display.

D Click on the Category name in the ID8 Display to bring up a pop-up where you can select Category or replace
the ID8 device with another device.

At the bottom of the pop-up, you can also choose “Browse Instruments...”. Selecting this allows you to replace the
ID8 device with another instrument device and load a new sound in that device.

Controlling Sounds
Parameter knobs

Each of the Sounds in the ID8 have two preset parameters assigned to the Parameter 1 and 2 knobs. The parameter
names are shown in the small displays to the right of the corresponding knobs, and are different depending on the
selected Sound.

213 ID8 INSTRUMENT DEVICE

 Pitch Bend and Mod Wheel

To the left on the ID8 front panel are the standard Pitch Bend and Mod Wheel. The Pitch Bend range is +/- 2 semi-
tones and is the same for all sounds. The Mod Wheel is assigned a little differently depending on the selected Sound,
but usually controls vibrato. In the Drums Category, however, the Mod Wheel has no effect, except on the Electronic
Drums where it controls the cutoff frequency of a lowpass filter.

Performance Controllers
The sounds in the ID8 also respond to the standard Performance controllers “Sustain Pedal”, “Aftertouch”, “Expres-
sion” and “Breath Control”. The parameter assignments can be a little different depending on selected sound. How-
ever, “Sustain Pedal” always controls sustain and “Expression” always controls volume.

Velocity mapping
A lot of the sounds in the ID8 are multi-sampled. They also have several velocity layers to faithfully reproduce the
original instruments. Some of the sounds also use different types of samples for the highest velocity layer. This
means that instead of just sounding louder, they will also sound different. For example, the Jazz Semi Guitar as well
as the Finger, Pick and Upright basses have glissando or sliding notes in the highest velocity layer. The Arco Strings
have pizzicato (picked) notes in the top velocity layer.

About saving edited Sounds

Since the ID8 is designed as a “preset” sound module, there is no dedicated function for saving edited Sounds. How-
ever, any edits you have made of the parameters in a Sound are automatically stored with the song (document) when
you save the song.
q You could also include one or several ID8 Instrument devices in a Combinator device and save the Combinator
patch. Doing so will automatically store the settings of the ID8 Parameters in the Combinator patch. See “The
Combinator” for more details.

214 ID8 INSTRUMENT DEVICE

Chapter 15
Klang
Tuned Percussion
 Introduction

The Klang Tuned Percussion instrument features an assortment of high-quality multi-sampled tuned percussion in-
struments - perfect for any music style. Each of the multi-sampled instruments can also be tailored and processed in
the high-quality filter, amp, delay and reverb sections.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel overview
The Klang front panel contains the following sections:

1 2

3 4 5 6 7 8

The Klang front panel sections.

• 1. Patch Selector (for browsing, loading and saving patches).
• 2. Master Volume.
• 3. Global performance and “play” controls.
• 4. Instruments section.
• 5. Filter section.
• 6. Amp Envelope section.
• 7. Delay effect section.
• 8. Reverb effect section.

216 KLANG TUNED PERCUSSION

 Using Klang
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Global performance and “play” controls

Note
The Note LED lights up each time Klang receives a MIDI Note On.

Range
D Set the desired Pitch Bend range for the “Pitch” wheel with the up/down buttons, or by click-holding on the
display and dragging up/down.
Range: +/-12 semitones (+/-1 octave) in steps of +/-1 semitone.

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Klang also responds to Pitch Bend MIDI
data from a connected MIDI master keyboard. You set the desired Pitch bend Range with the “Range” control to the
right of the Mod wheel.

Mod
The Mod wheel can be used for controlling the three predefined parameters to the right of the Mod wheel:
• S. Start
Here you set how/if the Mod wheel should affect the Sample Start position of the currently selected instrument.
The parameter is bi-polar, with zero modulation at the 12 o’clock position. A negative value moves the sample start
back and a positive value moves it forward.
! Note that if the S.Start parameter in the Instruments section (see “S. Start (Sample Start)”) is at 0 ms, the sam-
ple start cannot be moved back any further. Similarly, if the S.Start parameter in the Instruments section is at
150 ms, the sample start cannot be moved forward any further.
• F. Freq
Here you set how/if the Mod wheel should affect the Filter Cutoff parameter. The parameter is bi-polar, with zero
modulation at the 12 o’clock position.

217 KLANG TUNED PERCUSSION

 ! Note that if the Cutoff parameter in the Filter section (see “Cutoff”) is at 20 Hz, the frequency cannot be low-
ered any further. Similarly, if the Cutoff parameter in the Filter section is at 25 kHz, the frequency cannot be
raised any further.
• Level
Here you set how/if the Mod wheel should affect the Master Volume parameter. The parameter is bi-polar, with
zero modulation at the 12 o’clock position.
! The volume can never be modulated louder than the maximum +12 dB limit of the Master Volume control.

Panel controls
The Instruments section

Instrument selector

D Click the Instrument name selector to bring up a menu of the available instruments - and then select the de-
sired instrument from the menu.
Alternatively, click and drag up/down in the display above the selector to scroll through the instruments.
! Depending on the instrument size (in MB), it could take a short moment before the entire instrument is loaded
into RAM.
! Also note that the note ranges of the instruments extend outside their “natural” ranges, which could produce
nice artificial effects.
The available instruments are:

218 KLANG TUNED PERCUSSION

 • Alto Glockenspiel

The Alto Glockenspiel was played with hard mallets and was recorded with close mics (stereo) in a large hall,
slightly wet. The samples were taken from Soundiron's Alto Glockenspiel library.
• Bamblong

This instrument is also known as a bamboo log drum, from Indonesia. It was played with rubber mallets and was re-
corded with close mics (stereo) in a studio, dry. The samples were taken from Soundiron's Bamblong library.
• Circle Bells Mallet

This instruments is also known as Blossom Bells. It was played with rubber mallets and was recorded with close
mics (stereo) in a studio, dry. The samples were taken from Soundiron's Circle Bells library.

219 KLANG TUNED PERCUSSION

 • Cylindrum

This is an experimental instrument built from large diameter plastic piping, also known as a tubulum. It was played
with rubber paddle mallets and was recorded with close mics (stereo) in a studio, dry. The samples were taken
from Soundiron's Cylindrum library.
• Imbibaphones

These are wine glasses played with rubber mallets. They were recorded with close mics (stereo) in a studio, dry.
The samples were taken from Soundiron's "Imbibaphones" library.
• Kalimba

The Kalimba is also known as an mbira or thumb piano, from Africa. It was recorded with close mics (stereo) in a
studio, dry. The samples were taken from Soundiron's Kalimba library.

220 KLANG TUNED PERCUSSION

 • Music Box

This is a music box recorded with close mics (stereo) in a studio, dry. The samples were taken from Soundiron's
Musique Box library.
• Noah Bells

The Noah Bells from India are played with the fingertips and were recorded with close mics (stereo) in a large hall,
slightly wet. The samples were taken from Soundiron's Noah Bells library.
• Steel Tones

This instrument is also known as a hank drum or propane drum. It was played with felt mallets and was recorded
with close mics (stereo) in a room, dry. The samples were taken from Soundiron's Steel Tones library.

221 KLANG TUNED PERCUSSION

 • Whale Drum

The whale drum is a wooden box slit drum from Africa. It was played with rubber mallets and was recorded with
close mics (stereo) in a studio, dry. The samples were taken from Soundiron's Whale Drum library.

S. Start (Sample Start)

D Turn the S.Start knob to set where in the sample the playback should start.
Note that the effect could be different depending on the selected instrument.

Oct

D Set the octave transposition for the instrument.

Range: 5 (+/-2) octaves.
! Note that the note ranges of the instruments can be transposed far outside their “natural” ranges, which could
produce nice artificial effects.

Semi

D Set the pitch in semitone steps.

Range: +/-12 semitones (two octaves).

Fine

D Set the pitch in steps of 1 cent.

Range: +/- 50 cents (down or up half a semitone).

222 KLANG TUNED PERCUSSION

 The Filter section

Filter On/Off

D Click the On/Off LED button to switch on/off the Filter section.

Filter Type selector

D Click and drag up/down on the Filter Type selector to select one of the available filter types - or step through
the filter types by clicking the Up/Down arrow buttons.
The available filter types are:
• LP
This is a lowpass filter with 12db/octave slope.
• HP
This is a highpass filter with 12db/octave slope.
• BP
This is a bandpass filter with 6db/octave slopes.
• Comb
This is a comb filter for phaser/flanger type of effects.

Cutoff

D Set the cutoff/center frequency with the Cutoff knob.

The cutoff parameter sets where in the frequency range you want the resonance and attenuation to appear.
Range: 20 Hz to 25 kHz.

223 KLANG TUNED PERCUSSION

 Reso

D Set the resonance amount with the Reso knob.

The resonance parameter amplifies the frequencies at, and around the cutoff/center frequency.

Env

D With the Env knob you set how much you want the Filter Envelope (see below) to affect the Cutoff frequency.
Range: 0% to 100%.

Filter Envelope

The standard ADSR type envelope controls the filter cutoff frequency modulation over time. The ADSR envelope
characteristics are described in detail in “Amp Envelope”.

Vel

D Turn the Vel knob to set how much the cutoff/center frequency should be modulated by Keyboard Velocity.
Range: 0% to 100%.

Kbd

D Turn the KBD (Keyboard Track) knob to set how much the cutoff/center frequency should track incoming MIDI
Notes.
Range: 0% (no tracking (constant frequency)) to 100% (1 semitone per key).

224 KLANG TUNED PERCUSSION

 The Amp section

Vel

D Turn the Vel knob to set how much the amplitude should be modulated by Keyboard Velocity.
Range: 0% to 100%.

Amp Envelope

The Amp Envelope is a standard ADSR envelope which controls the amplitude of the signal over time. The picture
below shows the various stages of the ADSR envelope:

Level

Volume
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

• A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value
set with the Master Volume knob. How long this should take, depends on the Attack setting. If the Attack is set to
“0”, the Volume value is reached instantly. If the Attack value is raised, it will take longer time before the Master
Volume value is reached.

225 KLANG TUNED PERCUSSION

 • D(ecay)
After the Master Volume value has been reached, the level starts to drop. How long this should take is governed
by the Decay parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to
“0”, the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the
volume gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to
drop to some other value than zero, you raise the Sustain parameter.
• S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain
to full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain pa-
rameter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and
stays there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Volume value, then
gradually decreases to finally land to rest on a level somewhere in-between zero and the Master Volume value.
Note that Sustain represents a level, whereas the other envelope parameters represent times.
• R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume
to drop back to zero after you release the key.

The Delay section

This is a stereo delay which affects all voices globally.

Delay On/Off

D Click the On/Off LED button to switch on/off the Delay section.

Time

D Set the time between the delay repeats.

If Sync is active (see below), the Time parameter controls the time divisions.

226 KLANG TUNED PERCUSSION

 Feedback

D Set the number of delay repeats with the Feedback knob.

Sync

D Click the Sync button to sync the delay time to the main sequencer Tempo.
When active, the Time knob (see above) controls the time divisions.

Ping Pong

D Activate this to get the delay repeats alternating between the left and right channels.
Note that this also doubles the delay tempo.

Damp

D Raise the Damp value to gradually cut off the high frequencies of the delay repeats.

Amount

D Set the delay amount with the Amount knob.

Note that the Delay effect is routed as a “send effect”, with the Amount knob working as an “effect return” control.
This means that if you play a short note with the Amount knob set to zero, you might still get a delay effect if you
turn up the Amount afterwards (depending on the Feedback setting, see “Feedback”).

227 KLANG TUNED PERCUSSION

 The Reverb section

This is a stereo reverb which affects all voices globally.

Reverb On/Off

D Click the On/Off LED button to switch on/off the Reverb section.

Time

D Set the reverberation duration time.

In practice this sets the “size” of the reverberation “chamber/room/hall”.

Pre-Delay

D Set the pre-delay time of the reverb.

Range: 0-200 ms.

Hi Damp

D Raise the Hi Damp value to cut off the high frequencies of the reverb and thereby create a smoother, warmer
effect.

228 KLANG TUNED PERCUSSION

 Lo Damp

D Raise the Lo Damp value to cut off the low frequencies of the reverb signal, to make the reverb effect less
“muddy”.

Amount

D Set the reverb amount with the Amount knob.

Note that the Reverb effect is routed as a “send effect”, with the Amount knob working as an “effect return” con-
trol. This means that if you play a short note with the Amount knob set to zero, you might still get a reverb effect if
you turn up the Amount afterwards (depending on the Time setting, see “Time”).

Connections

! Remember that CV connections are NOT stored in the Klang patches! If you want to store CV connections be-
tween devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Klang from another CV/Gate device (typically a Matrix
or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/
off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

Modulation Inputs
These control voltage (CV) inputs can be used for modulating the Filter Cutoff and Resonance parameters, as well as
the Master Volume level.

Audio Out
These are the main audio outputs. When you create a new Klang device, these outputs are auto-routed to the first
available outputs in the I/O device.

229 KLANG TUNED PERCUSSION

230 KLANG TUNED PERCUSSION
Chapter 16
Kong Drum Designer
 Introduction
The Kong Drum Designer gives the visual impression of a pattern-based drum machine, like the legendary MPC
units. Indeed, it does have a matrix of 4 x 4 pads that are used for playing the sounds, just like the aforementioned
classics. There are significant differences, however.
Kong features 16 drum “sound channels” that can host one drum sound each. Each drum sound can consist of a
sound module routed through various types of FX and processing modules, allowing for completely open-ended
sound design possibilities. Individual drum sounds can be saved as Drum Patches and complete drum kits can be
saved as Kit Patches, allowing you to mix and match drum sounds and make up custom kits with ease.
! Please, note that this device is not available in Reason Lite Rack Plugin.

Overview

The Kong front panel.

Kong is an advanced drum sound synthesizer, sampler/sample player and REX loop/slice player with many unique
features. The design could be described as semi-modular, in that the sound, FX and audio processing modules are
open slots that allows you to select between an array of different sound generators, FXs and audio processor types.
As a result, Kong is capable of producing an astounding array of drum and percussion sounds - or any type of sound,
for that matter. While it offers a lot of scope for serious sound design, it still has a straight-forward and user-friendly
interface.
Kong also features audio inputs on the back panel. By connecting the output of another device to these inputs, you
can use Kong’s audio processing modules to process external sound. You can also route drum sounds for audio pro-
cessing in external devices.

The Pad Section

The pad section features 16 pads. Each of these pads can be assigned to a separate Drum sound. You can also
choose to assign several pads to one and the same Drum sound - or to link pads so that one pad will trigger several
other pads as well. To the right of the pads is the Pad Settings area where you can control the pad assignments and
behavior. See “Pad Settings”.

232 KONG DRUM DESIGNER

 The Drum Control Panel
The Drum Control Panel at the bottom left of the panel shows the name and “macro parameter” settings for the se-
lected pad in the pad section. From the Drum Control Panel you can also load and save Drum Patches. See “The
Drum Control Panel”.

The Drum and FX Section

By clicking the Programmer button below at the bottom of the Drum Control Panel you can bring up the Drum and
FX Section. Here is where you can edit your drum sounds and combine with various types of sound processors and
FX. See “The Drum and FX section”.

About using custom backdrops

As with the Combinator device, it is possible to customize the Kong front panel graphics with a user-designed skin. In
the Reason Download section at the Reason Studios website is the “Combi and Kong Backdrop Templates” zip file,
which can be used as starting point for designing your own Kong panel graphics. See the “Read Me.txt” file in the
Backdrops folder for more details. Note that you should use separate backdrops for folded vs. unfolded states.

About file formats

Kong can read the following file types:

Kit Patches
A Kong Kit Patch (Windows extension “.kong”) contains all settings for all 16 Drum sound channels, including file ref-
erences to any used drum samples (but not the actual samples themselves). Switching patches is the same as se-
lecting a new drum kit.

Drum Patches
A Kong Drum Patch (Windows extension “.drum”) contains all settings for the selected Drum sound channel, includ-
ing file references to any used drum samples (but not the actual samples themselves). Switching Drum Patches is
the same as selecting a new drum sound.

Drum Samples
The audio file format support differs depending on which computer OS you are using.
The NN-Nano Sampler module in Kong can read and play back sample files of the following formats:
• In Windows:
.wav, .aif, .mp3, .aac, .m4a and .wma.
• In macOS:
.wav, .aiff, .3g2, .3gp, .mp1, .mp2, .mp3, .mpeg, .mpa, .snd, .au, .sd2, .ac3, .aac, .adts, .amr, .caf, .m4a .m4r and .mp4.
• SoundFonts (.sf2)
SoundFonts are an open standard for wavetable synthesized audio, developed by E-mu systems and Creative
Technologies.
• REX file slices (.rx2, .rex, .rcy)
REX files are music loops created in the ReCycle program or when editing audio clips inline in Reason. The NN-
Nano lets you load separate slices from REX files as individual samples.

233 KONG DRUM DESIGNER

 • Any sample rate and practically any bit depth.
See “NN-Nano Sampler” for details.

REX Files
The Nurse Rex Loop Player module in Kong can read and play back files of the following formats:
• REX files (.rx2, .rex, .rcy)
REX files are music loops created in the ReCycle program or when editing audio clips inline in Reason.
See “Nurse Rex Loop Player” for details.

Using patches
When you create a new Kong device it is loaded with a default kit. If you like you can use the default kit - or you can
load another Kong Kit patch (or create one from scratch, by loading individual Drum patches). A Kong Kit patch con-
tains settings for the 16 Drum channels, complete with parameter settings and file references to any samples used.

Loading a Kit Patch

To load a patch, use one of the following methods:
D Use the browser to locate and open the desired patch.
To open the browser, select “Browse Kong Patches” from the Edit menu or device context menu, or click the folder
button in the patch section on the device panel.

D Once you have selected a patch, you can step between all the patches in the same folder by using the arrow
buttons next to the patch name display.
D If you click and hold on the patch name display on the device panel, a pop-up menu will appear, listing all Kong
Kit patches in all currently expanded folders in the Patch Browser.
This allows you to quickly select another patch without having to step through each one in turn.
D Use the drag and drop method to drag Kong Kit Patch files from the Browser and drop on the Kong panel.
The Kong panel is dimmed in orange and a Patch Replace symbol appears in the center.

234 KONG DRUM DESIGNER

 Checking the sounds in a Kit Patch
There are three ways you can listen to the sounds in a patch without using the main sequencer:
D By clicking the Pad buttons on the front panel.

! Note that the vertical click position on the pad determines the Velocity value. If you click towards the bottom of
a pad, the velocity is low and at the top of each pad the velocity value is high.

Velocity = 127

Velocity = 4

This will give you a good idea about the dynamics behavior of each drum sound. This also allows you to record in
the main sequencer using the full dynamic range of each drum sound, even without a connected MIDI keyboard/
control surface.
D By playing the keys C1 to D#2 or C3 to B6 on your MIDI keyboard or on the On-screen Piano Keyboard.
Pad 11

Pad 14
Pad 16

Pad 10

Pad 11
Pad 11
Pad 12

Pad 13
Pad 14

Pad 15
Pad 15
Pad 16
Pad 2
Pad 4

Pad 7
Pad 9

Pad 1
Pad 2

Pad 3
Pad 3
Pad 4

Pad 5
Pad 6

Pad 7
Pad 7
Pad 8

Pad 9
Pad 10
Pad 12
Pad 13
Pad 15

Pad 10
Pad 10
Pad 11
Pad 12
Pad 12
Pad 13
Pad 13
Pad 14
Pad 14
Pad 15
Pad 16
Pad 16
Pad 1
Pad 3
Pad 5
Pad 6
Pad 8

Pad 1
Pad 1
Pad 2
Pad 2
Pad 3
Pad 4
Pad 4
Pad 5
Pad 5
Pad 6
Pad 6
Pad 7
Pad 8
Pad 8
Pad 9
Pad 9

C1 C2 C3 C4 C5 C6

In the C1-D#2 range, each MIDI note will trigger one pad each, from Pad 1 to Pad 16. In the C3-B6 MIDI note
range each pad can be triggered from three adjacent keys on your MIDI keyboard. C3-D3 trigs Pad 1, D#3-F3
trigs Pad 2 and so on. The C3-B6 note range is perfect if you want to play fast passages by triggering the same
pad from several keys on your MIDI keyboard.

Creating a new Kit Patch

To create a patch of your own (or modify an existing patch), use the following basic steps:
1. Click on the pad for the drum sound you want to load or replace.
A blue frame surrounds the selected pad.

235 KONG DRUM DESIGNER

 2. Click the folder button on the Drum Control Panel.

D Alternatively, right-click (Win) or [Ctrl]-click (Mac) on the Pad and select “Browse Drum Patches...” from the
context menu.
The Patch Browser opens.
3. Locate and open a Kong Drum Patch (extension ‘.drum’) or a sample or REX file.
You will find a selection of Kong Drum Patches in the Factory Sound Bank (in the Kong Drum Patches folder).
Loading a sample will automatically open it in an NN-Nano Sampler module (see “NN-Nano Sampler”) and loading
a REX file will automatically open it in a Nurse Rex Loop Player module (see “Nurse Rex Loop Player”).
D Alternatively, drag a Kong Drum Patch file, a REX file, a sample or a REX slice from the Browser and drop on
the Drum Control Panel - or on any desired drum pad.
Depending on if you drop a Drum Patch file, a REX file or a sample/REX slice, the Drum Control Panel or pad is
dimmed in orange or blue and a Patch/Sample Replace symbol appears in the center.
! It is important that you drop REX files either on the Drum Control Panel or on a pad. Dropping it elsewhere will
replace the entire Kong device with a Dr. Octo Rex device and load the REX file in this device instead!
4. Change some parameter settings for the drum sound channel using the knobs on the Drum Control Panel.
These parameters are described in “The Drum Control Panel”. Note that the Drum Control Panel parameters are
“global” for each Drum channel. Each drum sound can consist of a number of different sound and FX modules,
each with their separate set of parameters. Refer to “The Drum modules”, “The Support Generator modules” and
“The FX modules” for details about all the modules that can be used to build up a complete Drum sound.
5. Repeat steps 1 and 4 for the other drum sound channels.
6. When you’re satisfied with the drum kit, you can save the patch by clicking the Floppy Disk button in the patch
section on the Drum panel.
Note however, that you don’t necessarily need to save the Drum patch - all settings are included when you save a
Kong Kit Patch (see “Saving Kit Patches”) and/or your song.

Creating an empty Kit Patch

To “initialize” the settings in the Kong, select “Reset Device” from the Edit menu or the device context menu. This re-
moves all samples for all drum sound channels, and sets all parameters to their default values.

Saving Kit Patches

Saving patches is done in the same way as with any other Reason device - see “Loading patches” and “Saving
patches”.
! Note that you don't have to save any of the 16 individual Drum patches first if you don’t want to - all settings for
each individual Drum patch are included in the Kong Kit patch.

236 KONG DRUM DESIGNER

 Pad Settings
In the Pad Settings section to the right of the Pad section you can perform various assignments and tricks pertaining
to how the Drum channels should be controlled from the Pads.

Assigning Drums to Pads

Kong features 16 pads and 16 Drum channels, as described earlier. Each pad can control a separate Drum sound
channel. You can also assign several pads to control a single Drum sound channel. This is especially useful if you
want to apply different settings, like Hit Types (see “Assigning Hit Type to Pads”), for each pad so that the controlled
Drum sound channel responds differently. By default the 16 pads are assigned to their corresponding Drum sound
channel; Pad 1 to Drum 1 and so on. If you want to change this assignment, proceed as follows:
1. Select the desired Drum sound channel by clicking on its corresponding pad - or on the Pad name below the
pad if you don’t want the pad to trigger the sound.
A blue frame surrounds the selected pad and the corresponding Drum is displayed in the Drum section to the left.

2. Select the other pad you want to control Drum 1 from.

In this example, we select Pad 2.

3. Click on button 1 in the Drum Assignment section to assign Pad 2 to Drum 1.

Now, Pad 2 is also assigned to play Drum 1. Below Pad 2 it now says “Drum 1” to indicate the current assignment.

237 KONG DRUM DESIGNER

 Assigning Drums to Pads using the Quick Edit function
If you want to assign several Drums to several pads quickly, you can do this by using the Quick Edit function.
1. Click the Quick Edit button in the Drum Assign section.

Each Pad now shows the current Drum assignment.

2. Change the Drum assignment clicking on the desired Drum channel number on each Pad.
3. When you are done, click the Quick Edit button or press [Esc] to exit to normal mode.

Renaming Pads
D Double click on the Pad name below the corresponding Pad, enter a new name and press [Enter].

Copying & Pasting Drums between Pads

It’s possible to copy a Pad with an assigned Drum and paste into another Pad location:
1. Select the source Pad.
2. Select “Copy Drum Patch” from the context menu.
3. Select the destination Pad.
4. Select “Paste Drum Patch” from the context menu.
Now, the complete Drum patch has been duplicated to the destination Pad and you can begin editing it as a separate
patch.

Assigning Hit Type to Pads

If you have assigned several pads to the same Drum sound channel, you can choose a different Hit Type for each of
the pads (where applicable). Depending on Drum sound type, some of the sounds can have up to four pre-defined Hit
Types. These Hit Types are shown in the Hit Type display.

For example, a Synth Hi-Hat Drum sound has four Hit Types by default: “Closed”, “Semi-Closed”, “Semi-Open” and
“Open”. By selecting a different Hit Type for each of the pads assigned to the same Drum, you can create a very nice
and “live” sound.
D To assign a Hit Type to a pad, select the pad and then select Hit Type by clicking the Hit Type button (or on the
name in the display).
The Hit Type assignment is saved when you save your Kong Kit Patch and/or song.

238 KONG DRUM DESIGNER

 Assigning Hit Type to Pads using the Quick Edit function
A quicker way of assigning Hit Type to several pads is by using the Quick Edit function.
1. Click the Quick Edit button in the Hit Type section.

Each Pad now shows the current Hit Type assignment.

2. Change the Hit Type assignment clicking on the desired Hit Type number on each Pad.
3. When you are done, click the Quick Edit button or press [Esc] to exit.

Muting and Soloing Pads

D Click the Mute button to mute the assigned Drum for the selected Pad.
This will also mute MIDI control of the assigned Drum. Muted pads are displayed in red color.
D Click the Solo button to solo the assigned Drum for the selected Pad.
Soloed pads are displayed in green color. All other pads are automatically muted. This also affects MIDI control of
the Drum channels.
D Click the CLR button to remove all Mute and Solo assignments.

Muting and Soloing Pads using the Quick Edit function

A quicker way muting and soloing several pads is by using the Quick Edit function.
1. Click the Quick Edit button at the top of the Pad Settings section.

Each Pad will now show a Mute and Solo button.

2. Click the Mute and/or Solo buttons on the desired Pads.
3. When you are done, click the Quick Edit button to exit.

239 KONG DRUM DESIGNER

 Working with Pad Groups

Kong features 9 Pad Groups, divided into 3 Mute Groups, 3 Link Groups and 3 Alt Groups. Each Pad can be as-
signed to one or more of these 9 Pad Groups independently. Pad Groups are useful if you, for example, want to trig-
ger several pads from a single pad, have one pad mute another, or randomly trigger other pads from one pad.

Mute Groups
Mute Groups can be used if you want one pad to automatically mute another sound in the same Mute Group. For ex-
ample, if you assign an open hi-hat and a closed hi-hat sound to the same Mute Group, playing on one pad will auto-
matically mute the sound assigned to the other pad.

Link Groups
Pads assigned to the same Link Group will play together when you trigger any of the pads in that group.

Alt Groups
If you play pads assigned to the same Alt Group, the pads will be triggered in a random fashion, one by one. It doesn’t
matter which pad you play in the group, the pad triggering is always random.

Assigning Pads to Pads Groups using the Quick Edit function

A quicker way of assigning several pads to Pad Groups is by using the Quick Edit function.
1. Click the Quick Edit button in the Pad Group section.

Each Pad now shows the current Pad Group assignment.

2. Edit the Pad Group assignment by clicking on the desired Pad Group letter on each Pad.

In the picture above, Pads 9 and 10 are assigned to Alt Group “G”, which means they will trigger alternating when
you play any of these Pads.
Pads 11 and 12 are assigned to Mute Group “B”, which means that playing Pad 11 will mute Pad 12 and vice
versa.
3. When you are done, click the Quick Edit button or press [Esc] to exit.

240 KONG DRUM DESIGNER

 The Drum and FX section

Drum Control Panel

Drum Module Slot

FX1 Slot Bus FX Slot

FX2 Slot Master FX Slot

The Drum and FX section in Kong is built up of the Drum Control Panel and the Drum and FX section.
D Click the Show Drum and FX button below the Drum Control Panel to unfold the Drum and FX section.
The Drum and FX section consists of five slots:
• The Drum Module Slot.
• The FX1 Slot.
• The FX2 Slot.
• The Bus FX Slot.
• The Master FX Slot.
The Drum, FX1 and FX2 slots are unique to each of the 16 Drum channels in Kong. The Bus FX and Master FX slots
are shared between all Drum channels in the Kong device. You can activate/deactivate any of the slots by clicking
the On button at the upper left of each slot.

241 KONG DRUM DESIGNER

 Signal flow
The output signal from a Drum module is sent via the FX1 and FX2 Slots to the Bus FX, Master FX or to a pair of the
individual outputs on the back of the Kong panel. There is also an internal Bus FX Send that can be used to send an
audio signal from the Drum via the FX1 and FX2 Slots to the Bus FX. The Bus FX Slot can the hold e.g. a reverb
module which can be used a send effect for all the Drum channels. As an extra bonus, you can also hook up an ex-
ternal effect device between the Bus FX and Master FX Slots, see “Using external effects with Kong”.
The signal routing in the Drum and FX section depends on the Drum Output selector setting at the bottom of the
Drum and FX section:

The different signals flows are described in the following paragraphs:

Master FX Drum Output

When the Drum Output is set to “Master FX”, the signal flow is according to the picture below. If you are using a Bus
FX, this is treated as a Send effect with the Bus FX level controlled by the Bus FX knob on the Drum Control Panel.
Bus FX

Drum Module FX1 FX2

Master FX

Main Out L & R

Signal flow when Drum Output is set to “Master FX”.

242 KONG DRUM DESIGNER

 Bus FX Drum Output
When the Drum Output is set to “Bus FX”, the signal flow is according to the picture below. Note that the Bus FX is
now routed both as an Insert effect and as a Send effect at the same time. Therefore, it might be a good idea to set
the Bus FX Send knob on the Drum Control Panel to zero in this configuration.
Bus FX

Drum Module FX1 FX2

Master FX

Main Out L & R

Signal flow when Drum Output is set to “Bus FX”.

243 KONG DRUM DESIGNER

 Separate Out Drum Output
When the Drum Output is set to any of the separate output pairs “3-4” to “15-16”, the signal flow is according to the
picture below. The signals to the selected separate output pair are taken directly after the FX2 via the Master Level
knob. Note that the signal via the Bus FX and Master FX is still available on the Main Out L & R and can be controlled
with the Bus FX Send knob on the Drum Control Panel.
Bus FX

Drum Module FX1 FX2

Master FX

Main Out L & R

Output 3 & 4

Signal flow when Drum Output is set to any of the separate output pairs “3-4” to “15-16”.

The Drum Control Panel

The Drum Control Panel features a set of “macro controls” that affect parameters in each Drum. These controls scale
the parameters in the Drum module and FX modules in the Drum and FX section. There are also some standard pa-
rameters that are identical for each Drum: Pan, Tone and Level.
• The Pitch Offset knob affects the Pitch parameters in all Drum modules.
No FX modules are affected, even if they feature a Pitch parameter.

244 KONG DRUM DESIGNER

 • The Decay Offset knob affects the amplitude Decay or Release parameters in all Drum modules plus any FX
modules that feature a Decay parameter.
For example, the reverb decay time in the Room Reverb FX module is affected by the Decay Offset parameter.
• The Bus FX Send knob affects the signal level sent to the Bus FX Slot.
Depending on current Drum Output setting, the effect will be a little different - see the examples in “Signal flow”.
• The Aux 1 and Aux 2 Send knobs controls the level to any devices connected to the Aux 1 and Aux 2 Send Out-
puts on the back of the panel, see “Aux Send Out”.
The signals to the Aux Send is tapped after the FX1 and FX2 Slots but before the Bus FX and Master FX Slots.
• The Pan parameter controls the panning of the signal in the stereo panorama.
The Pan parameter affects the signal after the FX1 and FX2 Slots but before it is sent to the Bus FX and Master
FX Slots.
• The Tone parameter is a built-in filter (similar to the filter in Redrum).
The Tone parameter affects the signal after the FX1 and FX2 Slots but before it is sent to the Bus FX and Master
FX Slots.

Editing the Drum Control Panel parameters using the Quick Edit function
A quicker way of editing the Drum Control Panel parameters for several Drum channels at once is by using the Quick
Edit function. The Drum Control Panel features four Quick Edit buttons.
1. Click the Quick Edit button below the Pitch and Decay Offset section.

Each Pad now shows the current Pitch and Decay Offset settings for each assigned Drum channel.
2. Edit the Pitch and Decay Offsets by clicking and dragging the “crosshair” on the desired Pads.

The Decay Offset is on the horizontal X-axis and the Pitch Offset is on the vertical Y-axis, as shown in plain text
on the big red frame around the Pad section. As you move the crosshair, the corresponding knobs on the Drum
Control Panel move as well - and vice versa.
3. When you are done, click the Quick Edit button or press [Esc] to exit - or click another Quick Edit button to
change other sets of parameters.

245 KONG DRUM DESIGNER

 Loading and Saving Drum Patches

Loading and Saving Kong Drum patches (“.drum”) are done in the same way as with any other Reason device - see
“Creating a new Kit Patch”, “Loading patches” and “Saving patches”.
A Kong Drum patch contains all parameter settings on the Drum Control Panel, including modules and parameter
settings in the Drums and FX section - with references to any used samples.
It’s also possible to load samples and REX loops in the Drum Control Panel section. Loading a sample will automati-
cally open it in an NN-Nano Sampler module (see “NN-Nano Sampler”) and loading a REX file will automatically open
it in a Nurse Rex Loop Player module (see “Nurse Rex Loop Player”).

The Drum Module slot

Each Drum channel in Kong has a main module slot - the Drum Module slot - to which you can load one of 9 different
types of drum sound modules for designing drum sounds.
D Select Drum Module type by clicking the button to the right of the On button and selecting the module from the
pop-up.
the following Drum Module types can be selected: NN-Nano Sampler, Nurse Rex Loop Player, Physical Bass
Drum, Physical Snare Drum, Physical Tom Tom, Synth Bass Drum, Synth Snare Drum, Synth Tom Tom and Synth
Hi-Hat. See “The Drum modules” for details about each Drum module.
! Note that only four pre-defined parameters per Drum Module can be automated!
At the bottom below the Drum Slot is the Pitch Bend Range parameter which controls the Pitch Bend Range for
the Drum Slot. This parameter is global for all types of Drum Modules but is unique to each of the 16 Drum chan-
nels.

The Pitch Bend Range knob for each of the 16 Drum channels

246 KONG DRUM DESIGNER

 The FX slots

Each Drum channel also has 2 insert effect slots - the FX 1 and FX 2 Slots - to which you can load one of two dif-
ferent types of support sound generators or one of 9 different effect modules.
D Select Module type by clicking the button to the right of the On button and selecting module from the pop-up.
the following module types can be selected for the FX 1 and FX 2 Slots: Noise generator, Tone generator, Room
Reverb, Transient Shaper, Compressor, Filter, Parametric EQ, Ring Modulator, Rattler, Tape Echo and Overdrive/
Resonator. See “The Support Generator modules” and “The FX modules” for details about each module type.
! Note that only two pre-defined parameters per FX/Support Generator Module can be automated!
• For the Bus FX and Master FX slots, all module types except the Noise and Tone generators can be selected.

247 KONG DRUM DESIGNER

 The Drum modules
! Note that only four pre-defined parameters per Drum Module can be automated!

NN-Nano Sampler

The NN-Nano Sampler is based on the NN-XT Sampler and was designed to be ideal for drums and percussion
sounds.
The NN-Nano can handle samples or sets of samples for each of the four different Hit Types described in “Assigning
Hit Type to Pads”. Each Hit Type can contain one or several samples which can be layered and/or altered and con-
trolled individually via velocity.

Loading samples
1. Select the Hit you want to load the sample(s) into by clicking in the display.

2. Click the Browse Samples (folder) button and select one or several WAV, AIFF or SoundFont Samples or REX
slice files.
3. Click the Load button in the Browser.
The sample(s) are loaded in the selected Hit.

D Alternatively, drag a sample, a REX slice or a SoundFont file from the Browser and drop on the NN-Nano panel.
The NN-Nano panel is dimmed in blue and a Sample Replace symbol appears in the center.
If you selected several samples in the Browser, these will be loaded as separate Layers in the selected Hit.
If you like you can load additional samples, either into another Hit or into a new Layer in the same Hit. To load a new
sample in a new Layer in the same Hit, proceed as follows:

248 KONG DRUM DESIGNER

 1. Select the Hit and then click the Add Layer button.

An additional space is created in the Hit.

2. Select the empty Layer in the display and load a new sample according to the description in “Loading samples”
above.
The NN-Nano Sampler module in Kong can read and play back sample files of the following formats:
• In Windows:
.wav, .aif, .mp3, .aac, .m4a and .wma.
• In macOS:
.wav, .aiff, .3g2, .3gp, .mp1, .mp2, .mp3, .mpeg, .mpa, .snd, .au, .sd2, .ac3, .aac, .adts, .amr, .caf, .m4a .m4r and .mp4.
• SoundFonts (.sf2)
SoundFonts are an open standard for wavetable synthesized audio, developed by E-mu systems and Creative
Technologies.
• REX file slices (.rx2, .rex, .rcy)
REX files are music loops created in the ReCycle program or when editing audio clips inline in Reason (see
“Bounce Clip to REX Loop”). The NN-Nano lets load separate slices from REX files as individual samples.
• Any sample rate and practically any bit depth.

Replacing samples
D To replace one or several samples, select the sample(s) in the display and then load new samples according to
the description in “Loading samples”.
This way it is possible to e.g. replace three selected samples with three new samples in one go.

Adding and Removing Layers

D To add a new Layer to a Hit, select the Hit and click the Add Layer button.

An additional space is created in the Hit for the created Layer.

D To remove a Layer from a Hit, select the Layer and click the Remove Layer button.

The Layer is removed together with its sample (if any).

249 KONG DRUM DESIGNER

 Sample parameters
There are a number of parameters that are unique to each individual sample and Hit in the NN-Nano. These param-
eters are visible in the display for the selected (highlighted) sample:

• Velocity
The Velocity range can be set, either by clicking and dragging the Velocity bar sideways to the right of the sample,
or by clicking and dragging the Vel Lo and Hi values vertically at the bottom of the display.
• Level
Set the sample level by clicking and dragging the Level value up or down in the display.
• Pitch
Set the sample pitch by clicking and dragging the Pitch value up or down in the display.
• Alt
Click the Alt box for several samples in the same Hit to make them play back alternating.
• Hit Name
Edit the Hit Name if you like by clicking in the Hit Name box, typing in a new name and then pressing [Enter]. The
name will appear in the Hit Type display on the main panel (see “Assigning Hit Type to Pads”).
! It’s also possible to select multiple samples and edit them together. If the selected samples have different
Level, Velocity, Range and/or Pitch values this is indicated by an “M” (for multiple) symbol next to the param-
eter:

If you change the values of any of the “M” parameters, all selected samples will get the exact same value.

Global parameters
The parameters located on the panel, outside the display, are global and affect all samples in all Hit groups equally.

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 • Polyphony
“Full” is, as the word implies, full polyphony. This means that all Hits can sound with full polyphony. Several Hits can
also sound together if controlled from separate Pads that are assigned to different Hit Types.
“Exclusive Hits” means that when one Hit plays it will automatically mute any other sounding Hits. The polyphony
is still full within each Hit, though.
“Monophonic” is... well, monophonic.
• Mod Wheel
If you want the Mod Wheel to affect the pitch and/or decay of the sound, you can set this with the Mod Wheel ->
Pitch and/or the Mod Wheel -> Decay knobs. Both parameters are bipolar (+/-).
• Velocity
In the Velocity section you can control how the velocity should affect a number of parameters. The parameters are:
Pitch, Decay, Level, Bend and Sample Start. All parameters are bipolar (+/-).
• Pitch
Here you can set the global Pitch, Pitch Bend Amount and Pitch Bend Time for all samples. The Pitch and Pitch
Bend Amount parameters are bipolar (+/-).
• Osc
In the Osc section you can set the global Sample Start and Reverse parameters for all samples in the NN-Nano.
• Amp Env
The Amp Env section contains an Attack-Decay Envelope and the global Level parameter for all samples. There is
also an envelope trig mode selector for choosing between Gate and Trig mode. In Gate mode (the square symbol),
the Decay time defines the minimum gate time. If you hold down a key or pad on your MIDI keyboard/control sur-
face, the Decay stage will set in after you released the key/pad.

Nurse Rex Loop Player

The Nurse Rex Loop Player is based on the Dr. OctoRex Loop Player but has been modified to be ideal for playing
and triggering drum and percussion sounds.
The Nurse Rex can load standard REX files and play back the loops and/or slices in a variety of ways depending on
the selected Hit Type (see “Assigning Hit Type to Pads”).

Loading REX files

1. Click the Browse Samples (folder) button.

2. Select a REX file and click the Load button in the Browser.

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 D Alternatively, drag a REX file from the Browser and drop on the Nurse Rex panel.
The Nurse Rex panel is dimmed in orange and a Patch Replace symbol appears in the center.
The REX file is loaded in Nurse Rex with the loop shown in the display.

Hit Types (playback modes)

Depending on selected Hit Type, the REX loop will play back differently. The editing possibilities also differs depend-
ing on selected Hit Type for the assigned pad.
• Loop Trig
In Loop Trig mode, you trigger the REX loop to play one single cycle every time you hit the assigned pad. Loop Trig
can also be used together with the “Stop” mode on another pad to immediately stop the loop playback, see “Stop”
below.
Loop range (Start and End)

A REX loop with “Loop Trig” as Hit Type.

D Set Start and End slice, either by clicking and dragging the S and E numerical values up/down in the boxes, or
by clicking and dragging the handles sideways in the “ruler” above the REX loop in the display.

Different ways of editing the Start and End Slice values.

• Chunk Trig
In Chunk Trig mode, you can assign several pads to play back shorter sections - chunks - of the REX loop. The
number of chunks is determined by the number of pads you have assigned to the REX loop using the Chunk Trig
Hit Type. The chunks are distributed in equal sections across the REX loop. Chunk Trig can also be used together
with the “Stop” mode on another pad to immediately stop the chunk playback, see “Stop” below.
In the picture below, we have assigned four pads to the same REX loop and we have selected “Chunk Trig” as Hit
Type on all four pads:

Four Chunks distributed

equally across the REX loop

Pad 1-4 assigned to the same

Nurse Rex module

Four pads assigned to the same REX loop and Hit Type set to “Chunk Trig”.

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 D Set the size of the chunks by clicking and dragging the right edges of the “tabs” above the REX loop in the dis-
play.
Doing so will automatically move the start position of the subsequent chunk so that the chunks will always be ad-
jacent to each other.

Editing the sizes of the chunks.

D Change the Start position of the first chunk and the End position of the last chunk by changing the REX loop
Start and End values.
In effect, this is the same as setting the overall REX loop start and end position.

Editing the start position of the first chunk and end position of the last chunk.
• Slice Trig
In Slice Trig mode, you can assign a pad to play back one single slice of the REX loop - or several slices alternat-
ingly. By default, Slice 1 of any REX loop loaded into the Nurse Rex is set to play back when you have selected
“Slice Trig” as Hit Type.

Slice 1 plays back by default

in Slice Trig mode

A REX loop with “Slice Trig” as Hit Type.

D Change the slice to play back by first removing the tick in the Trig checkbox and then clicking on another slice
in the display and ticking the Trig checkbox for that slice.

Slice 3 plays back instead

Slice 3 selected for playback in Slice Trig mode.

Another way of assigning a slice for playback, or to assign several slices to play back alternating, is by using the
mouse in combination with the [Ctrl](Win)/[Cmd](Mac) key.

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 D Hold down [Ctrl](Win)/[Cmd](Mac) and click on the slice(s) in the display you want to assign or deassign.

Slices 3, 5, 8 and 11 selected and will now play

back alternating in Slice Trig mode

Slices 3, 5, 8 and 11 selected for playback in Slice Trig mode, forcing them to play back alternating.
Selected slices are displayed with a red background. The currently “focused” slice is displayed with an orange
background. Selected slices also get their corresponding Trig checkbox ticked automatically.
• Stop
The fourth Hit Type is named “Stop”. The Stop mode can be used if you want to use a pad for immediately stop-
ping the currently playing REX loop or Chunk. The Stop mode should be used in combination with any of the Hit
Types “Loop Trig” or “Chunk Trig”, otherwise it won’t be useful.

“Stop” selected as Hit Type for a pad assigned to a Nurse Rex module.
1. Assign one pad to a REX loop in Nurse Rex and select any of the Hit Types “Loop Trig” or “Chunk Trig”.
2. Assign another pad to the same Nurse Rex module and select “Stop” as Hit Type.
Now, when you play the first pad, the loop or chunk will play. Once you hit the second pad, the loop/chunk play-
back will immediately stop.

Combining Hit Types

By combining the different Hit Types for the Nurse Rex module you can create really interesting setups. For example,
you could load a REX loop and assign a couple of pads to the “Chunk Trig” Hit Type, one pad to Loop Trig, another
one to Slice Trig and another pad to Stop. Playing the different pads can now generate really inspiring results. The
picture below shows an example of this type of setup:

Eight pads assigned to the same Nurse Rex module, with the pads set to different Hit Types (in Quick Edit mode).

254 KONG DRUM DESIGNER

 If we click the Hit Type Quick Edit button, we can see that Pad 1 is set to Loop Trig, Pads 2-6 are set to Chunk Trig,
Pad 7 is set to Slice Trig and has four slices set to Trig in the REX loop display for alternate playback. Finally Pad 8
is set to Stop so we could stop the loop and chunks playback whenever we like.

Editing Slice Parameters

In the REX loop display you can edit parameters that are unique to each separate slice:

Slice parameters

Slices parameters for a loaded REX loop.

D To select a slice for editing, click on the desired slice in the REX loop display.
Alternatively, click and drag up/down in the Slice number box or use the Slice Select knob below the Slice item.
The Slice Parameters are:
• Trig
Click the Trig check box for the slices you want to alternate between using the Slice Trig Hit Type.
• Pitch
Set the pitch for each individual slice in the REX loop by clicking and dragging the Pitch value up/down.
• Level
Set the level for each individual slice in the REX loop by clicking and dragging the Level value up/down.
• Reverse
Click the Reverse box for the slices you want to play back backwards.

The Nurse Rex panel parameters

On the Nurse Rex panel you can edit parameters that are common to all slices in the loaded REX loop:

• Env Type
Sets the amplitude envelope type to “Gate” or “ADSR” (Attack, Decay, Sustain, Release). In Gate mode, the gate
time is set with the Decay parameter.
• Attack with Velocity control
Sets the attack time for the amplitude envelope when ADSR is selected as Env Type. The attack time can also be
velocity controlled according to the sensitivity set with the Vel knob.
• Decay with Velocity and Modulation controls
Sets the decay time for the amplitude envelope when ADSR is selected as Env Type. When Gate is selected as
Env Type, the Decay parameter sets the gate time. The decay/gate time can also be velocity controlled according
to the sensitivity set with the Vel knob. You can also control the decay/gate time from the Mod Wheel with the
amount set with the Mod knob.

255 KONG DRUM DESIGNER

 • Sustain
Sets the sustain level of the amplitude envelope when ADSR is selected as Env Type. In Gate mode, the Sustain
parameter has no effect.
• Release with Velocity and Modulation controls
Sets the release time for the amplitude envelope when ADSR is selected as Env Type. The release time can also
be velocity controlled according to the sensitivity set with the Vel knob. You can also control the release time from
the Mod Wheel with the amount set with the Mod knob. In Gate mode, the Release parameter has no effect.
• Pitch with Velocity control
Sets the overall pitch of all slices in the REX loop. The pitch can be velocity controlled according to the Vel knob
setting. A negative Vel setting will lower the pitch with increasing velocity and a positive setting will raise the pitch
with increasing velocity.
• Level with Velocity control
Sets the overall level of all slices in the REX loop. The level can be velocity controlled according to the Vel knob
setting. A negative Vel setting will lower the level with increasing velocity and a positive setting will raise the level
with increasing velocity.
At the top of the Nurse Rex panel is a button for setting the polyphony:

• Polyphonic means full polyphony

Retriggering the same slice/chunk will keep on adding more voices without muting any sustaining sounds.
• Monophonic will make any new triggered loop/slice/chunk mute any currently playing/sustaining sound.

Physical Bass Drum, Snare Drum and Tom Tom

The Physical Bass Drum, Snare Drum and Tom Tom use very faithful mathematical models for generating acoustic
drum sounds. The sounds of the PM drums are generated using physical modelling; mathematical real-time calcula-
tions of physical acoustic phenomena. The physical modelling technique allows for a lot more creative freedom, and
much wider sonic ranges, compared to sample playback.

General parameters
• Level
This controls the overall output level of the Drum module to the FX1 and FX2 Slots (see “Signal flow”). The Level
is also affected by velocity.

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 Drum head and shell parameters
The Physical Modelling Drums feature the following drum head and shell parameters:
• Pitch
Sets the overall pitch of the drum. The Pitch parameter can be considered the total size of the drum and affects all
other head and shell parameters.
• Tune 1 and Tune 2 (PM Bass Drum and PM Tom Tom)
The Tune 1 and Tune 2 parameters set the drum’s harmonic character, similar to the effect of individually adjusting
the rim tension screws of the top drum head.
• Tune (PM Snare Drum)
This controls the top drum head tension and thus affects the harmonic character of the sound.
• Bend Amount (PM Bass Drum and PM Tom Tom)
This sets the dynamic “pitch bend” effect you get when hitting a drum head.
• Damp
This controls the damping of the drum head.
• Decay
The Decay parameter doesn’t have any actual counterpart in real life. It simply controls the decay time of the drum
sound.
• Shell Level (PM Bass Drum and PM Tom Tom)
This controls how much of the drum shell sound should be present in the sound.
• Shell Size (PM Tom Tom)
This controls the depth (“length”) of the shell.
• Edge Tune (PM Snare Drum)
This controls the head tuning when Hit Type 4 (Edge Hit) is selected for the pad (see “Assigning Hit Type to
Pads”).
• Snare Tension (PM Snare Drum)
This controls the tension of the snare and the distance between the snare and bottom drum head.
• Bottom Pitch (PM Snare Drum)
This controls the pitch of the bottom drum head.
• Bottom Mix (PM Snare Drum)
This controls how much of the bottom head sound should be present in the drum sound.

Beater and Stick parameters

• Density (PM Bass Drum)
This controls the “hardness” of the bass drum pedal beater.
• Tone (PM Bass Drum and PM Tom Tom)
This is a filter which controls the tone of the hit.
• Beater Level (PM Bass Drum)
This controls how much of the beater hit sound should be present in the drum sound.
• Stick Level (PM Tom Tom)
This controls how much of the drum stick hit sound should be present in the drum sound.

257 KONG DRUM DESIGNER

 Synth Bass Drum, Snare Drum and Tom Tom

The Synth Bass Drum, Snare Drum and Tom Tom use analog modelling to generate classic synth drum sounds. The
Synth Tom Tom was faithfully modelled after a famous hexagonal shaped analog drum system from the 80’s.

General parameters
• Level
This controls the overall output level of the Drum module to the FX1 and FX2 Slots (see “Signal flow”). The Level
is also affected by velocity.

Drum parameters
The Synth Drums feature the following parameters:
• Pitch
This sets the overall pitch of the drum. The Noise pitch is not affected by this parameter.
• Tone (Synth Bass Drum)
This is a filter similar to the one used in Redrum and affects the tone of the drum. The Noise is not affected by this
parameter.
• Attack (Synth Bass Drum)
Sets the attack time of the drum sound. This also affects the Noise.
• Decay
Sets the Decay time of the drum sound. This also affects the Noise decay on the Synth Bass Drum and is added
to the Noise Decay parameter on the Synth Snare and Synth Tom Tom drums. It is also added to the Harmonic
Decay value on the Synth Snare Drum. The Decay time is also affected by velocity.
• Harmonic Balance (Synth Snare Drum)
Sets the level balance between the fundamental tone and the harmonic tone.
• Harmonic Frequency (Synth Snare Drum)
Sets the frequency of the harmonic tone.
• Harmonic Decay (Synth Snare Drum)
Sets the decay time of the harmonic tone. This is also affected by the Decay parameter.
• Click Frequency (Synth Bass Drum)
Sets the frequency of the click sound in the attack.
• Click Resonance (Synth Bass Drum)
Sets the resonance amount of the click sound in the attack.
• Click Level (Synth Bass Drum and Synth Tom Tom)
Sets the level of the click sound in the attack.

258 KONG DRUM DESIGNER

 • Bend Time (Synth Bass Drum and Synth Tom Tom)
Sets the time it should take to change the pitch from the Bend Amount value (se below) back to the original pitch.
• Bend Amount (Synth Bass Drum and Synth Tom Tom)
Sets the upper pitch to bend from. The Bend Amount is also affected by velocity.
• Noise Tone (Synth Snare Drum and Synth Tom Tom)
This is a filter which sets the frequency content of the noise.
• Noise Decay (Synth Snare Drum and Synth Tom Tom)
This sets the decay of the noise in the sound. The Noise Decay is also affected by the regular Decay parameter.
• Noise Mix (Synth Snare Drum and Synth Tom Tom)
Sets the noise level in the drum sound.

Synth Hi-hat

The Synth Hi-hat uses analog modelling to generate sounds. The Synth Hi-hat can be used for generating the typical
hi-hat sounds of the early analog drum machines.

Parameters
• Pitch
This sets the overall pitch of the hi-hat sound.
• Decay
This sets the decay time of the hi-hat sound.
• Level
This controls the overall output level of the Synth Hi-hat module to the FX1 and FX2 Slots (see “Signal flow”). The
Level is also affected by velocity.
• Click
This controls the click level in the attack of the hi-hat sound.
• Tone
This is a filter similar to the one used in Redrum and affects the frequency content of the hi-hat sound.
• Ring
Sets the level of the resonance peaks in the sound. The higher the value, the more “metallic” the sound.

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 The Support Generator modules
There are two types of Support Generator modules in Kong, one for generating noise and another one for generating
a tone. The Support Generator modules can be used as companions to any of the Drum modules, or stand-alone. The
Support Generators can be loaded into the FX1 and/or FX2 slots.
! Note that only two pre-defined parameters per Support Generator module can be automated!

Noise Generator

• Hit Type buttons

These buttons allow you to choose for which Hit Type(s) the Noise generator should be active. By default, the
Noise generator is active for all Hit Types (see “Assigning Hit Type to Pads”).
• Pitch
This sets the center pitch of the noise.
• Attack
This sets the attack time of the noise.
• Decay
This sets the decay time of the noise.
• Reso
This sets the resonance amount of the noise around the center pitch.
• Sweep
This sets the upper start pitch of the sweep range. The Sweep range is also affected by velocity.
• Click
This sets the level of the click in the attack of the noise.
• Level
This sets the overall level of the Noise generator. The level is also affected by velocity.

260 KONG DRUM DESIGNER

 Tone Generator

• Hit Type buttons

These buttons allow you to choose for which Hit Type(s) the Tone generator should be active. By default, the Tone
generator is active for all Hit Types (see “Assigning Hit Type to Pads”).
• Pitch
This sets the pitch of the oscillator.
• Attack
This sets the attack time of the tone.
• Decay
This sets the decay time of the tone.
• Bend Decay
This sets the decay time for the Bend.
• Bend
This sets the upper start pitch of the bend range. The Bend range is also affected by velocity.
• Shape
This sets the tonal character of the sound, from less to more harmonics.
• Level
This sets the overall level of the Tone generator. The level is also affected by velocity.

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 The FX modules
The FX modules can be used in any of the FX1, FX2, Bus FX and Master FX slots.
! Note that only two pre-defined parameters per FX Module can be automated!

Using CV modulation of Bus FX and Master FX parameters

When the FX modules are used in the Bus FX and/or Master FX slots, it is possible to route external CV signals to
the first two Effect module parameters for modulation. If you hover with the mouse over the first or second parameter
of an FX module loaded in the Bus FX or Master FX slot, a tool tip appears:

The tool tip shows which CV modulation input on the back of the unfolded Kong panel will control that parameter. For
FX modules loaded in the Bus FX slot, the tool tip displays “Bus FX P1: nn” for the first FX module parameter and
“Bus FX P2: nn” for the second one. For FX modules loaded in the Master FX slot, the tool tip instead reads “Master
FX P1: nn” for the first FX module parameter and “Master FX P2: nn” for the second one. The “nn” in the tool tip in-
dicates the current parameter value.
By connecting cables to the CV modulation inputs on the back of the Kong panel, you can modulate the correspond-
ing FX module parameters in the Bus FX and/or Master FX slots.

D Control the FX parameter modulation amounts with the corresponding attenuation knobs.
If you decide to replace the FX modules in the Bus FX and/or Master FX slots, the modulation routing will be pre-
served - but the CV signals will now control the first two parameters of the replacement module(s).

262 KONG DRUM DESIGNER

 Drum Room Reverb

The Drum Room Reverb is a reverb with a room-type reverb algorithm. It’s perfect for adding ambience to single drum
sounds or to the entire mix of all 16 drum channels. The parameters are as follows:
• Size
This sets the “size” of the room, from small to large.
• Decay
This sets the reverb decay time.
• Damp
This sets the high frequency damping amount of the reverb effect, from none to heavy.
• Width
This sets the stereo effect of the reverb, from mono to wide stereo.
• Dry/Wet
This sets the mix between Dry (no effect) and Wet (reverb) signal.

Transient Shaper

The Transient Shaper is a type of dynamics processor which produces a result that could be compared to that of a
compressor. As opposed to a “normal” compressor, the Transient Shaper mainly affects the signal’s attack, or tran-
sients in the signal, making the signal transients cut through in the mix. The parameters are as follows:
• Attack
A positive Attack value will produce an amplified attack/transient whereas a negative value will reduce the attack/
transient volume.

263 KONG DRUM DESIGNER

 • Decay
This sets the decay time from amplification/attenuation back to normal amplitude level.
• Amount
This controls the amplification amount. A high Amount in combination with a positive Attack value will produce a
very pronounced attack/transient in the sound.

Compressor

The Compressor levels out the audio, by making loud sounds softer. To compensate for the volume loss, the Com-
pressor has a make-up gain control for raising the overall level by a suitable amount. The result is that the audio levels
become more even and the sounds can get more “power” and longer sustain. The parameters are as follows:
• Amount
This sets the sensitivity of the compressor. A high amount will make the compressor more sensitive and react to
weak input signals.
• Attack
This sets how fast the compression should be applied to the incoming signal. A low value will make the compres-
sion set in immediately whereas a high value will let the attack/transients through before compression sets in.
• Release
This sets how long it should take before the compressor lets the sound through unaffected again. Set this to short
values for more intense, “pumping” compressor effects, or to longer values for a smoother change of the dynamics.
• Make up gain
This sets the overall level compensation. A low value will produce a softer output signal whereas a high value will
amplify the output signal.

264 KONG DRUM DESIGNER

 Filter

The Filter is a state variable filter with a switch for selecting Lowpass, Bandpass or Highpass state. It has controls for
cutoff/center frequency and resonance amount and can also be controlled from a built-in MIDI controlled envelope
generator for sweeping the frequency. When used in the Bus FX Slot, MIDI Note E2 (#52) trigs the envelope. When
used in the Master FX Slot, MIDI Note F2 (#53) trigs the envelope. The parameters are as follows:
• Frequency
Sets the cutoff frequency in the LP and HP states and center frequency in the BP state.
• Resonance
This sets the amplification amount of the frequencies around the cutoff/center frequency.
• LP/BP/HP
Sets the state of the filter to either Lowpass, Bandpass or Highpass.
• MIDI Trig EG Amount
This sets the amount of the MIDI controlled filter envelope. The Amount value is bipolar (+/-). Set to a positive
value, the envelope will sweep the filter frequency from a high value down to the set Frequency value. Set to a neg-
ative Amount, the envelope will sweep the filter frequency from a low value up to the set Frequency value. The
Amount is also affected by velocity.
• MIDI Trig EG Decay
This sets the MIDI controlled envelope decay time.

Parametric EQ

The Parametric EQ is a single-band parametric equalizer with controls for center frequency, gain and bandwidth (Q-
value). The parameters are as follows:

265 KONG DRUM DESIGNER

 • Frequency
Sets the center frequency of the equalizer.
• Gain
Sets the amplification (positive Gain value) or attenuation (negative Gain value) around the center Frequency.
• Q
Sets the bandwidth around the center Frequency, from wide to a narrow peak.

Ring Modulator

The Ring Modulator takes the input signal and multiplies it with an internal sinewave signal. The result is often a syn-
thetic metallic sound. The Ring Modulator also features a MIDI controlled envelope generator for sweeping the inter-
nal sinewave frequency. When used in the Bus FX Slot, MIDI Note E2 (#52) trigs the envelope. When used in the
Master FX Slot, MIDI Note F2 (#53) trigs the envelope. The parameters are as follows:
• Frequency
Sets the frequency of the internal sinewave oscillator. The higher the frequency, the higher the resulting output
signal pitch.
• Amount
Sets the level of the internal sinewave oscillator. The higher the level, the more the ring modulation effect.
• MIDI Trig EG Amount
This sets the amount of the MIDI controlled envelope. The Amount value is bipolar (+/-). Set to a positive value, the
envelope will sweep the internal sinewave oscillator frequency from a high value down to the set Frequency value.
Set to a negative Amount, the envelope will sweep the oscillator frequency from a low value up to the set Fre-
quency value. The Amount is not affected by velocity.
• MIDI Trig EG Decay
This sets the MIDI controlled envelope decay time.

266 KONG DRUM DESIGNER

 Rattler

The Rattler adds the effect of a snare “attached” to whatever sound is fed through it. Using the Rattler in combination
with other types of sounds than “usual” snare drum sounds can produce really interesting results! Ever played a snare
bass drum, or a snare hi-hat, for example? The parameters are as follows:
• Snare Tension
This sets the tension of the snare. Note that when the Snare Tension is increased, the effect will actually be less
pronounced since the snare will have “less contact” with the sound source.
• Tone
This is a filter similar to the one used in Redrum and affects the frequency content of the output signal.
• Decay
This sets how long the snare will “ring”.
• Tune
This sets the snare tuning, from low to high, and affects the frequency content of the signal.
• Level
This sets the overall level of the Rattler. The level is also affected by velocity.

Tape Echo

The Tape Echo is based on the principles of classic tape echo effects. The original tape echo effects were electro-
mechanical devices that used an endless magnetic tape in combination with recording and playback heads inside the
box. Depending on the speed of the tape, and on which playback heads were used, the echo repetition and echo pat-
terns could be controlled. Later on, a lot of tape echo effects were replaced by digital delay effects. The Tape Echo
in Kong simulates the classic tape echo effect and features the following parameters:

267 KONG DRUM DESIGNER

 • Time
This sets the time between the delays, from short to long.
• Feedback
This sets the number of delay repetitions, from one to... many.
• Wobble
This sets the tape speed wobbling effect. Since it emulates a magnetic tape, a wobbling speed also automatically
produces a wobbling pitch of the signal.
• Frequency
This sets the change in frequency of the delay repetitions. For every delay, the frequency content will shift accord-
ing to the Frequency setting. A low value will make each repetition sound muddier than the previous one, whereas
a high value will make each delay sound brighter.
• Resonance
This sets the resonance amount of the delay repetitions. Depending on the Frequency parameter setting above,
different frequencies will be amplified.
• Dry/Wet
This is a traditional dry/wet parameter for controlling the relationship between unprocessed and processed signal.

Overdrive/Resonator

The Overdrive/Resonator is a combined distortion and resonator module. It can be used to add a nice distortion to
the input signal. There is also a resonator section with a number of selectable characteristics, similar to the Body sec-
tion in the Scream 4 Sound Destruction Unit. The parameters are as follows:
• Drive
Sets the overdrive distortion amount.
• Resonance
Sets the resonance amount for the resonator.
• Size
Sets the size of the virtual “resonance chamber”, from small to large.
• Model
Click to select one of five different resonator “body” characteristics.

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 Connections
On the back panel of Kong are a number of connectors. Many of these are CV/Gate related. Using CV/Gate is de-
scribed in the chapter “Routing Audio and CV”.

Sequencer Control
The Sequencer Control CV and Gate inputs allow you to play Kong from another CV/Gate device (typically a Matrix
or a Redrum). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/
off along with velocity.

Modulation Input
These control voltage (CV) inputs (with associated voltage trim pots), can modulate various Kong parameters from
other devices. These inputs can control the following parameters:
• Volume
This controls the Master Level in Kong.
• Pitch
This controls the Pitch Bend wheel in Kong.
• Mod
This controls the Mod Wheel in Kong.

Aux Send Out

The two stereo Aux Send Outputs can be used for connecting external effect devices for external signal processing.
The signal levels to these Aux Send Outputs are controlled from the Aux 1 and Aux 2 Send knobs on the Drum Con-
trol Panel, see “The Drum Control Panel”.

269 KONG DRUM DESIGNER

 Gate In and Out
• The Gate Inputs can receive a CV signal to trigger each of the 16 pads individually.
You are still able to control the pads from the panel and/or via MIDI even when the Gate Inputs are being used.
• The Gate Outputs send out a CV Gate signal each time the corresponding pad is played.
The Gate signals can be used for triggering sounds in other devices.

Audio Out 3-16

There are 14 separate audio output jacks on Kong’s back panel - arranged as seven separate stereo pairs. These
outputs are never auto-routed but can be manually connected and selected as individual outputs for any of the Drum
channels by using the Drum Output selector in the Drum and FX section, see “Signal flow”.

Main Audio Out

These are the main audio outputs. When you create a new Kong device, the Main Audio Output pair is auto-routed to
the first available outputs of the I/O device.

Using Kong as an effect device

Besides using the wide array of internal sound possibilities in Kong, you can also use it as an external effect device.
If you unfold the Drum and FX section and flip the rack around, a set of additional audio jacks are visible at the bot-
tom of the back panel.

These audio jacks can be used for connecting external devices and processing their audio in Kong. As you can see,
the signal flow for processing external audio is printed on the back panel. Even if you want to use Kong for process-
ing external signals, you can still play and use its internal Drum channels just like before.
Proceed as follows to connect an external device for audio processing in Kong:
1. Connect the outputs of your other device (a synth, for example) to the Audio Inputs to the left.
If your device only has a mono output, connect it to the Left Audio Input on Kong.

270 KONG DRUM DESIGNER

 2. Play a couple of notes on your other device.
The audio is now routed via Kong’s Bus FX slot and further via the Master FX slot, to the Main Audio Outputs of
Kong.
3. Select suitable FX devices for the Bus FX and Master FX slots in Kong according to the descriptions in “The
FX modules” and tweak the parameters to your liking.
D By connecting CV signals to the Parameter inputs in the Bus FX and Master FX sections on the back of the
panel, you can modulate the first two parameters on the selected FX modules, see “Using CV modulation of
Bus FX and Master FX parameters”.
Attenuate the CV signal with the corresponding knobs next to the modulation jacks.

Using external effects with Kong

It’s also possible to hook up an external effect device in the signal chain to process the Kong audio, or to process any
external audio routed via the Audio Inputs to the left on the unfolded back panel of Kong.
D Connect the external effect device to the External Effect Outputs and Inputs.
In the picture below, an RV7000 Reverb is connected to Kong’s External Effect section allowing the RV7000 to
process the signal between Kong’s internal Bus FX and Master FX slots:

An RV7000 Reverb connected to Kong for processing the Kong audio signals
! Note that if you have selected “Master FX” or “Separate Out” as output in the Drum Output selector, the BUS
FX Send knob on the Drum Control Panel controls the signal level also to the External Effect, see “Signal
flow”.

271 KONG DRUM DESIGNER

272 KONG DRUM DESIGNER
Chapter 17
Malström Synthesizer
 Introduction
The Malström is a polyphonic synthesizer with a great number of different routing possibilities. It is based on the con-
cept of what we call “Graintable Synthesis” (see below), and is ideally suited for producing swirling, sharp, distorted,
abstract special effect types of synthesizer sounds. In fact, you could go so far as to say that the Malström can pro-
duce sounds quite unlike anything you’ve ever heard from a synthesizer.
For a complete run-down of the principles behind it and thorough explanations of the controls, read on...
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Features
The following are the basic features of the Malström:
• Two Oscillators, based on Graintable Synthesis.
See “The Oscillator section” for details.
• Two Modulators, featuring tempo sync and one-shot options.
See “The Modulator section”.
• Two Filters and one Shaper.
A number of different filter modes in combination with several routing options and a Waveshaper makes it possible
to create truly astounding filter effects.
• Three Envelope generators.
There is one amplitude envelope for each oscillator and a common envelope for both filters. See “The amplitude
envelopes” and “The Filter Envelope” for details.
• Polyphony of up to 16 voices.
• Velocity and Modulation control.
See “The Velocity controls”.
• A number of CV/Gate Modulation possibilities.
See “Modulation Input”.
• A variety of Audio Input/Output options.
You can for instance connect external audio sources for input to the Malström, and you can also control its output.
See “Audio Input” for more details.

274 MALSTRÖM SYNTHESIZER

 Theory of operation
There are a number of different synthesis methods for generating sound, e.g. subtractive synthesis (which is used in
the Subtractor), FM synthesis, and physical modelling synthesis to mention but a few.
To give you a clear understanding of the inner workings of the Malström, it might be in order with a brief explanation
of what we call Graintable Synthesis.
What we refer to as graintable synthesis is actually a combination of two synthesis methods, granular synthesis and
wavetable synthesis.
• In granular synthesis, sound is generated by a number of short, contiguous segments (grains) of sound, each
typically from 5 to 100 milliseconds long.
The sound is varied by changing the properties of each grain and/or the order in which they are spliced together.
Grains can be produced either by a mathematical formula or by a sampled sound. This is a very dynamic synthesis
method capable of producing a great variety of results, although somewhat hard to master and control.
• Wavetable synthesis on the other hand, is basically the playback of a sampled waveform.
An oscillator in a wavetable synth plays back a single period of a waveform, and some wavetable synths also allow
for sweeping through a set of periodic waveforms. This is a very straightforward synthesis method, easily con-
trolled, but somewhat more restricted in results. The Malström combines these two into a synthesis method that
provides a very flexible way of synthesizing sounds with incredible flux and mutability.
The Malström combines these two into a synthesis method that provides a very flexible way of synthesizing sounds
with incredible flux and mutability.
It works like this:
• The oscillators in the Malström play back sampled sounds that have been subjected to some very complex pro-
cessing and cut up into a number of grains.
• A set of these periodic waveforms (grains) are spliced together to form a Graintable, which may be played
back to reproduce the original sampled sound.
• A Graintable may be treated just like a wavetable; e.g., you may choose to sweep through it, to move through
it at any speed without affecting pitch, to play any section of it repeatedly, to select from it static waveforms, to
jump between positions, etc., etc.
• It is also possible to perform a number of other tricks, all of which are described further on in this chapter.

Loading and Saving Patches

Loading and saving patches is done in the same way as with any other Reason device - see “Loading patches” and
“Saving patches”.

275 MALSTRÖM SYNTHESIZER

 The Oscillator section

The two oscillators (osc:A and osc:B) of the Malström are the actual sound generators, and the rest of the controls
are used for modulating and shaping the sound. The oscillators actually do two things; they play a graintable and gen-
erate the pitch:
• A graintable is several short, contiguous segments of audio (see above).
• Pitch is the frequency at which the segments are played back.
When creating a Malström patch, the fundamental first building block is usually to select a graintable for one or both
of the oscillators.
D To activate/deactivate an oscillator, click the On/Off button in the top left corner.
When an oscillator is activated, the button is lit.

An activated oscillator.
D To select a graintable, either use the spin controls or click directly in the display to bring up a pop-up menu
with the available graintables.
The graintables are sorted alphabetically into a number of descriptive categories, giving a hint as to the general
character of the sound. Note that the categories are only visible in the pop-up menu, not in the display.

Selecting a graintable by clicking in the Oscillator display.

276 MALSTRÖM SYNTHESIZER

 Setting oscillator frequency
You can change the frequency - i.e. the tuning - of each oscillator by using the three knobs marked “Octave”, “Semi”
and “Cent”.

D The Octave knob changes the frequency in steps of one full octave (12 semitones).
The range is -4 – 0 – +4 where 0 corresponds to middle “A” on your keyboard at 440 Hz.
D The Semi knob changes the frequency in steps of one semitone.
The range is 0 to +12 (one full octave up).
D The Cent knob changes the frequency in steps of cents, which are 100ths of a semitone.
The range is -50 – 0 – +50, i.e. down or up by up to half a semitone.

Controlling playback of the graintable

Each oscillator features three controls that determine how the loaded graintables are played back. These are: The
“Index” slider, the “Motion” knob and the “Shift” knob.

D The Index slider sets the playback starting point in the graintable.
By dragging the slider, you set which index point in the graintable should be played first when the Malström re-
ceives a Note On message. Playback will then continue to the next index point according to the active graintable.
With the slider all the way to the left, the first segment in the graintable is also the one that will be played back first.
! Note that the Malström’s Graintables are not all of the same length, and that the range for the Index slider (0-
127) does not reflect the actual length of the graintables. I.e. regardless of whether a graintable contains 3 or
333 grains, the Index slider will always span the entire graintable even though the slider range says 0-127.
D The Motion knob controls how fast the Malström should move forward to play the next segment in the graint-
able, according to its motion pattern (see below).
If the knob is kept in the middle position the speed of motion is the normal default. Turning the knob to the left
slows it down and turning it to the right results in higher speed. If the knob is set all the way to the left, there will
be no motion at all, which means that the initial segment, as set with the Index slider, will play over and over as a
static waveform.
D The Shift knob changes the timbre of the sound (the formant spectrum).
What it actually does is change the pitch of a segment up or down by re-sampling. However, since the pitch you
hear is independent of the actual pitch of the graintable (see above), pitch-shifting a segment instead means that
more or less of the segment waveform will be played back, resulting in a change of harmonic content and timbre.

277 MALSTRÖM SYNTHESIZER

 About motion patterns
Each graintable has a predefined motion pattern and a default motion speed.
When a graintable is looped (i.e. if the Motion knob is not set all the way to the left), it follows one of two possible mo-
tion patterns:
D Forward
This motion pattern plays the graintable from the beginning to the end, and then repeats it.
D Forward - Backward
This motion pattern plays the graintable from the beginning to the end, then from the end to the beginning and
then repeats it.
The motion speed can be changed with the Motion knob, as described above, but it is not possible to alter the motion
pattern of a graintable.

The amplitude envelopes

Each oscillator features a standard ADSR (Attack, Decay, Sustain, Release) envelope generator, and a Level control.
These are used for controlling the volume of the oscillator. One thing that makes the Malström different from many
other synths though, is the fact that the amplitude envelopes are placed before the filter and routing sections in the
signal path.
The amplitude envelopes control how the volume of a sound should change from the moment you strike a key on
your keyboard to the moment that you release it again.

Vol
The Volume knobs set the volume level out from each oscillator.
! For an overall description of the general envelope parameters (Attack, Decay, Sustain, Release), please refer
to the Subtractor chapter.

278 MALSTRÖM SYNTHESIZER

 The Modulator section

The Malström features two Modulators (mod:A and mod:B) These are in fact another type of oscillators, called LFOs
(Low Frequency Oscillators). They each generate a waveform and a frequency, much like osc:A and osc:B. However,
there are a couple of important differences:
• Mod:A and mod:B do not generate sound. They are instead used for modulating various parameters to change
the character of the sound.
• They only generate waveforms of low frequency.
Furthermore, both modulators are tempo syncable and possible to use in one shot mode, in which case they will ac-
tually work like envelopes.

Modulator parameters
The two Modulators have a few controls in common, but there are also some differences. Both the common param-
eters and the ones that are unique for each Modulator (the destinations) are described below.
D To activate/deactivate a Modulator, click the On/Off button in the top left corner.
When a Modulator is activated, the button is lit.

An activated Modulator.

Curve
This lets you select a waveform for modulating parameters. Use the spin controls to the right of the display to cycle
through the available waveforms. Some of these waveforms are especially suited for use with the Modulator in one
shot mode (see below).

Rate
This knob controls the frequency of the Modulator. For a faster modulation rate, turn the knob to the right.
The Rate knob is also used for setting the time division when synchronizing the Modulator to the song tempo (see
below).

279 MALSTRÖM SYNTHESIZER

 One Shot
To put the Modulator into one shot mode, click this button so that it is lit.
Normally, the Modulators will repeat the selected waveforms over and over again, at the set rate. However, when one
shot mode is activated and you play a note, the Modulator will play the selected waveform only once (at the set rate)
and then stop. In other words, it will effectively be turned into an envelope generator!
Note that even though all waveforms can be used with interesting results, some waveforms are explicitly well suited
for use in one shot mode. For example, try using the waveform with just one long, gently sloping curve.

Sync
Clicking this button so that it is lit synchronizes the Modulator to the song tempo, in one of 16 possible time divisions.
! When sync is activated, the Rate knob is used for selecting the desired time division. Turn the Rate knob and
observe the tool tip for an indication of the time division.

A/B selector

This switch is used for deciding which oscillator and/or filter the Modulator should modulate - A, B or both. With the
switch in the middle position, both A and B will be modulated.

Destinations
The following knobs are used for determining what each of the two modulators should modulate.
• Note that these knobs are bi-polar, which means that if a knob is in the middle position, no modulation is ap-
plied. If you turn a knob either to the left or to the right, an increasing amount of modulation is applied to the
parameter. The difference is that if you turn a knob to the left, the waveform of the modulator is inverted.

Mod:A

Mod:A can modulate the following parameters of either oscillator:

D Pitch
Use this if you want Mod:A to offset the pitch of osc:A, osc:B, or both (see “Setting oscillator frequency”).
D Index
Use this if you want Mod:A to offset the index start position of osc:A, osc:B, or both (see “Controlling playback of
the graintable”).
D Shift
Use this to have Mod:A affect the harmonic content of osc:A, osc:B, or both (see “Controlling playback of the
graintable”).

280 MALSTRÖM SYNTHESIZER

 Mod:B

Mod:B can modulate the following parameters of either oscillator:

D Motion
Use this if you want Mod:B to affect the motion speed of osc:A, osc:B, or both (see “Controlling playback of the
graintable”).
D Vol
Use this if you want Mod:B to change the output level of osc:A, osc:B, or both (see “Vol”).
D Filter
Use this if you want Mod:B to offset the cutoff frequency of filter:A, filter:B, or both (see “Filter controls”).
D Mod:A
Use this if you want Mod:B to change the total amount of modulation from Mod:A.

The Filter section

The filter section lets you further shape the overall character of the sound. Contained herein are two multimode fil-
ters, a filter envelope and a waveshaper.

281 MALSTRÖM SYNTHESIZER

 The Filters

Both filter:A and filter:B have the exact same parameters, all of which are described below.
D To activate/deactivate a filter, click the On/Off button in the top left corner.
When a filter is activated, the button is lit.

An activated filter.

Filter types
To select a filter type, either click the Mode button in the bottom left corner or click directly on the desired filter name
so that it lights up in yellow:
• LP 12 (12 dB lowpass)
Lowpass filters let low frequencies through and cut off high frequencies. This filter type has a roll-off curve of
12dB/Octave.
• BP 12 (12 dB bandpass)
Bandpass filters cut both high and low frequencies, leaving the frequency band in between unaffected. Each slope
in this filter type has a 12 dB/Octave roll-off.

282 MALSTRÖM SYNTHESIZER

 • Comb + & Comb –
Comb filters are basically delays with very short delay times with adjustable feedback (in Reason controlled with
the Resonance knob). A comb filter causes resonating peaks at certain frequencies.
The difference between “+” and “–” is in the position of the peaks, in the spectrum. The main audible difference is
that the “–”-version causes a bass cut.
The Resonance parameter in both cases controls the shape and size of the peaks.

Comb + Low Resonance. Comb + High Resonance.

Comb – Low Resonance. Comb – High resonance.

• AM
AM (Amplitude Modulation) is often referred to as Ring Modulation. A Ring Modulator works by multiplying two
signals together. In the case of Malström, the filter produces a sine wave which is multiplied with the signal from
osc:A or osc:B. Resonance controls the mix between the clean and modulated signals. The Ring Modulated output
will then contain added frequencies which are generated by the sum of, and the difference between the two sig-
nals. This can be used for creating complex, non-harmonic sounds.

Filter controls
Each filter contains the following four controls:
• Kbd (keyboard tracking)
By clicking this button so that it is lit, you activate keyboard tracking. If keyboard tracking is activated, the fre-
quency of the filter will change according to the notes you play on your keyboard. That is, if you play notes higher
up on the keyboard, the filter frequency will increase and vice versa. If keyboard tracking is deactivated, the filter
frequency will remain at a fixed value regardless of where on the keyboard you play.

283 MALSTRÖM SYNTHESIZER

 • Env (envelope)
If you click on this button so that it is lit, the cutoff frequency (see below) will be modulated by the filter envelope.
If you leave this deactivated, the Filter Envelope will have no effect.
• Freq (frequency)
The function of this parameter depends on which filter type you have selected:
With all filter types except AM, it is used for setting the cutoff frequency of the filter. In the case of the lowpass fil-
ter for example, the cutoff frequency determines the limit above which high frequencies will be cut off. Frequen-
cies below the cutoff frequency will be allowed to pass through. The farther to the right you turn the knob, the
higher the cutoff frequency will be.
If you have selected AM as filter type, this will instead control the frequency of the signal generated by the filter.
The same control range applies though; the farther to the right you turn the knob the higher the frequency will be.
• Res (resonance)
Again, the function of this parameter depends upon which filter type is selected:
If the selected filter is any other type than AM, it sets the filter characteristic, or quality. For the lowpass filter for
example, raising the filter Res value will emphasize the frequencies around the set filter frequency. This generally
produces a thinner sound, but with a sharper, more pronounced filter frequency “sweep”. The higher the filter Res
value, the more resonant the sound becomes until it produces a whistling or ringing sound. If you set a high value
for the Res parameter and then vary the filter frequency, this will produce a very distinct sweep, with the ringing
sound being very evident at certain frequencies.
In the case of the AM filter type though, this control instead regulates the balance between the original signal and
the signal resulting from amplitude modulation. The farther to the right you turn the knob, the more dominant the
AM signal will be.

The Filter Envelope

This is a standard ADSR envelope with two additional controls; inv and amt. The filter envelope is common for both
filter:A and filter:B, and controls how the filter frequency should change over time.

Inv (inverse)
This button toggles inversion of the envelope on and off. The Decay segment of the envelope will for instance nor-
mally lower the frequency, but if the envelope is inverted it will instead raise the frequency.

Amt (amount)
This controls to which extent the filter envelope affects the filters, or rather - the set filter cutoff frequencies. For ex-
ample; if the cutoff frequency is set to a certain value, the filter will already be opened by this amount when you hit a
key on your keyboard. The amount setting then controls how much more the filter will open from that point. Turn the
knob to the right to increase the value.
! For an overall description of the general envelope parameters (Attack, Decay, Sustain, Release), please refer
to the Subtractor chapter.

284 MALSTRÖM SYNTHESIZER

 The Shaper

Before filter:A is an optional waveshaper. Waveshaping is a synthesis method for transforming sounds by altering the
waveform shape, thereby creating a complex, rich sound. Or, if that’s more to your taste, truncating and distorting the
sound to lo-fi heaven!
A guitar distortion box could be viewed as a type of waveshaper for example. An unamplified electric guitar produces
a sound with fairly pure harmonic content, which is then amplified and transformed by the distortion box.
D To activate/deactivate the Shaper, click the On/Off button in the top left corner.
When the Shaper is activated, the button is lit.

The Shaper activated

Mode
You can select one of five different modes for shaping the sound, each with its own characteristics.
To select a mode, either click the Mode button in the bottom left corner or click directly on the desired mode name
so that it lights up in yellow.
• Sine
This produces a round, smooth sound.
• Saturate
This gives a lush, rich character to the sound.
• Clip
This introduces clipping - digital distortion - to the signal.
• Quant
This lets you truncate the signal by bit-reduction, thus making it possible to achieve that noisy, characteristic 8 bit
sound for example.
• Noise
This is actually not strictly a shaper function. Instead it multiplies the sound with noise.

285 MALSTRÖM SYNTHESIZER

 Output Signal

Sine
Saturate
Clip
Quant

Input Signal

Amt (amount)
This controls the amount of shaping applied. By turning the knob to the right you increase the effect.

286 MALSTRÖM SYNTHESIZER

 Routing
The Malström puts you in total control of how the signal should be routed from the oscillators, through the filters and
on to the outputs. Below is first a general description of the routing options, followed by examples of how to route the
signal in order to achieve a certain result.
D Click on a button so that it is lit, to route the signal correspondingly.
See below for descriptions.
If this button is lit, the signal from osc:A is routed to filter:A via the shaper. If neither this nor the other routing
button from osc:A (to filter:B) is lit, the signal will go straight to the outputs.

If this button is lit, the signal from osc:A is routed

to filter:B. If neither this nor the other routing
button from osc:A (to filter:A/shaper) is lit, the
signal from osc:A will go straight to the outputs.

If this button is lit, the signal from osc:B is

routed to filter:B. If this is not lit, the signal from
osc:B will go straight to the outputs.

If this button is lit, the signal from filter:B is routed to filter:A via the shaper. The signal from
filter:B can originate from either osc:A, osc:B or both. If this is not lit, the signal from filter:B
will go straight to the outputs.

! Note that the result depends both on the routing buttons and on whether the filters and shaper are activated or
not!

287 MALSTRÖM SYNTHESIZER

 Routing examples
One or both oscillators without filters

With this configuration, the signals from the oscillators will bypass the filters and the shaper and go directly to the re-
spective output. Using both oscillators allows you to use the Spread parameter to create a true stereo sound.

One or both oscillators to one filter only

Both oscillators routed to filter:B only. Both oscillators routed to filter:A only.
With these configurations, the signal from osc:A and/or osc:B will go to either filter:A or filter:B and then to the out-
puts. This is essentially a mono configuration and hence Spread should probably be set to “0”.

288 MALSTRÖM SYNTHESIZER

 Both oscillators to one filter each

With this configuration, the signals from osc:A and osc:B will go to filter:A and filter:B respectively, and then to the
outputs.
Again, this configuration allows you to work in true stereo.

Oscillator A to both filters in parallel

With this configuration, the signal from osc:A will go to both filter:A and filter:B, with the filters in parallel.
! This configuration is only possible with osc:A. Osc:B can be routed to both filters as well, but only in series
(see below).

289 MALSTRÖM SYNTHESIZER

 One or both oscillators with both filters in series

Osc:A routed through both filters in series. Osc:B routed through both filters in series.
With these configurations, the signal from osc:A and/or osc:B will go to both filter:A and filter:B, with the filters in se-
ries (one after the other).

Adding the shaper

The signal from one or both oscillators can also be routed to the shaper. The signal will then pass through the shaper
to the outputs, with or without also passing through the filters.

In the left figure, the signal from osc:A is routed to the shaper and then directly to the outputs. In the right figure, the signal from osc:B is
routed to filter:B, then to the shaper and then to filter:A.

290 MALSTRÖM SYNTHESIZER

 The output controls

These two parameters control the output from the Malström in the following way:

Volume
This knob controls the master volume out from the Malström.

Spread
This controls the stereo pan-width of the outputs from Osc:A/B and Filter:A/B respectively. The farther to the right
you turn the knob, the wider the stereo image will be. In other words, the signals will be panned further apart to the
left and right.
! If you are only using one output (A or B), it is strongly recommended that you set Spread to “0”.

The play controls

To the far left on the Malström’s “control panel” are various parameters that are affected by how you play, and lets you
apply modulation by MIDI controls. The following is a description of these controls.

291 MALSTRÖM SYNTHESIZER

 Polyphony - setting the number of voices

This lets you set the polyphony for the Malström. Polyphony is the number of voices it can play simultaneously. The
maximum number is 16 and the minimum is 1, in which case the Malström will be monophonic.
! The number of voices you can play depends of course on the capacity of your computer. Even though the max-
imum number is 16 it doesn’t necessarily mean that your system is capable of using that many voices. Also
note that voices do not consume CPU capacity unless they are really “used”. That is, if you are using a patch
that plays two voices but have polyphony set to four, the two “unused” voices do not consume any of your sys-
tem resources.

Porta (portamento)

This is used for controlling portamento. This is a parameter that makes the pitch glide between the notes you play,
rather than changing the pitch instantly as soon as you hit a key on your keyboard. By turning this knob you set how
long it should take for the pitch to glide from one note to the next as you play them.
With the knob turned all the way to the left, portamento is disabled.

Legato

By clicking this button you activate/deactivate Legato. Legato in Malström is unique in that it allows you to control
whether the sound is monophonic or polyphonic by using your playing style:
D If you play legato (hold down a key and then press another key without releasing the previous), the sound is
monophonic.
Also note that the pitch changes, but the envelopes do not start over. That is, there will be no new “attack”.
D If you play non-legato (separated notes), with polyphony set to more voices than 1, each note will decay sepa-
rately (polyphonic).
This will be most apparent with longer release times.

292 MALSTRÖM SYNTHESIZER

 The Pitch Bend and Modulation wheels

• The Pitch Bend wheel is used for bending the pitch of notes, much like bending the strings on a guitar or other
string instrument.
• The Modulation wheel can be used for applying modulation while you are playing.
Virtually all MIDI keyboards have Pitch Bend and Modulation controls. The Malström does not only feature the set-
tings for how incoming MIDI Pitch Bend and Modulation wheel messages should affect the sound, but also two func-
tional wheels that can be used for applying real time modulation and pitch bend if you don’t have these controllers on
your keyboard, or if you aren’t using a keyboard at all. The wheels on the Malström also mirror the movements of the
wheels on your MIDI keyboard.

Pitch Bend Range

The Range parameter sets the maximum amount of pitch bend, i.e. how much it is possible to change the pitch by
turning the wheel fully up or down. The maximum range is 24 semitones (2 Octaves). You change the value by click-
ing the spin controls to the right of the display.

The Velocity controls

Velocity is used for controlling various parameters according to how hard or soft you play notes on your keyboard. A
typical use of velocity control is to make sounds brighter and louder if you strike a key harder. By using the knobs in
this section, you can control how much the various parameters will be affected by velocity.
! All of the velocity control knobs are bi-polar, which means that the amount can be set to either positive or neg-
ative values, while keeping the knobs in the center position means that no velocity control is applied.
The following parameters can be velocity controlled:
• Lvl:A
This lets you velocity control the output level of osc:A.
• Lvl:B
This lets you velocity control the output level of osc:B.

293 MALSTRÖM SYNTHESIZER

 • F.env
This sets velocity control for the Filter Envelope Amount parameter. Positive values will increase the envelope
amount the harder you play, and negative values will decrease the amount.
• Atk (attack)
This sets velocity control for the Amp Envelope Attack parameter of osc:A and/or osc:B. Positive values will in-
crease the Attack time the harder you play, and negative values will decrease it.
• Shift
This lets you velocity control the Shift parameter of osc:A and/or osc:B.
• Mod
This lets you velocity control all modulation amounts of mod:A and/or mod:B.
! Note that you can set the last three parameters (Atk, Shift and Mod) to be velocity controlled for either or both
of oscillator/modulator A and B. This is done with the A/B selector switch.

The Modulation wheel controls

The Modulation wheel can be set to control a number of parameters. You can set positive or negative values, just like
in the Velocity Control section (see above).
The following parameters can be affected by the modulation wheel:
• Index
This sets modulation wheel control of the currently active graintable’s index (see “Controlling playback of the
graintable”) for osc:A and/or osc:B. Positive values will move the index position forwards if the modulation wheel
is pushed forward. Negative values will move it backwards.
• Shift
This sets modulation wheel control of the Shift parameter of osc:A and/or osc:B (see “Controlling playback of the
graintable”).
• Filter
This sets modulation wheel control of the Filter Frequency
parameter (see “Filter controls”). Positive values will raise the frequency if the
wheel is pushed forward and negative values will lower the frequency.
• Mod
This sets modulation wheel control of the total amount of modulation from mod:A and/or mod:B. Positive values
will increase the settings if the wheel is pushed forward and negative values will decrease the settings.
! You can set whether these parameters on either or both oscillator/modulator/filter A and B will be affected by
the modulation wheel. This is done with the A/B selector switch.

294 MALSTRÖM SYNTHESIZER

 Connections

Flipping the Malström around reveals a wide array of connection possibilities. Most of these are CV/Gate related. Us-
ing CV/Gate is described in the chapter “Routing Audio and CV”.

Audio Output
These are the Malström’s audio outputs. When you create a new Malström device, they are auto-routed to the first
available channel on the audio mixer:
• Shaper/Filter:A (left) & Filter:B (right)
These are the main stereo outputs. Each of the two filters are connected to a separate output, and by connecting
both, you can have stereo output. Whether the output really will be in stereo however, is determined by the routing
and the Spread parameter. See “Routing” for details about this.
• Osc:A & osc:B
These make it possible to output the sound directly after the Amp Envelope of each oscillator, bypassing the filter
section. Connecting one or both of these to a channel on the audio mixer will break the Malström’s internal signal
chain. That is, it is not possible to process the sound by using the filters and the shaper of the Malström. the sound
instead goes directly to the mixer.
q Note also that you can connect the outputs Osc:A & Osc:B to the Audio Inputs on the Malström for some inter-
esting effects - see “Routing external audio to the filters”.

Audio Input
• Shaper/Filter:A
• Filter:B
These inputs let you connect either other audio sources, or the Malström’s own internal signal directly to the filters
and the shaper - see “Routing external audio to the filters”.

Sequencer Control
The Sequencer Control CV and Gate inputs allow you to play the Malström from another CV/Gate device (typically a
Matrix or a Redrum). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note
on/off along with velocity.
! For best results, you should use the Sequencer Control inputs with monophonic sounds.

295 MALSTRÖM SYNTHESIZER

 Gate Input
These inputs can receive a CV signal to trigger the following envelopes:
• Amp Envelope
• Filter Envelope
! Note that connecting to these inputs will override the normal triggering of the envelopes. For example, if you
connected a Modulation output to the Gate Amp in-put, you would not trigger the amp envelope by playing
notes, as this is now controlled by the Modulator. In addition you would only hear the Modulator triggering the
envelope for the notes that you hold down.

Modulation Input
These control voltage (CV) inputs (with associated voltage trim pots and A/B selector switches), can modulate vari-
ous Malström parameters from other devices, or from the modulation outputs of the same Malström device. These in-
puts can control the following parameters:
• Oscillator Pitch
• Filter Frequency
• Oscillator Index offset
• Oscillator Shift
• Amp Level
• Mod Amount
• Mod Wheel

Modulation Output
The Modulation outputs can be used to voltage control other devices, or other parameters in the same Malström de-
vice.
The Modulation Outputs are:
• Mod:A
• Mod:B
• Filter Envelope

296 MALSTRÖM SYNTHESIZER

 Routing external audio to the filters

The audio inputs on the back of the Malström allows you to connect any audio signal to the filters and Shaper.
To use this feature, it’s important to understand the following background:
Normally the Malström behaves like any regular polyphonic synthesizer, in that each voice has its own filter. The filter
settings are the same, but each filter envelope is triggered individually when you play a note.
However, when you connect a signal to the audio inputs, it is routed to an “extra” filter. The envelope for this filter is
triggered each time any of the other filter envelopes is triggered. In other words, the “extra” filter envelope is triggered
each time you play a note on the Malström.
There are two different uses for the audio inputs:

Connecting an external signal source

Connecting an audio signal from another device in the rack to the audio input allows you to process the signal
through the filters and/or Shaper of the Malström. The processed signal will then be mixed with the Malström’s “own”
voices (if activated) and sent to the outputs.
The result depends on the following:
• To which jack you connect the signal.
• Whether the filters and/or Shaper are activated on the front panel.
• The routing button for filter:B.
If this is activated and you connect a signal to the Filter:B input, the signal will be processed in filter:B and then
sent to the Shaper and filter:A (just as when routing Malström’s own oscillators on the front panel).
Note again that the filter envelope is triggered by all voices. To make use of the filter envelope, you either need to
play the Malström or use gate signals to trigger it or the filter envelope, separately.

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 Connecting the signals from the Malström itself

If you connect one or both oscillator outputs to the audio input(s), the internal signal path from the oscillators to the
filters is broken. In other words, no signals will pass internally from the oscillators to the filters, and the three routing
buttons for the oscillators are ignored.
This may seem pointless at first, but there are several uses for this:
D When you play the Malström in this mode, the filter envelope will be triggered for each note you play, affecting
all sounding notes.
This is due to the monophonic “extra” filter described above. On older synthesizers, this feature is called “Multiple
triggering”.
D Since all notes you play are mixed before being sent into the filter, the result of using the Shaper will be totally
different (if you play more than one note at a time).
This is similar to playing a guitar chord through a distortion effect, for example.
D You can patch in external effects between the oscillators and the filters.
Just connect an oscillator output to the input of the effect device, and the effect output to the Malströms’s audio
input.
q You can use combinations of connections and routing. You could for instance connect an external audio signal
to one of the inputs, one of the Malström’s oscillators to the other input and then use the routing options on
the front panel for the other oscillator. All of these signals will then be mixed and sent to the Malström’s main
outputs.

298 MALSTRÖM SYNTHESIZER

Chapter 18
MIDI Out Device
 Introduction

The MIDI Out Device is designed for routing MIDI out of the Reason Rack Plugin instance to other tracks/destina-
tions in your main DAW. A typical scenario would be to route MIDI from a Player device in Reason Rack Plugin to an-
other instrument plugin in your song/project.
The MIDI Out Device does not produce any sound of its own; it only directs MIDI from the Reason Rack Plugin in-
stance to a selected MIDI Channel.

Using the MIDI Out Device

Setting up for MIDI controlling an external track/plugin
In this example we will route the MIDI from a Dual Arpeggio Player device via a MIDI Out Device.
1. Drag a MIDI Out Device from the Instruments palette in the Browser and drop in the rack.
The MIDI Out Device is created in the rack.
2. Click the Players palette and drag a Dual Arpeggio Player and drop above the MIDI Out Device:

The Dual Arpeggio Player is automatically attached to the MIDI Out Device.
3. Create a MIDI track in the DAW sequencer.
4. Select Reason Rack Plugin as MIDI Input port for that track (refer to the DAW manual).

300 MIDI OUT DEVICE

 ! With most DAWs, a Reason Rack Plugin instance can produce both audio and MIDI at the same time (i.e. a
Reason Rack Plugin instrument can also send out MIDI). However, with DAWs using the AU plugin format (e.g.
Logic), you need to add Reason Rack Plugin in a special MIDI FX slot for it to output MIDI. That instance will be
a MIDI effect only and won't output audio.
! If you get problems with “hanging” notes, click the Panic button to send out an “All Notes Off”.

q To route MIDI from other Player devices in the Reason Rack Plugin instance, simply create another MIDI Out
Device and attach another Player to it. Then, select a different MIDI Channel on the MIDI Out Device.
• You don’t have to attach a Player to the MIDI Out Device - you could use the MIDI Out Device just for “through-
put” of the MIDI from the Reason Rack Plugin instance.
In these situations it doesn’t matter where in the rack you place the MIDI Out Device.

About recording the Player MIDI from the MIDI Out Device
If you want to record the MIDI from the Player, you will have to do that in real-time on the destination track in your
host DAW, see “Getting the Player MIDI output onto a track in your DAW”.

Modulating MIDI Controllers from CV signals

The MIDI Out Device features eight CV inputs for routing modulation signals from the Reason Rack Plugin instance
to MIDI CC# of your choice. These MIDI CC# changes are then transmitted on the selected MIDI Channel of the
MIDI Out Device.
1. Flip the rack around and connect some modulation sources to the desired CV inputs:

2. Flip the rack back to the front and click the On button to activate the CV IN section:

CV signals routed to any of the four CV IN pairs are indicated by lit LEDs:

3. Click to select which CV IN pair to edit:

301 MIDI OUT DEVICE

 4. Drag up/down to select the desired MIDI CC# to route the CV modulation signals to:

5. Turn the Scale knobs to change the modulation range, from static (0-0) to full (0-127):

6. Turn the Offset knobs to change the modulation offset (0-127):

The modulation level(s) are shown in the CC Output displays:

7. Repeat steps 3-6 to assign and set up the other CV IN pairs.

• CV can be bipolar (have negative or positive values) but MIDI CC values are always positive. Any negative CV
values will be truncated to zero when converted to MIDI CC. To preserve the shape of e.g. a modulation LFO,
you can use the Scale and Offset controls to convert the bipolar CV (going between -127 and +127) to a MIDI
CC going between 0 and 127. In that case, set both Scale and Offset halfway up.
• You can also use the Offset knobs for manually setting MIDI CC values without anything connected to the CV
inputs.
• The CV Input ON button enables CV inputs. Turn this off if you want to make manual settings, select MIDI CCs
etc, without the values being modulated by CV.

Connections

Sequencer Control
The Sequencer Control CV In and Gate In inputs allow you to play the MIDI Out Device from another CV/Gate device
(typically a Matrix or an RPG-8). The signal to the CV In controls the note pitch, while the signal to the Gate In delivers
note on/off along with velocity. There are also inputs for modulating the Pitch Bend and ModWheel parameters.

CV In to MIDI CC Out
These eight CV inputs can be used for modulating the desired MIDI CC#. The affected MIDI CC#s are defined on the
front panel, see “Modulating MIDI Controllers from CV signals”.

302 MIDI OUT DEVICE

Chapter 19
Mimic Creative Sampler
 Introduction

The Mimic Creative Sampler is a powerful yet very straight-forward sampler, tailor-made for quick and easy triggering,
chopping and manipulation of samples. It features eight sample slots, where each slot can hold one sample. Each slot
also has its own complete synth parameters setup, with pitch controls, filter, envelopes, LFO and effects.
You could either load a sample from your computer or sample straight into Mimic (Reason stand-alone only). You can
then select various sample playback modes and high-quality stretch algorithms to manipulate and process the audio.
You could also use Mimic as a traditional sampler and just play back samples in a regular “tape-style” fashion.
Mimic also features a great-sounding “lo-fi style” multi-effect to spice up your sounds even more.
Don’t forget to check out the Mimic video tutorial here!

304 MIMIC CREATIVE SAMPLER

 Panel overview
The Mimic front panel contains the following sections:

1 2 3
4 5

8 9 10 11

12 13 14

The Mimic front panel sections.

• 1. MIDI Note On LED.
• 2. Patch selector (for browsing, loading and saving patches).
• 3. Master Volume controls.
• 4. Playback Mode section.
• 5. Slot selector.
• 6. Sample section (for sample loading/sampling and sample playback functions).
• 7. Playback Start Position, Speed, Stretch and Slices sections.
• 8. Performance and “play” controls.
• 9. Pitch section.
• 10. Filter section.
• 11. Envelopes section.
• 12. LFO section.
• 13. Amp section.
• 14. Compressor, Effects and EQ sections.

305 MIMIC CREATIVE SAMPLER

 Signal flow
Slots (x8)

Pitch
or
(x1)
Slice
Separate Out (x8)
Master Out
(x8) (x8)
Master Volume
Multi Slot
or
Multi Pitch (x8)

Send 2 (x8)

Send 1 (x8)
Send 1 Stereo Out Send 2 Stereo Out

Mimic signal flowchart.

• Mimic features eight Slots, where each Slot holds a complete configuration of sample parameters and panel
parameter settings.
Each Slot features a Sample section, where you can load your sample - or sample your own (Reason stand-alone
only). You can then select if you want to play back the sample as Slices or Pitched (chromatically). You can also
choose among various sample Stretch algorithms. The signal is then routed through a complete configuration of
synth parameters, such as Pitch, Filter, Amp and Effects.
• Depending on the selected Slot Mode, the signals from the Slots are be routed a bit differently.
In Pitch Mode and Slice Mode, only one single Slot can be played back at a time.
In Multi Slot Mode and Multi Pitch Mode you can play back up to eight Slots simultaneously.
• The synth sections in Mimic’s eight Slots are described in “Panel reference”.

306 MIMIC CREATIVE SAMPLER

 Playing and using Mimic
Loading and saving patches

A Mimic patch contains the parameter settings for all used Slots, i.e. up to eight complete parameter setups. Loading
and saving patches is done in the same way as with any other internal Reason device, see “Loading patches” and
“Saving patches” for details.
! Like with the other sampler devices in Reason, the patch does not include the actual sample - only a
reference to it. Therefore, the sample has to be stored separately (self-contained with the song, or already on
disk or in a ReFill on your computer).

Global output controls

Master Volume
This is the main stereo output volume control and controls the volume of all eight Slots together.

Slot Mode

Mimic has four Slot Modes, which defines how the sample of each Slot should play back - and also if you can use one
or several Slots simultaneously:

Pitch Mode

In Pitch Mode you select one of the eight Slots for melodic playback (playing the sample from the start to the end lo-
cators). The sample can then be played chromatically pitched over the entire keyboard range.

307 MIMIC CREATIVE SAMPLER

 Slice Mode

C1 C2 C3 C4 C5 C6

In Slice Mode you select one of the eight Slots for playback, and slice a (longer) sample, manually and/or automati-
cally. The slices are then triggered from the keyboard (chromatically, with the leftmost slice playing back from note
C1) without affecting the original pitch of the sample. See “Slices” for more details.

Multi Slot Mode

In Multi Slot Mode different keys (C-G in each octave) triggers Slots 1-8 (from the start to the end locators), drum-
machine style - without affecting the sample pitches. This is displayed in the Note/range indicator as follows (the
currently selected Slot is high-lighted on the keyboard - the others are dark blue):

Slot 2 Slot 2 Slot 2 Slot 2 Slot 2 Slot 2

Slot 4 Slot 4 Slot 4 Slot 4 Slot 4 Slot 4

Slot 3 Slot 3 Slot 3 Slot 3 Slot 3 Slot 3

Slot 1 Slot 1 Slot 1 Slot 1 Slot 1 Slot 1 Slot 1
Multi Slot Mode with samples loaded in Slots 1-4.
q Multi Slot Mode works great with pad controllers.

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 Multi Pitch Mode

C1 C2 C3 C4 C5 C6

C1 C2 C3 C4 C5 C6

C1 C2 C3 C4 C5 C6

C1 C2 C3 C4 C5 C6

One MIDI keyboard playing four Slots simultaneously (layered)

Multi Pitch Mode with samples loaded in Slots 1-4, played back in a layered fashion.
In Multi Pitch Mode you can play back up to eight samples (one sample per Slot) simultaneously, for melodic play-
back (playing the sample from the start to the end locators). You can set different keyboard ranges (overlapping and/
or adjacent) for the eight Slots and then play back the samples chromatically pitched. You can set the desired key-
board range in the Note/range indicator as follows (the currently selected Slot is high-lighted on the keyboard - the
others are dark blue):

Setting the key range for the currently selected Slot.

1. Click the desired Slot button (see “Slot Select” below) to select it.
2. Drag the left and right note range markers to set the desired note range.
The minimum “range” for a Slot is one note.
3. Click another Slot button and set a new desired note range.
q Note that Slots can have overlapping note ranges (also completely overlapping so that you could play all eight
Slots together in a layered fashion).

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 Slot Select

Mimic has eight Slots, which can hold one sample each. Each Slot also has a complete parameter setup on the Mimic
front panel, so each slot can have its own unique parameter configuration.
D Click the desired Slot Select button to bring up all parameters for that slot, including the sample and its set-
tings.
! Note that in Pitch Mode and Slice Mode (see above), only one single Slot can play back its sample at a time (as
indicated by the grayed out waveforms below the other Slot Select buttons).

Performance and “play” controls

Each of the eight Slots has their own individual set of performance/play controls on the front panel:

Porta
Portamento makes note pitches glide from previous notes to new ones, at the time set with the Time knob. Porta-
mento can be used in all Key modes (see above).
• When On in Poly Key Mode (see below), the pitches will glide from any of the available voices.
The results will be unpredictable since there is no way of controlling from which note(s) the glide(s) will com-
mence. The effect is very nice, though.
• When On in Mono Retrig or Mono Legato Key Mode (see below), the pitch will glide between consecutive
notes.
• In Auto mode, the pitch will glide between consecutive monophonic notes only when you play legato. If you
have selected Poly Key Mode (see below), Auto will have no effect at all.
If you release the previous key before hitting the new key, there will be no portamento effect.

Key Mode
Here you choose how Mimic should respond to MIDI Note data:
• Poly
Select this if you want to play Mimic polyphonically.
• Mono Retrig
Select this if you want to play Mimic in monophonic mode and always retrigger the envelopes as soon as you play
a new note.

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 • Mono Legato
The Mono Legato mode is also monophonic. However, if you play a new note without having released the previous
one, the envelopes and sample playback position won’t start over.
q Also see the description of the “Global Position”, which allows new notes to continue playing at the current
sample playback position (and not restart playback).

Pitch Range
D Set the desired Pitch Bend range for the “Pitch Bend” wheel with the up/down buttons, or by click-holding on
the display and dragging up/down.
Range: +/-24 semitones (+/-2 octaves) in steps of +/-1 semitone.

Pitch Bend
The Pitch Bend wheel can be used for bending note pitches up and down. Mimic also responds to Pitch Bend MIDI
data from a connected MIDI master keyboard. You set the desired Pitch Bend Range with the “Pitch Range” control
above the Pitch Bend wheel.
! Note that if you have selected the Tape Stretch algorithm (see “Tape”), pitch bending the sample will also
affect its playback speed.

Mod Wheel
The Mod Wheel can be used for modulating a number of parameters in the eight Slots of Mimic. Use the Mod Wheel
as a source parameter for the panel parameters that feature modulation source drop-down selectors.

LFO Scale
The LFO Scale knob can be used for scaling the LFO amount with the Mod Wheel - perfect for gradually introducing
Vibrato effects, for example. If the LFO Scale knob is set to 100% the LFO Amount will be 0 when the Mod Wheel
is at 0. See “The LFO section” for more details about the LFO.

311 MIMIC CREATIVE SAMPLER

 Panel reference
The Sample section
Here is where you load/sample the audio for the currently selected Slot. The Waveform display differs somewhat, de-
pending on which Slot Mode you have selected (see “Slot Mode”).
Sample range markers
Root Key section
Start marker indicator End marker indicator

Sample Overview
Slice markers
End
marker
Start
marker Position marker Waveform display

Sample Load/Sampling section Note/range indicator

A sample loaded in a Slot with Slice Mode selected.

Loading samples
D Load a sample using drag & and drop, or by clicking the Browse sample button, or by using the Up/Down but-
tons to scroll and load a sample from the currently selected folder.
D Drag a sample from the browser and drop on a Slot Select button, to load the sample in the desired Slot.
! It’s possible to load/sample stereo signals. However, the waveform will always be displayed as a mono signal,
regardless if it’s mono or stereo.

Root Key
A sample is automatically analyzed for its original pitch at the Sample Start position. The analyzed pitch is displayed
in the Root Key section. If you move the Sample Start marker, the sample is automatically re-analyzed.
D Click the “SET” button to use the analyzed Root Key.
This will automatically place the analyzed Root Key on the correct note on the keyboard, as indicated by an orange
key in the Note/range indicator below the Waveform display.
D Alternatively, set the Root Key manually by dragging up/down in the “Root” and “Tune” (cent) boxes:

! Note that the Root Key function only works in Pitch Mode and Multi Pitch Mode.

312 MIMIC CREATIVE SAMPLER

 Setting the sample range
First you could decide how much of the original sample you want to use - and where in the sample you want to work:
D Zoom and/or scroll in the Sample Overview to define the Sample range you want to work in.

To scroll, click and drag sideways between the dark yellow sample range markers. To zoom, click and drag any of
the sample range markers sideways. The set Sample range is automatically updated and displayed in the wave-
form display.
D To work in the entire Sample range, drag the left Sample range marker all the way to the left, and the right
Sample range marker all the way to the right, in the Sample Overview.

Zooming in the Waveform Display

D Click and hold the mouse button anywhere in the waveform display, then drag down to zoom in, and up to
zoom out in the waveform.
q Dragging sideways in the Waveform Display will scroll left/right.

Setting the sample start and end

D Drag the Start and End marker handles to where you want the sample to begin and stop playing back.
Note that the Start and End markers cannot be set in reversed order. If you want to play the sample backwards,
from the End marker to the Start marker, click the Reverse button (see “Reverse”).
• If you drag and move the Start/End marker up into the Slice Marker area above the waveform, the Start/End
marker will snap to the closest slices. This works in all modes, not just in Slice Mode:

D Alternatively, double click in the Waveform Display to position the Start marker. Hold down [Shift] and double
click to position the End marker.

313 MIMIC CREATIVE SAMPLER

 Previewing samples
D Preview the sample by holding down [Option]/[Alt] and clicking at the desired position in the Waveform Dis-
play.
The sample will play back for as long as the mouse button is depressed.
q It’s possible to automate the sample start and end settings, see “Automating the sample Start and End mark-
ers”.

Loop and Loop Length

Looped region

It’s also possible to loop (Forward Loop) samples - or slices in Slice Mode. The loop always happens at the end of the
sample/slice. In all modes except for Slice Mode the Loop Length is visually indicated with a transparent red region
in the Waveform Display, so you can see exactly where the loop is.
D Enable Loop by clicking the Loop LED.
D Adjust the Loop Length by turning the Length knob, or by holding down [Command]/[Ctrl] and clicking/drag-
ging sideways in the waveform display:

If you are playing back the sample reversed (see “Reverse”) the looped region originates from the Start Marker in-
stead and is displayed in transparent blue.
! In Slice Mode the looped region is not indicated visually in the Waveform Display.

Start Position

The Start Position section contains controls for determining where the sample/slice playback should begin, and in
which direction the sample/slices should be played back.

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 Global Position
D Click the Global Position LED to start the playback of new/additional voices from the global position, i.e. from
where the red Position marker is currently positioned in the Waveform Display:

This function is great for polyphonic rhythmic sounds, where you want to have all the voices synced in time.
If not active, new/additional voices will always start playing back from the blue Sample Start marker.

Reverse
D Click the Reverse LED to have the sample/slices play back backwards, from the end to the start.

Start Position Mod

D Turn the Mod knob to set a modulation amount for the sample/slice playback start position.
Note that the Mod amount is bipolar (-/+) so that you can modulate the start position either earlier or later than the
default start position.
D Select a Modulation source from the drop-down selector, to modulate the sample/slice start position.
! The Start Position Modulation behavior also depends on the “Snap to Slices” setting described below.

Snap to Slices
D Click the Snap to Slices LED to always start the playback from a Slice Marker (and not from somewhere in be-
tween Slice Markers).
! Note that this function is only useful in when modulating the Start Position (see above) in Pitch Mode, Multi
Slot Mode and Multi Pitch Mode, since the Start Position always snaps to slices in Slice Mode (see “Slice
Mode”).

Speed and Speed Mod

The Speed control determines how fast the play position (“playhead”) moves in the waveform.

Speed
D Set the sample playback speed with the Speed knob.
Depending on which Stretch type (see “Stretch”) is currently selected, the sonic result will vary. If you have se-
lected the Tape Stretch type (see “Tape”), the Speed knob also affects the pitch. Note that the Speed can be set
all the way down to 0%, i.e. “stop”. Great for Tape Stop effects in Tape mode and for static playback in other
Stretch types, for example.

Speed Mod
D Modulate the Speed with the Mod knob.

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 D Assign a modulation source from the drop-down selector next to the Mod knob.
For example, selecting an Envelope as a modulation source could cause the playback speed to start fast and grad-
ually slow down when you hold a note - or vice versa.

Stretch

Mimic features five different Stretch types, which can be selected from the drop-down selector. Depending on the
selected Stretch type, there are also some additional controls to modify the sample characteristics.

Tape

This is the good old “tape recorder” type, where speed and pitch are coupled. This means that to achieve a higher
pitch you simply increase the playback speed of the sample/slices - and vice versa.
With the Tape stretch type selected, there is a Loop X-Fade knob present. This controls the crossfade amount when
the Loop function is active for the sample/slices (see “Loop and Loop Length”).
If Loop is off, the Loop X-Fade knob has no effect.

Advanced

This is a high-quality stretch algorithm suitable for most type of polyphonic and complex audio material.
With the Advanced stretch type selected, there is a Preserve Transients button present. Transients are regions in the
sample where the level quickly goes from quiet to loud, for example in percussive hits and other types of “attacks”.
D Click the Preserve Transients button to preserve any transients in the sample/slices.
When off, any transients will be “smeared out” and less prominent, which might be desired in some situations.

Melody

This is the Melody stretch type used for audio in the Reason sequencer, i.e. a high-quality stretch algorithm suitable
for monophonic audio material.
With the Melody stretch type selected, there is a Preserve Transients button present.
D Click the Preserve Transients button to preserve any transients in the sample/slices.
When off, any transients will be “smeared out” and less prominent, which might be desired in some situations.
q The Melody stretch type is well suited if you are using the Loop function (see “Loop and Loop Length”), since
it usually reduces clicks/pops at the loop point.

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 Vocal

This is the Vocal stretch type used for audio in the Reason sequencer, i.e. a high-quality stretch algorithm suitable for
monophonic vocal audio material.
With the Vocal stretch type selected, there are two additional controls present:
D Turn the Formant knob to change the formant of the sample/slices.
Turning this up will be like creating a smaller “body” for the sound, and turning it down will be like creating a larger
body. If you are using a vocal sample, changing the Formant would be like changing the character from “adult” to
“child” like.
D Click the Fixed Pitch button to “auto-tune” the sample/slices to the currently played (note) pitch.
This is really cool for creating processed vocals that you could pitch from the keyboard.

Granular
This is a “vintage” type of digital pitch shift/stretch method, where grains of the sample are being looped and cross-
faded.

The Granular stretch type utilizes playback of a series of snippets of audio data - grains - “extracted” from the sample.
The grains could be of a selectable length and overlap. The grains could then be played back in a number different
ways - with or without crossfades between the grains.

317 MIMIC CREATIVE SAMPLER

 The picture below shows the basic principle of the Granular Stretch type:

Level

Original sample

Time

5 “extracted” grains

Level
The resulting signal is generated by
appending and crossfading the grains.

Time

An example of a signal generated from 5 grains of a sample.

Here is what happens in the example above:
• The original sample at the top is used as base for the granular stretch.
• 5 grains (of the same lengths and the same distances between them) are “extracted” from the original sample.
The distance between the grains is determined by the current sample playback speed. The grains could contain
common audio data in some parts (like in the beginnings and ends in the example above).
• The 5 grains are then placed after one another, partly overlapping each other.
The distance between the grains is determined by the playback rate.
• When the grains are played back, big parts of the grains are played back together (since they are overlapping).
In the example above, there are also crossfades between the grains to make the overlaps smoother.
With the Granular stretch type selected, there are four additional controls present:
D Control the grain lengths with the Length knob.
This sets the lengths of the grains. At 0% you get the shortest grains and the sound could be a little gritty at this
setting. At 100% you get the longest grains.
D Control how much the grains should overlap each other with the Overlap knob.
This sets how much the grains should overlap each other and also the crossfade amount. At 0% you get no over-
lap and almost no crossfade at all. At 100% you get the longest overlaps and also smooth crossfades between the
grains.

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 D Set the playback position deviation with the Jitter knob.
The Jitter function modulates the sample playback position minutely and randomly. The Jitter function can be
great for generating “chorus”-like effects and to make a sound more “alive”, depending on the other settings in the
sound. At 0%, the timing and playback position is completely accurate and at 100% it is completely random.
D Set how much you want the grains to be panned in the stereo panorama with the Spread knob.
0% means the signal will be unaffected and 100% means every other grain will be panned hard left and hard right.
Great for nice and wide stereo effects!

Slices

This section is mainly useful when you are working in Slice Mode (see “Slice Mode”).
q Even though slices are the core of Slice mode, they are available in the other modes too. They can be used for
snapping the Start position (see “Setting the sample start and end”) - either when you drag it manually or
when you modulate or automate it.

Sensitivity
Slices are added automatically at transients according to the Sensitivity knob setting.
D Turn the Sensitivity knob to increase or decrease the number of automatically detected slices.
Automatically generated slice markers are indicated in yellow.
• In Slice Mode the number of available keys changes according to the number of slices in the loop. For exam-
ple, if there are 10 slices in the loop, the first 10 notes (counted upwards from C1) are used. If the loop contains
15 slices, the first 15 notes are used, and so on:

! Note that the maximum number of slices that can be detected and used is 92 (from C1 to G8). If you are using
a very long sample and a high Sensitivity setting, the detected slices might not cover the entire sample - only
the first part. If you want to access slices further into the sample you might therefore have to move the Sample
Start marker (or reduce the Sensitivity setting).

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 Moving, adding and removing Slice markers
D Drag the Slice markers sideways to move them:

D Add slices manually by double clicking in the Slice Marker lane directly above the waveform:

! Any manually moved/added slice markers are automatically indicated in a different color, which also means
they are no longer affected by the Sensitivity knob.
q You can also click on a slice marker handle (without moving it) to deactivate it from the Sensitivity knob.
D Delete slice markers by double clicking the slice marker handles in the Slice Marker lane.

Reset
D Click the Reset button to restore the slices to the ones auto-generated by transient detection/Sensitivity.
Any manually added slice markers will be removed.

Play Thru
D Click the Play Thru LED to force the playback to continue beyond the following slice markers, for as long as the
notes sustain.
When Off, the playback will automatically stop at the next slice marker, even if you have sustaining notes.

Pitch

Semi
D Set the pitch in semitone steps.
Range: +/-24 semitones (+/-2 octaves).
! Note that if you are using the Tape Stretch type (see “Tape”), the pitch settings also affect the playback speed.

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 Tune
D Changes the pitch in steps of 1 cent.
Range: +/- 50 cents (down or up half a semitone).

LFO
D Sets how much the pitch should be affected by the LFO (see “The LFO section”).
Range: +/- 100%.
q By using the LFO Scale function (see “LFO Scale”), you can gradually introduce the LFO modulation amount by
using the Mod Wheel.

Pitch Mod
D Sets how much the pitch should be modulated by the source assigned in the drop-down selector to the right.
Range: +/- 100%.

The Filter section

The Filter section features eight different filter types.

Filter Type selector

D Click the Filter type drop-down selector to choose any of the following filter types:
• LP 24

Amplitude

RESO

Frequency
FREQ
This is a standard 24dB/octave lowpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.

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 • HP 24

Amplitude

RESO

Frequency
FREQ
This is a standard 24dB/octave highpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
• LP 12

Amplitude

RESO

Frequency
FREQ
This is a standard 12dB/octave lowpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
• BP 12

Amplitude

RESO

Frequency
FREQ
This is a standard 12dB/octave bandpass filter. Set the center frequency with the Freq knob and the
resonance amount with the Reso knob. Note that raising the resonance also makes the passband narrower.
• HP 12

Amplitude

RESO

Frequency
FREQ
This is a standard 12dB/octave highpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.

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 • Notch
Q
Amplitude
RESO

Frequency
FREQ
A notch filter (or band reject filter) could be described as the opposite of a bandpass filter. It cuts off frequencies
in a narrow midrange band, letting the frequencies below and above through. Set the center frequency with the
Freq knob and the notch width with the Reso knob. The higher the Resonance, the narrower the notch.
• Comb -

Amplitude

RESO

Frequency
FREQ
This is a comb filter with a positive feedback loop - but without feed forward - ideal for flanger and phaser types
of effects. Set the cutoff frequency with the Freq knob and the resonance amount with the Reso knob. The differ-
ence between “Comb +” (see below) and “Comb –” is in the position of the peaks in the spectrum. The main audi-
ble difference is that the “Comb –” version causes a bass cut.
• Comb +

Amplitude

RESO

Frequency
FREQ
This is a multi notch filter, great for phaser types of effects. Set the cutoff frequency with the Freq knob and the at-
tenuation amount - and consequently the bandwidth - of the notches with the Reso knob. The difference between
“Comb +” and “Comb –” (see above) is in the position of the peaks in the spectrum. The main audible difference is
that the “Comb +” version lets through more bass frequencies.

Freq
D Set the cutoff frequency (for the HP and LP filter types) or the center frequency (for the BP and Notch filter
type).
Range: 37.0 Hz to 16.00 kHz.

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 Reso
D Set the resonance amount.

Drive
D Set the amount of overdrive distortion in the filter.

Kbd
D Set how much you want the filter cutoff/center frequency to track the keyboard.
At 0%, the filter frequency is static regardless where on the keyboard you play. At 100% the filter tracks the key-
board 1:1, i.e. one semitone per note.

Vel
D If you want the Filter Envelope amount to be controlled from keyboard velocity, turn up the Vel knob.

Env
D Set the cutoff/center frequency modulation amount from the Filter Envelope (see “The Filter Envelope and
Amp Envelope sections”).
Range: +/- 100%.

Freq Mod
D Select a modulation source, for modulating the filter cutoff/center frequency, from the drop-down selector.
D Set the desired modulation amount with the Mod knob.
Range: +/- 100%.

The Filter Envelope and Amp Envelope sections

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 The Filter Envelope and Amp Envelope feature standard ADSR envelopes that control the modulation amounts of the
respective destinations (the Filter Freq and the Amp Gain). The picture below shows the various stages of the ADSR
envelope:

Level

Amp
Gain
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the maxi-
mum frequency value (Filter Envelope) or Gain level (Amp Envelope). How long this should take, depends on the At-
tack setting. If the Attack is set to “0”, the maximum Freq/Gain value is reached instantly. If the Attack value is raised,
it will take longer time before the maximum Freq/Gain value is reached.

D(ecay)
After the maximum Freq/Gain value has been reached, the level starts to drop. How long this should take is governed
by the Decay parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to “0”,
the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the volume
gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to drop to
some other value than zero, you raise the Sustain parameter.

S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain to
full level, the Decay setting is of no importance since the frequency/volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain param-
eter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays
there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the maximum Freq/Gain
value, then gradually decreases to finally land to rest on a level somewhere in-between zero and the maximum Fre-
quency/Gain value. Note that Sustain represents a level, whereas the other envelope parameters represent times.

R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the Freq/Gain
to drop back to zero (or to the set Freq value) after you release the key.

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 The LFO section

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO
modulate the pitch of a signal to produce vibrato, but there are countless other applications for LFOs.
The LFO section features one general purpose LFO, which can be assigned to control selectable parameter(s) in
other sections on the front panel.
q By using the LFO Scale function (see “LFO Scale”), you can gradually introduce the LFO modulation amount by
using the Mod Wheel.

Wave
D Select an LFO waveform by clicking the spin controls to the right of the waveform display, or by click-holding
in the display and dragging up or down.
Besides the standard waveforms (sine, triangle, pulse, etc.) there are random, slope and stepped waveforms. The
shape of the waveforms are shown in the display.

Rate
D Set the LFO frequency with the Rate knob.

Key Sync
D Click the Key Sync button to restart the LFO at every new Note On.

Beat Sync
D Click the Beat Sync button to sync the LFO to the main sequencer Tempo.
The Rate parameter now controls the time divisions.

Delay
D Turn the Delay knob to introduce a delay before the LFO modulation kicks in after a note is played.
Turn clockwise for longer delay times.

The Amp section

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 Gain
D Set the desired maximum level for the Amplifier with the Gain knob.
This is the maximum level the envelope will reach after the Attack stage is completed (see “A(ttack)”.).

Vel
D If you want the Gain level to be controlled from keyboard velocity, turn up the Vel knob.

Gain Mod
D Assign a modulation source in the drop-down selector to the right. Then, control the modulation amount with
the Mod knob.
q If you want to create a tremolo effect, select the LFO as Source in the drop-down selector.

Pan
D Set the panning of the output signal from the Amplifier in the stereo panorama.

Pan Mod
D Assign a modulation source in the drop-down selector to the right. Then, control the modulation amount with
the Mod knob.
q Since Pan works individually per voice, you can assign e.g. Keyboard Velocity, an Envelope or “Random” as
source to create cool panning effects.

The Compressor

This is a Compressor, which can be used for compressing the signal and evening out the signal levels.
D Turn the Squeeze knob to set the compression amount.
The red LED above the knob indicates the signal compression.

The Effect section

The Effect section features seven types of distortion/modulation effects, to spice up your sound.

327 MIMIC CREATIVE SAMPLER

 D Select the desired Effect algorithm from the drop-down selector.
“Noise” adds noise to the signal (when a signal is present).
“Reso Noise” adds resonant noise to the signal (when a signal is present).
“Ring Mod” is a ring modulator effect.
“Bitrate” gives the effect of sample rate reduction.
“Lowres” combines sample rate reduction with bit depth reduction.
“Sine Fold” is a sinewave shaping distortion.
“Scream” produces a less bright type of distortion.

Effect Mod
D Set the character of the selected effect algorithm.

Mix
D Set the mix between the dry signal and the effect signal.

The EQ section

This is a Lo Cut and Hi Cut filter, which lets you cut out bass (lo cut) and treble (hi cut) frequencies from the sound.

Lo Cut
D Turn up the Lo Cut knob to cut out bass frequencies from the signal.
Range: 20.0 Hz to 4 kHz.

Hi Cut
D Turn down the Hi Cut knob to cut out treble frequencies from the signal.
Range: 200 Hz to 20 kHz.

The Send section

The Send knobs can be used for tapping the output signal of the Slot to the corresponding FX Send Out jacks on the
rear panel (see “FX Send Out”). You could then route the signals to external effect devices and then further to a sep-
arate mixer/audio channel.
D Control the Send output levels with the corresponding Send knobs.

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 Connections

! Remember that CV connections are NOT stored in the Mimic patches! If you want to store CV connections
between devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Mimic from another CV/Gate device (typically a Matrix
or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/
off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

CV In 1-4
These four assignable control voltage (CV) inputs can be used for modulating parameters on the Mimic front panel,
by selecting “CV In” in the Mod drop-down selector for the desired parameter.

Slot Out 1-8

These outputs are breakout jacks for each individual Slot. Connect these to have the audio from the desired Slot(s)
routed to its own stereo output pair.
! Connecting any of the Slot Outs will automatically remove the corresponding Slot from the Master Audio Outs.

FX Send Out
Here you can route the audio from all eight Slots to external effect devices for further processing. Since there are no
FX Return jacks on Mimic, route the processed signal to a separate mixer/audio channel. You can control the Send
level of each Slot with the respective Send knobs (see “The Send section”).

Audio Output
These are the main audio outputs. When you create a new Mimic device, these outputs are auto-routed to the first
available Mix Channel in the main mixer. If there is no Mix Channel available, a new one will be automatically created.

329 MIMIC CREATIVE SAMPLER

 Tips and Tricks
Optimizing performance/DSP Load
Under some circumstances Mimic could be quite DSP demanding. If you experience any excessive DSP Load or
playback problems, please check the following:
D Try another Stretch Type (see “Stretch”).
Especially the “Advanced” Stretch Type can be quite DSP consuming.
• If you are playing high-pitched notes using a low-pitched sample, the stretch function has to calculate a lot. To
reduce the workload, try loading a higher pitched sample (which has a higher Root Key) and use that instead.
D Try to avoid having a lot of notes decaying at the same time. Try shortening the Release time of the Amp Enve-
lope (see “The Filter Envelope and Amp Envelope sections”) and/or deactivating Play Thru in Slice Mode (see
“Play Thru”).

Creating a “velocity layered” instrument

A neat way to simulate a velocity layered instrument is to sample the same note repeatedly, of let’s say a bass or pi-
ano, and then modulate which of the recorded notes should play back. This will give you a more “live” and organic
sound when you play. In this example we have sampled the same note of a bass guitar eight times in a single record-
ing, played with a little different nuances between the eight picks:

1. Make sure Pitch Mode is selected.

2. Click the “Snap to Slices” LED in the Start Position section.
Do this to ensure that the playback will start from a slice marker (and not from in between slice markers).
3. Select “Velocity” in the Start Position Mod drop-down list and turn up the Start Position Mod amount.
We are now controlling which “slice” should play back from Keyboard Velocity. You can of course choose another
modulation source (e.g. Mod Wheel) if you want to control the start position using the Mod Wheel instead.
q Make sure you set the Amp Envelope “Release” time so that the consecutive “slice” doesn’t play back.

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 Extending the sample “tail” (without looping)
Sometimes it could be useful to extend the tail of a sample (a drum sample, for example), to have it decay longer than
the original sample length. An easy way of doing this is by modulating the “Speed” parameter as follows:

1. Turn down the “Speed” knob close to minimum.

2. Turn up the “Speed Mod” knob and select “Filt Env” as Speed Mod source.
3. Select the “Advanced” stretch type in the Stretch section.
4. Raise the Filter Envelope (D)ecay slider to the desired value.
When you play and hold a key down, the Speed will quickly raise and then slow down in the sample tail, making the
playback time extend.

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 Automating the sample Start and End markers
Besides the regular modulation capabilities in Mimic, you could also automate and modulate the sample Start and
End markers from the main sequencer, to have them reposition in real-time during sequencer playback. This could be
useful if you have a longer recording/sample that you want to alter the playback position in:
1. Record some notes on the Mimic sequencer track in the sequencer and then hit Stop twice.
2. Hit Record again in the sequencer and drag the Sample Start and Sample End markers during recording:

q If you like, drag and move the Start/End marker up into the Slice Marker area above the waveform, to have the
Start/End marker snap to the closest slices. This works in all modes, not just in Slice Mode.
3. Hit Stop in the sequencer twice when you are done recording.
If you moved both the Sample Start marker and the Sample End marker you will now have two Parameter Auto-
mation lanes with clips on them on the Mimic sequencer track:

• The clips on the Start Pos and End Pos lanes represent the movements of the Sample Start and Sample End
markers respectively.

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Chapter 20
Monotone
Bass Synthesizer
 Introduction

The Monotone Bass Synthesizer is a great little monophonic bass synthesizer. Despite its fairly small size, it’s very
versatile can produce really fat and punchy bass synth sounds.
Monotone features two oscillators, a 24 dB lowpass ladder filter, amp envelope and chorus and delay effects. It also
has an LFO and an additional envelope for modulation purposes.

334 MONOTONE BASS SYNTHESIZER

 Panel overview
The Monotone front panel contains the following sections:

1 2 3
4 5 6 7 8 9

10

The Monotone front panel sections.

• 1. MIDI Note On LED.
• 2. Patch Selector (for browsing, loading and saving patches).
• 3. Master Volume.
• 4. Voicing section and global controls.
• 5. Oscillator section.
• 6. Filter section.
• 7. Amplitude Envelope.
• 8. Chorus section.
• 9. Delay section.
• 10. Modulation section.

335 MONOTONE BASS SYNTHESIZER

 Signal flow
The picture below shows the basic signal flow in Monotone:
OCT Wave

Drive Freq Res A D S R Amount Amount

Oscillator 1 Noise

FM Env Detune Filter Amplifier Chorus Delay

OSC
MIX Master
Volume
Oscillator 2 ENV
(Freq) KEY LFO VEL Rate Spread Time Feedb.
(Freq)

LFO
(pitch) OCT Wave Semi
Shape
Envelope LFO
Rate
VEL

A D S R : audio signal
: control signal

Monotone signal flowchart.

• The “heart” of Monotone is the Oscillator section, which generates the basic audio signal.
There are two oscillators in Monotone, that can produce a number of traditional “analog” waveforms - plus a Noise
generator, which produces white noise. The two oscillators can be detuned, to create a fatter sound. Oscillator 2
can also frequency modulate Oscillator 1, for metallic/bell type sounds.
• The signal from the Oscillator section is routed to the Filter.
The Filter in Monotone is a classic 24 dB/octave lowpass ladder filter, with overdrive control.
• The signal from the Filter is routed to the Amplifier.
The Amplifier is controlled by a standard ADSR (Attack-Decay-Sustain-Release) envelope.
• The signal from the Amplifier is then routed to the Chorus module.
This module generates a chorus effects, to make the sound fat and wide.
• The signal from the Chorus is then routed to the Delay module.
Here you can add audio delay effects.
• The remaining section in Monotone (Envelopes and LFO) can be used for modulating some Oscillator and
Filter parameters.

336 MONOTONE BASS SYNTHESIZER

 Playing and using Monotone
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Global output controls

Master Volume
This is the main stereo output volume control.

Global performance and “play” controls

Portamento
Portamento makes the note pitch glide from the previous note to the new one, at the time set with the Portamento
knob.
• When On, the pitch will always glide between consecutive notes.
• In Auto mode, the pitch will glide between consecutive notes only when you play legato.
If you release the previous key before hitting the new key, there will be no portamento effect.

Retrig
D Click the Retrig button if you want to play Monotone and always retrigger the envelopes as soon as you play a
new note.
When Off, the envelopes will retrigger only if you have released the previous note before playing the new note.

337 MONOTONE BASS SYNTHESIZER

 Range
D Set the desired Pitch Bend range for the “Pitch” wheel with the up/down buttons, or by click-holding on the
display and dragging up/down.
Range: +/-24 semitones (+/-2 octaves) in steps of +/-1 semitone.

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Monotone also responds to Pitch Bend
MIDI data from a connected MIDI master keyboard. You set the desired Pitch bend Range with the “Range” control
above the Pitch bend wheel.

Mod
The Mod wheel can be used for controlling the Filter Frequency and LFO intensity in Monotone.
D Raise the FILT knob above the Mod wheel to set the Filter Frequency modulation amount.
D Raise the LFO knob above the Mod wheel to set the LFO intensity modulation amount.
Note that for the LFO modulation to work you need to already have some LFO modulation set in the Oscillator
(see “LFO”) and/or Filter (see “LFO”) sections.

Panel reference
The Oscillator section

Here is where you choose oscillator waveforms and set the pitches for the two oscillators. You can also add noise
and frequency modulate Oscillator 1 from Oscillator 2.

Waveform selector
D Turn the Wave knob to select one of four wave shapes.
The wave shapes are:
• Ramp
Also known as sawtooth. Generates a rich tone with both even and odd harmonics (overtones).
• Square (Pulse in Oscillator 2)
The square wave has a symmetric square shape and contains only odd harmonics. The Pulse wave is basically a
square wave with non-symmetrical shape, i.e. a duty cycle that is not 50%. The Pulse wave generally sounds a little
thinner than a perfect square wave.

338 MONOTONE BASS SYNTHESIZER

 • Triangle
The Triangle wave also only contains odd harmonics, but at lower intensities than the square wave overtones. Ev-
ery other harmonic is also phase-shifted by 180°. This makes it sound a little “rounder” and with less bite than the
square wave.
• Sine
The Sine wave doesn’t contain any overtones - only the fundamental. This makes it sound dull and makes it perfect
as a sub bass an octave or two below another waveform in the other oscillator.

Oct
D Set the pitch in octave steps.
Range: 5 octaves.

Osc Mix
D Set the mix of the Oscillator 1 and 2 signals.

Noise
D Turn up the Noise knob to introduce white noise to the oscillator signal mix.

Detune
D Change the pitch in steps of 1 cent (in opposite directions for the two oscillators).
Range: +/- 50 cents (down or up half a semitone).

LFO
D Turn the LFO knob to introduce pitch modulation to both oscillators from the current setting of the LFO section
(see “The LFO section”).

FM Env
D Turn the knob to have the oscillator 2 signal frequency modulate the oscillator 1 signal according to the cur-
rent Envelope settings (see “The Envelope section”).
Range: 0% (no tracking (constant pitch)) to 100% (1 semitone per note).

Osc 2 Semi
D Set the pitch of Oscillator 2 in semitone steps.
Range: +/-12 semitones (two octaves).

The Filter section

The Filter in Monotone is a classic 24 dB/octave lowpass ladder filter. If you raise the Resonance high enough, the
filter will start to self-oscillate.

339 MONOTONE BASS SYNTHESIZER

 The picture below shows the lowpass filter’s basic characteristics at four different resonance levels:
Amplitude

RESONANCE

Frequency
FREQ

Drive
D Turn the Drive knob to amplify and introduce an overdrive type of distortion to the signal fed into the filter.

Freq
D Set the cutoff frequency for the filter.
The cutoff frequency is where the filter starts to cut out/dampen the higher frequencies of the signal.

Resonance
D Set the resonance amount.
This controls the resonance peak level at the currently set cutoff frequency (see “Freq” above).
The picture below shows a ramp oscillator signal lowpass-filtered at three different Resonance levels:

Amplitude

Amplitude Amplitude

Time Time
RESONANCE

Frequency
FREQ
(Cutoff Frequency)

Amplitude

Amplitude Amplitude

Time Time
RESONANCE

Frequency
FREQ
(Cutoff Frequency)

Amplitude

Amplitude Amplitude

Time Time
RESONANCE

Frequency
FREQ
(Cutoff Frequency)

! Be careful when using high Resonance values as this could generate quite loud audio levels!

340 MONOTONE BASS SYNTHESIZER

 Env
D Turn the Env knob to set the frequency modulation amount from the current settings of the Envelope (see “The
Envelope section”).

Key
D Turn the Kbd knob to set the keyboard tracking amount.
At 0% the filter cutoff frequency is static and doesn’t track the keyboard at all.
At 100% the filter cutoff frequency tracks the keyboard 1 semitone per note.

LFO
D Turn the LFO knob to set the frequency modulation amount from the current settings of the LFO (see “The LFO
section”).

The Amplifier section

The Amplifier section contains a standard ADSR envelope, which controls the amplitude of the audio signal.
The picture below shows the various stages of the ADSR envelope:

Level

Volume
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to max level.
How long this should take, depends on the Attack setting. If the Attack is set to “0”, maximum level is reached in-
stantly. If the Attack value is raised, it will take longer time before the maximum level is reached.

D(ecay)
After maximum level has been reached, the level starts to drop. How long this should take is governed by the Decay
parameter.

341 MONOTONE BASS SYNTHESIZER

 If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to “0”,
the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the volume
gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to drop to
some other value than zero, you raise the Sustain parameter.

S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain to
full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain param-
eter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays
there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to max level, then gradually
decreases to finally land to rest on a level somewhere in-between zero and maximum level. Note that Sustain
represents a level, whereas the other envelope parameters represent times.

R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume to
drop back to zero after you release the key.

Vel
D Turn up the Vel knob if you want the maximum level to be controlled from Keyboard Velocity.
The harder you play, the louder the maximum volume.

Chorus

This is a stereo Chorus effect, which can be used for generating a fatter and wider sound.

Amount
D Set the Dry/Wet amount of the chorus effect.
Set to 0% for a completely dry (unprocessed) signal.

Rate
D Set the rate/speed of the chorus modulation.

Spread
D Set the stereo width of the chorus effect.
Set to 0% for a if you want the signal to be in mono.

342 MONOTONE BASS SYNTHESIZER

 Delay

This is a stereo delay, which generates delayed copies of the original signal.

Amount
D Use this parameter to adjust the send level to the Delay effect.
Set to 0% for a completely dry (unprocessed) signal.

Time
The delay time is synced to the main sequencer tempo.
D Set the sync division to the main sequencer tempo with the Time knob.
Range: 1/16, 1/8T, 1/8, 2/8T, 3/16, 1/4, 5/16, 4/8T, 7/16 and 2/4.

Feedback
D Set the number of delay repeats.

Ping Pong
D Activate Ping Pong to have the delay repeats alternate between left and right in the stereo panorama.

The LFO section

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO
modulate the pitch of a signal to produce vibrato, but there are also other applications for LFOs. The LFO section fea-
tures an LFO which can be set to control Oscillator pitch (see “LFO”) and/or Filter frequency (see “LFO”).

Rate
D Set the LFO rate/speed.
Range: 0.06-94.0 Hz

Shape
D Turn the Shape knob to select one of three LFO wave shapes.
The wave shapes are: Sine, Triangle and Square.

343 MONOTONE BASS SYNTHESIZER

 The Envelope section

The Envelope section features a standard ADSR envelope, which can be used for controlling Oscillator Frequency
Modulation (see “FM Env”) and/or Filter Frequency (see “Env”).
The various envelope stages work exactly like those of the Amplifier, see “The Amplifier section”.

Level

Volume
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
D Set the time it should take to reach from zero to maximum level.

D(ecay)
D Set the time it should take to go from maximum level to the Sustain level (see below).

S(ustain)
D Set the level the envelope should rest at, after the Decay stage (see above).

R(elease)
D Set the time it should take to go from the Sustain level back to zero, after you have released the note.

Vel
D Turn up the Vel knob if you want the maximum level to be controlled from Keyboard Velocity.
The harder you play, the higher the maximum level.

344 MONOTONE BASS SYNTHESIZER

 Connections

! Remember that CV connections are NOT stored in the Monotone patches! If you want to store CV connections
between devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Monotone from another CV/Gate device (typically a
Matrix or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers
note on/off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

Modulation inputs
These control voltage (CV) inputs and can be used for modulating the corresponding parameters from external mod-
ulations sources.

Audio Output
These are the main audio outputs. When you create a new Monotone device, these outputs are auto-routed to the
first available outputs in the I/O device.

345 MONOTONE BASS SYNTHESIZER

346 MONOTONE BASS SYNTHESIZER
Chapter 21
NN-XT Sampler
 Introduction
The basic functions of the NN-XT are very similar to those of its sampler companion in the Reason rack - the NN-19
(see “NN-19 Sampler”). Just like the NN-19, NN-XT lets you load samples and create multi-sample patches by map-
ping samples across the keyboard. The sound can then be modified by a comprehensive set of synth-type parame-
ters. There are however some major differences between the two. The NN-XT has:
D Support for SoundFonts.
Presets and samples from SoundFont banks can be loaded and used in the NN-XT (see “Loading SoundFonts”).
D 8 stereo output pairs.
This makes it possible to route different samples to different mixer channels for individual effect processing (see
“Out”).
D The possibility to create layered sounds.
This is done by mapping several samples across the same keyboard range (see “Creating layered sounds”).
D The possibility to create sounds that only play over certain velocity ranges, velocity switched key maps and ve-
locity crossfading.
See “Setting velocity range for a Zone”.
D Key maps with individual synth parameter settings for each sample.
See “Synth parameters”.
Even though the NN-XT is a more advanced sample player than NN-19, it should not be considered as a successor
to the NN-19, but rather as a complement to it. The NN-19 will for example probably still be the sampler of choice for
those of you who want to be able to quickly load a couple of samples and start playing, since that particular aspect
takes a little more doing with the NN-XT.
! Please, note that this device is not available in Reason Lite Rack Plugin.

348 NN-XT SAMPLER

 Panel overview
The main panel
When the NN-XT is added to the rack, you will initially only see the main panel.

The NN-XT main panel.

The main panel is where you load complete sample patches. It also contains the “global controls”. These are controls
that affect and modify the sound of entire patches rather than the individual key zones.

The Remote Editor panel

To show/hide the remote editor panel, use the fold/unfold arrow at the bottom left.

Remote
Editor
Fold/Unfold
button

The remote editor panel is where you load individual samples, create key maps, modify the sound of the samples with
synth parameters etc.
! The main panel of the NN-XT can be folded like any other Reason device. Note that folding the main panel will
also fold the remote editor regardless of its current state.

349 NN-XT SAMPLER

 Loading complete Patches and REX files
As previously alluded, you can load complete sample patches as well as individual samples into the NN-XT.
• A patch is a complete “sound package”. It contains information about all the samples used, assigned key
zones, associated panel settings etc. Loading a sample patch is done by using the patch browser on the main
panel, and works in the same way as with any other Reason device.

The Browse Patch button on the main panel.

For general instructions on how to load and save patches, please see “Loading patches” and “Saving patches”.
• Loading separate samples is done in a similar way, but via the sample browser on the remote editor panel. If
you load samples, map them across keyboard ranges and set up the sound the way you want it, you can save
your settings as a Patch for easy access later.

The Browse Sample button on the remote editor.

More about loading samples later in this chapter.

Loading NN-XT Patches

NN-XT Patches are patches made specifically for the NN-XT. Reason ships with a large number of NN-XT Patches,
some in the Factory Sound Bank but most in the Orkester Sound Bank. NN-XT Patches have the extension “.sxt”.
1. Click the Browse Patch button on the front panel to set browse focus to the NN-XT device.
2. Navigate to the folder that contains the NN-XT patch you wish to load, select it and click Load in the Browser.
D Alternatively, drag an NN-XT patch from the Browser and drop it on the NN-XT device in the rack.
The panel is dimmed in orange and the Patch Replace symbol appears in the center.

Loading NN-19 Patches

NN-19 Patches have the extension “.smp”. Note that when loading NN-19 patches into the NN-XT, some parameters
will not be applicable since the NN-19 and the NN-XT to some extent differ from each other in terms of controls. In
these cases, the concerned parameters will either be ignored by the NN-XT or mapped to the most equivalent con-
trol.
1. Click the Browse Patch button on the front panel to set browse focus to the NN-XT device.
2. Navigate to the folder that contains the NN-19 patch you wish to load, select it and click Load in the Browser.
D Alternatively, drag an NN-19 patch from the Browser and drop it on the NN-XT device in the rack.
The panel is dimmed in orange and the Patch Replace symbol appears in the center.

350 NN-XT SAMPLER

 Loading SoundFonts
The SoundFont format was developed by E-mu systems in collaboration with Creative Technologies. It is a standard-
ized data format containing wavetable synthesized audio and information on how it should be played back in wavet-
able synthesizers - typically on audio cards. The SoundFont format is an open standard so there is a vast amount of
SoundFont banks and SoundFont compatible banks developed by third parties.
Loading SoundFonts is no different from loading NN-XT Patches. As with NN-19 Patches, the NN-XT does its best
to map all the SoundFont settings to NN-XT parameters.
You can load SoundFont presets by using the patch browser, and single SoundFont samples by using the sample
browser.

Loading complete REX files as Patches

REX files are music loops created in the ReCycle program or when editing audio clips inline in Reason (see “Bounce
Clip to REX Loop”). In Reason, REX files are primarily used in the Dr. Octo Rex loop player, but they can be used in
the NN-XT as well. Possible extensions are “.rx2”, “.rcy” and “.rex”.
1. Click the Browse Patch button on the front panel to set browse focus to the NN-XT device.
2. Navigate to the folder that contains the REX loop you wish to load, select it and click Load in the Browser.
D Alternatively, drag a REX loop from the Browser and drop it on the NN-XT device in the rack.
The panel is dimmed in orange and the Patch Replace symbol appears in the center.
When loading a REX file, each slice in the file is assigned to one key, chromatically, starting from C1. All parame-
ters are set to their default settings.
When using REX files in the Dr. Octo Rex loop player, it is possible to make a track play the slices in order to recreate
the original loop. To do the same in the NN-XT requires a few extra steps.
1. Use the Browser to load the REX file into an NN-XT sampler.
2. Create a Dr. Octo Rex loop player and load the same REX file into a Loop Slot of this device.
3. Use the Copy Loop To Track feature on the Dr. Octo Rex to create playback data (a group) on the track as-
signed to the Dr. Octo Rex.
4. Move that group to the track that plays the NN-XT and play it back from there.
5. Delete the Dr. Octo Rex loop player.

351 NN-XT SAMPLER

 Using the main panel
All of the controls on the main panel are used for globally modifying certain parameters for all of the samples in a
patch, by the same amount.
! Movements of the parameters on the main panel can be recorded as automation. However, controls on the re-
mote editor panel (described later) can not!
The following is a description of the controls and parameters on the main panel.

The Pitch and Modulation wheels

Most MIDI keyboards come equipped with Pitch Bend and Modulation wheels. The NN-XT features settings for how
incoming MIDI Pitch Bend and Modulation wheel messages should affect the sound. The wheels on the NN-XT will
also mirror the movements of the wheels on your MIDI keyboard.
If you don’t have Pitch Bend or Modulation controls on your keyboard, or if you aren’t using a keyboard at all, you can
use the two fully functional wheels on the NN-XT to apply real time modulation and pitch bend.
• The Pitch Bend wheel is used for “bending” the played notes up and down to change their pitch - much like
bending the strings on a guitar or other string instrument. The Pitch Bend Range is set on the remote editor
panel (see “Pitch Bend Range”).
• The Modulation wheel can be used for applying modulation to the sound while you’re playing. It can also be
used for controlling a number of other parameters, as described in “The Modulation controls”.

The External Control wheel

This section can be used in three ways:

Receiving MIDI controller messages from external sources

NN-XT can receive common MIDI controller messages, and route these to various parameters. You use the “Source”
selector switch to determine which type of message should be received:
• Aftertouch (Channel Pressure)

352 NN-XT SAMPLER

 • Expression Pedal
• Breath Control
If your MIDI keyboard is capable of sending aftertouch messages, and/or if you have connected an expression pedal
or a breath controller to it, you can use these to modulate NN-XT parameters. Which parameters should be modu-
lated is set in the remote editor panel (see “The Modulation controls”).

Recording MIDI controller messages with the wheel

The wheel in the external control section can be used for recording any or all of the three MIDI controller message
types into the main sequencer. If your MIDI keyboard isn’t capable of sending aftertouch messages or you don’t have
access to an expression pedal or a breath controller, you can use the wheel instead. This is done just as with any
other automation recording.

High Quality Interpolation

This switch turns High Quality Interpolation on and off. When it is activated, the sample pitch is calculated using a
more advanced interpolation algorithm. This results in better audio quality, especially for samples with a lot of high
frequency content.
• High Quality Interpolation uses more computer power - so if you don’t need it, it’s a good idea to turn it off! Lis-
ten to the sounds in a context and determine whether you think this setting makes any difference.

Global Controls

All of these knobs change the values of various parameters in the remote editor panel and affect all loaded samples.
Thus they can be used for quickly adjusting the overall sound.
The knobs are bi-polar, which means that when they are centered, no parameter change is applied. By turning them
to the right you increase the corresponding value, and by turning them to the left, you decrease the value.
Again, the movements of these parameters can be recorded as automation. This is done just as with any other auto-
mation recording.
The controls are, from left to right:

353 NN-XT SAMPLER

 Filter
These two knobs each control a parameter of the filter (see “The Filter section”). Note that the filter must be on for
these to have any effect.
• Frequency
This changes the cutoff frequency of the filter.
• Resonance
This changes the resonance parameter of the filter, meaning - the filter characteristic, or quality.

Amp Envelope
These three knobs control the Amplitude Envelope (see “The Amplitude Envelope”) in the following way:
• Attack
This changes the Attack value of the Amplitude Envelope. That is, how long it should take for the sound to reach
full level after you press a key on your keyboard.
• Decay
This changes the Decay value of the Amplitude Envelope. Decay determines how long it should take for the sound
to go back to the sustain level after it has reached full value (see “The Amplitude Envelope”) and the key that trig-
gered the sound is still being pressed.
• Release
This changes the Release value of the Amplitude Envelope. Release works just like Decay with the exception that
it determines how long it should take for the sound to become silent after the key has been released.

Mod Envelope
This knob controls the Decay value of the Modulation Envelope (see “The Modulation Envelope”). Also see above
for a brief description of Decay.

Master Volume
This controls the main volume out from the NN-XT. Turn the knob to the right to increase the volume.

354 NN-XT SAMPLER

 Overview of the Remote Editor panel
It is in the Remote Editor Panel that the main NN-XT action is going on, especially if you’re creating your own
patches. The remote editor is dominated by the key map display, and this is also the part on which we will concentrate
to begin with.

The Key Map display

The key map display consists of a number of separate areas that let you do different things. To help you navigate the
key map display, these areas are described below.
The Keyboard area The Tab Bar area
The Info
area The Scrollbars

The Sample
area

The Group
area

The Key Range area

The Info area

This displays the following information about the currently selected sample: Sample rate, mono/stereo information,
bit resolution and file size.

The Sample area

This area displays the names of the samples in each zone. It also allows you to change the order of the zones by
clicking and dragging them up and down.

The Group area

This area does not show any information. However, by clicking in it, you can instantly select all the zones that belong
to a certain group. See “Working with Grouping” for information on how to create groups.

The Keyboard area

Aside from the fact that is a guideline for setting up key ranges, it is also used for setting the root keys of, and audi-
tioning loaded samples. See “About the Root Key” and “Auditioning samples” respectively for more information.

The Tab Bar area

This area gives you a visual indication of the key range of a selected zone. By clicking and dragging the “handles” at
the key range boundaries, you can resize the key ranges, and by clicking in between the handles, you can move the
key ranges without changing their length.

355 NN-XT SAMPLER

 The Key Range area
This area in the middle of the key map display is where you keep track of all the zones and the relationship between
them. You can also move and resize the zones just like in the Tab Bar area, as described above.

The Scrollbars
There are both horizontal and vertical scrollbars that work just like regular scrollbars. Whenever there is more infor-
mation in the key map display than what fits on a “single screen”, you can use the scrollbars to reveal it. Either click
on the arrows or click and drag the scrollbar handles.

Sample parameters

This area shows the current values of basic parameters you can set for zones, such as root key, play mode, output
etc. The parameters are changed by using the knobs directly below the key map display.

Group parameters

These parameters are adjusted on a per group basis (see “Group parameters” for more information on groups). Most
of them relate to performance or playing style.

356 NN-XT SAMPLER

 Synth Parameters

The bulk of the parameters on the remote editor are used for adjusting the sound of the samples by applying filtering,
envelope shaping, modulation (like vibrato and tremolo) and so on. We call these the synth parameters, since they are
to a large extent identical to those on a regular synthesizer.

About Samples and Zones

For a clear understanding of the terminology used when describing the various operations that can be performed in
the key map display, it is important to clarify the distinction between a sample and a zone:
• A Sample is a piece of audio that can be loaded into the NN-XT and played back.
• A Zone could be viewed as a “container” for a loaded sample.
All loaded samples are placed in “Zones” in the key map display. You can then organize the zones as you please, and
make various settings such as key- and velocity ranges separately for each zone.
In other words, the settings you make are actually performed on the zones, but affect the samples in them. Hence,
when we talk about making settings for a zone, it is synonymous with making settings for a sample - the sample that
the zone contains.
• Two or more zones can play the same sample, but with different parameter settings, making them sound com-
pletely different.
• A zone can be empty, playing no sample at all.

357 NN-XT SAMPLER

 Selections and Edit Focus
Almost all operations in the remote editor are performed on one or more selected zones or on the zone with edit fo-
cus. Several zones can be selected at once, but only one zone at a time can have edit focus.
This is important since:
• Editing operations that can be performed on several zones (like deleting), always apply to the selected zones.
• Editing operations that can be performed on one zone only, always apply to the zone with edit focus.
• The front panel always shows the settings for the zone with edit focus.

Here no zone is selected.

Here the middle zone is selected but does not have edit focus.

Here the middle zone has edit focus but is not selected. Notice the thicker border and the additional handles in the key range area.

Here the middle zone is selected and has edit focus.

Here, all three zones are selected, but the middle one has edit focus.

358 NN-XT SAMPLER

 Selecting Zones
D To select a zone, click on it.
Clicking on a zone will also automatically give it edit focus.
You can also select multiple zones in several ways:
D By holding down [Shift], or [Ctrl](Win) or [Cmd](Mac) and clicking on the zones you wish to select.
This way you can select several non-contiguous zones. You can also deselect a selected zone by clicking on it
again.
D By clicking and dragging a selection box in the key map area.
Making a selection box like this....

...will select these zones:

Note that the zones don’t have to be completely encompassed by the selection box. The selection box only have to
intersect parts of the zones to include them in the selection.

359 NN-XT SAMPLER

 Selecting zones via MIDI
You can also select zones via your MIDI keyboard. By clicking the button marked “Select zones via MIDI” above the
key map display so that it lights up, you enable selection via MIDI.

This way, you can select a zone and give it edit focus by pressing a key that lies within the zone’s key range (see later
in this chapter for information about setting up key ranges).

In this case, this zone can be selected by pressing any key between C2 - C3 on your MIDI keyboard.
Note also, that selection via MIDI is velocity sensitive. Zones may have specific velocity ranges. This means that they
won’t be played unless the key that triggers the zone is played with a certain velocity. The same rules apply when se-
lecting via MIDI, only zones that meet the velocity criteria will be selected. Read more about setting up velocity ranges
on “Setting velocity range for a Zone”.

Selecting All Zones in a Group

The concept of zone groups is fully introduced on “Working with Grouping”. For now we will only describe how to se-
lect all samples that belong to the same group:
Clicking in the
Group column...

...selects all zones in

the group

360 NN-XT SAMPLER

 Moving Edit Focus
Moving Edit Focus
A zone can be given edit focus independently of selection:
D When you click on an unselected zone, it both gets selected and gets edit focus.
D When you select several zones using [Shift], or [Ctrl](Win) or [Cmd](Mac), the one you select last always gets
edit focus.
D To set edit focus to a zone when several zones are already selected, click on it without holding down any mod-
ifier keys.
This way, you can move edit focus between the selected zones freely without deselecting any of them.

Adjusting parameters
Adjusting Synth parameters
The synth parameters are the ones that occupy the bulk of the remote editor panel (see “Synth Parameters”).
Changes you make to synth parameters always apply to all selected zones.
D The panel always shows the settings for the zone with edit focus.
More about this below.
D To make adjustments to one zone, select it (which also gives it edit focus) and adjust the parameter on the
front panel.
D To set several zones to the same value, select them and adjust the parameter.
All zones will be set to the same value for the parameter you adjusted.

Adjusting Group parameters

Group parameters apply to a group. That is, they are settings that are shared by all zones in a group.
D To make adjustments to one group, select one or more zones that belong to the group, and adjust the param-
eter on the front panel.
D To set several groups to the same value, select at least one zone in each group you want to adjust, and adjust
the parameter.
All groups will be set to the same value. More about this below.

About “Conflicting” parameters

Often you will find yourself in a situation where you select multiple zones and parameter settings differ between
them. This is quite normal. For example, you will often find yourself making adjustments to for example level and fil-
tering to balance the sound between several samples across the keyboard. However, if you have multiple selections
this can sometimes lead to confusion: Enter the NN-XT’s “conflicting parameters” indication:

361 NN-XT SAMPLER

 Whenever two or more selected zones have conflicting parameter settings, NN-XT will notify you about this by show-
ing a small “M” (for multiple) symbol, next to the parameter.

In this example, Level and Spread have conflicting settings.

D The controls on the panel always show the setting for the zone with edit focus.
D By clicking your way through the zones within the selection, you can see the settings for each zone.
D If you adjust a parameter, all selected zones will be set to the same value for this parameter.
You can put this functionality to good use when checking how a patch has been created and when checking that
your own settings are consistent through the various zones.

Sample parameters
The Sample parameters allow you to specify various properties for one or several selected zones, such as tuning, key
and velocity ranges.

D To set several zones to the same value, select them and adjust the parameter.
All zones will be set to the same value for the parameter you adjusted.

Copying parameters between zones

You can easily copy parameter settings from one zone to any number of other zones. Proceed as follows:
1. Select all the zones you want to involve in the operation.
By this we mean the zone with the settings you wish to copy, and the zone(s) to which you want to copy the set-
tings.
2. Make sure the zone that contains the settings you want to copy has edit focus.
3. Pull down the Edit menu or the NN-XT context menu and select “Copy Parameters to Selected Zones”.
All the selected zones will now get the exact same parameter settings.
! Observe that this only applies to the synth parameters (see “Synth parameters”). Sample parameters (root
key, velocity range etc.) can not be copied.

362 NN-XT SAMPLER

 Managing Zones and Samples
Creating a Key Map
When you add an NN-XT sampler to the rack and select “Reset Device” from the context menu or Edit menu, its key
map display becomes empty. That is, it contains no samples.
To create a new key map, proceed as follows:
1. Either click the Browse Samples button, select Browse Samples from the Edit menu or select Browse Samples
from the NN-XT’s context menu.
The NN-XT device gets browse focus.

The Browse Samples button.

2. Select the sample or samples that you want to load in the browser and click the “Load” button in the Browser.
The selected sample(s) are loaded into the NN-XT.
D Alternatively, drag one or several sample files from the Browser and drop it on the NN-XT device in the rack.
The panel is dimmed in blue and a Sample Replace symbol appears in the center.
When new samples are loaded into the NN-XT they have the following properties:
• Each sample is placed in its own zone.
• Each zone spans a key range of five octaves on the keyboard - C1 to C6.
• All the newly added sample(s)/zones are automatically selected.
• The first added zone gets edit focus.

A key map with four newly added samples.

Setting Root Notes and Key Ranges

The next step after loading the samples is most likely to adjust the key range, root note and tuning of the samples, so
that they play sensibly across the key range. There are many ways of doing this, described in “Working with Key
Ranges” and onwards. However, we will here briefly describe a procedure for quickly creating a complete key map
out of a set of loaded samples.
This example assumes that the samples you load is a set of multisamples for a pitched instrument (like guitar, piano,
flute etc.).
1. Load the samples.
2. Use “Select All” on the Edit menu to select all the loaded samples.
3. Use “Set Root Notes from Pitch Detection” to automatically set up the root notes (pitches) for the samples.

363 NN-XT SAMPLER

 4. Select “Automap Zones” from the Edit menu.
All selected zones are automatically arranged into a basic key map. You can now proceed with adjusting the synth
parameters on the front panel to shape the sound!

About file formats and REX slices

The audio file format support differs depending on which computer OS you are using.
The NN-XT can read audio files in the following formats:
• In Windows:
.wav, .aif, .mp3, .aac, .m4a and .wma.
• In macOS:
.wav, .aiff, .3g2, .3gp, .mp1, .mp2, .mp3, .mpeg, .mpa, .snd, .au, .sd2, .ac3, .aac, .adts, .amr, .caf, .m4a .m4r and .mp4.
• SoundFont samples
This is a standardized data format containing wavetable synthesized audio and information on how it should be
played back in wavetable synthesizers - typically on audio cards. SoundFont banks are hierarchically organized into
different categories: User Samples, Instruments, Presets etc. The NN-XT lets you load single samples from within
a Soundfont bank.
• REX file slices
A slice is a snippet of sound in a REX File (see “Loading complete REX files as Patches”). To import a REX slice,
browse to a REX file and open it as if it was a folder. The browser will then display the slices as files inside that
“folder”. In the rest of this manual, when we refer to importing samples, all that is said applies to REX slices as well.
• Any sample rate and practically any bit depth.

Adding more samples to the Key Map

You can add additional samples to an existing key map way described above.
1. Make sure that no already loaded sample has edit focus.
If you don’t, there’s a risk that the selected sample will be replaced, see below. To remove the edit focus, click in an
unoccupied area in the Sample column or the key map area.
2. Open the Sample Browser.
3. Select the sample(s) you want to load in the browser and click the “Load” button in the Browser.
D Alternatively, drag one or several sample files from the Browser and drop it on the NN-XT device in the rack.
The panel is dimmed in blue and a Sample Replace symbol appears in the center.
The new sample(s) are added to the key map.

Replacing a sample
To replace the sample in a zone, proceed as follows:
1. Make sure the zone has edit focus and do one of the following:
D Click the Browse Samples button.
D Select Browse Samples from the Edit menu or the NN-XT context menu.
D Double click in the zone.
Any of these methods will set browse focus and open the standard file browser in which you can select new sam-
ples for the zone.

364 NN-XT SAMPLER

 2. Select one and only one sample in the Sample Browser.
If you select more than one sample in the browser the samples you load will not replace the one with edit focus.
They will instead be added below it.
3. Click the Load button in the Browser.
D Alternatively, drag a sample file from the Browser and drop it on the NN-XT device in the rack.
The panel is dimmed in blue and a Sample Replace symbol appears in the center.

Quick browsing through samples

If you want to quickly browse through a number of samples, for example to see which one of them would fit best in a
certain context, proceed as follows:
1. Set up the zone as desired and make sure it has edit focus:
2. Use the arrow buttons in the Browse Samples section to select the next/previous sample in the directory.

Removing samples
D To remove a sample from a zone, select it by clicking on it and then select “Remove Samples” from the Edit
menu or the NN-XT context menu.
This will remove the sample from the zone, leaving it empty. Note that you can remove the samples from several
selected zones at the same time.

Auditioning samples
You can audition the loaded samples in two ways:
D By pressing [Alt](Win) or [Option](Mac) and clicking a sample in the sample column.
The mouse pointer will take on the shape of a speaker symbol when you move it over the sample column.
Clicking a sample will play it back at its root pitch (see “About the Root Key”). Furthermore, the sample will play
back in its unprocessed state. That is, without any synth-parameters applied (see “Synth parameters”).
D By pressing [Alt](Win) or [Option](Mac) and clicking a sample in the keyboard column.
The difference here is that you will hear the sample at the pitch corresponding to the key you clicked and with any
and all processing applied. The click mimics a key played with velocity 100. Also note that this may trigger several
samples, depending on whether they are mapped across the same or overlapping key ranges, and the velocity
range settings (see “Setting up Key Ranges” and “Setting velocity range for a Zone” respectively).

Adding empty Zones

You can add empty zones to a key map. Empty zones are treated just like zones containing samples, in that they are
automatically selected, gets edit focus and are assigned a five octave key range when they are first created. How-
ever, you can only add one zone at a time. It is also possible to resize, move and edit empty zones in the same way as
zones containing samples.
D To add an empty zone, pull down the Edit menu or the NN-XT context menu and select “Add Zone”.
An empty zone is added below any existing zones in the key map. An empty zone is indicated with the text “**No
Sample**”.
After you have added an empty zone, you can assign a sample to it, just as when Replacing a Sample, or when Quick
Browsing, as described above.

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 Duplicating Zones
You can duplicate any number of already existing zones (containing samples or empty).
1. Select the zone(s) you want to copy.
2. Pull down the edit menu or the NN-XT context menu and select “Duplicate Zones”.
The selected zones will now be copied and automatically inserted below the last one in the key map display.
The duplicated zones will contain references to the same samples as the original zones. They will also have the exact
same key ranges and parameter settings.

Using Copy and Paste

The Copy Zones function on the Edit menu allows you to copy all selected zones to the clipboard. Selecting Paste
Zones from the Edit menu will paste the zones into the selected NN-XT device, below the existing zones.
This is a handy way to transfer zones (complete with all settings) from one NN-XT device to another.

Removing Zones
To remove one or several zones, select them and do one of the following:
D Press [Delete] or [Backspace] on the computer keyboard.
D Select “Delete Zones” from the Edit menu or the NN-XT context menu.
When removing zones, you will remove any samples in them as well.

Rearranging Zones in the List

D To move a zone to another position in the list, click on it in the samples column and drag up or down.
An outline shows you where the zone will appear when you release the mouse button.

Working with Grouping

About Groups
Grouping has two purposes:
D To allow you to quickly select a number of zones that “belong together.”
For example if you have created a layered sound consisting of piano and strings, you could put all string samples
in one group and all piano samples in one group. Then you can quickly select all piano samples and make an ad-
justment to them by trimming a parameter.
D To group zones that need to share group settings together.
For example, you may want to set a group to legato and monophonic mode and add some portamento so that you
can play a part where you slide between notes.
Note that there is always at least one group, since the zones you create are always grouped together by default.

Creating a Group
1. Select the zones you want to group together.
The zones don’t have to be contiguous in order to be grouped. Regardless of their original positions in the samples
column, they will all be put together in succession.

366 NN-XT SAMPLER

 2. Select “Group Selected Zones” from the Edit menu or the NN-XT context menu.
The zones are grouped.
Selecting these zones and grouping them...

...will create these two groups instead of the original one large group.

Moving a Group to another position in the List

D Click on the group in the Groups column and drag up or down with the mouse button pressed.
An outline of the group you move is superimposed upon the display to help you navigate to the desired position.

Dragging a group to a new position.

3. Release the mouse button at the desired position.
The group and all its zones appear at the new position.

Moving a Zone from one Group to another

This is done just as when rearranging samples in the list, as described on the previous page. The only difference is
that you drag the zone from one group to another.

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 Selecting a Group and/or Zones in a Group
D Clicking on a group in the groups column selects the group and all the zones in the group.

D Clicking on a zone in the samples column selects the group (and that zone).

The Group Parameters

There are a few parameters on the front panel that apply specifically to groups. see “Group parameters” for details.

Working with Key Ranges

About Key Ranges
Each zone can have its own separate key range, the lowest and the highest key that will trigger the sample.
A good example of use for this is when sampling a certain instrument. Sampling of a piano for example is usually per-
formed by making several recordings of different notes at close intervals, and then mapping these samples to sepa-
rate, contiguous, fairly narrow key ranges. This concept is called multi-sampling.
The reason for this is that if one single sample is played across the entire keyboard, it will most likely sound very un-
natural when played too far from its original pitch, since the amount you can transpose a sound without negatively af-
fecting its timbre is very limited.

Setting up Key Ranges

You can adjust the key range of zones in a number of ways:

By Dragging the Zone Boundary Handles

1. Select the zone in the Key Range area.
2. Point and click on one of the handles that appear at each end.

368 NN-XT SAMPLER

 3. Drag the handle left/right.
Dotted lines extend from the edges of the zones up to the keyboard area. These lines give you a visual indication
of which keys the key range will encompass. There is also an alphanumerical indication at the bottom left of the
display.
Clicking and dragging the high key boundary handle of a zone with the default key range of C1 - C6...

...to change the key range to C1 - C2.

4. Repeat the procedure with as many zones as you wish, to create a complete key map.

By using the Lo Key and Hi Key controls

Directly below the key map area you will find the sample parameters. These are used for changing various parame-
ters that affect how the zones are played back. They can affect single or multiple selected zones. In the middle of the
sample parameters area are two knobs called “Lo Key” and Hi Key”.

These can be used for setting the low key and the high key of a zone’s key range.
1. Make sure the zone which you want to set the key range for is selected.
2. Use the Lo Key/Hi Key knobs to change the key range.
Check the display right above the knobs for an indication of the key. You can also keep an eye on the lines extend-
ing from the zone edges to the keyboard area.
Setting key ranges for multiple zones
You can set key ranges for multiple selected zones simultaneously. This can only be done by using the Lo and Hi Key
controls. It works as follows:

369 NN-XT SAMPLER

 D If you have three selected zones that each have different high keys and then turn the Hi Key knob, they will all
automatically get the same High Key value as the zone with edit focus.
In other words, if the selected zone with edit focus has the high key set to C4, and you change this to D4 by turn-
ing the Hi Key knob, all other selected zones will also be extended to D4 as the High Key.

D If any selected zone’s low key setting is higher than the edit focused zone’s high key before turning the Hi Key
knob, the zone range will be scaled down to one semitone, starting from the low key setting.
The high key can naturally never be set to a value lower than one semitone above its low key setting - the zone
would otherwise disappear!
D The inverse is also true - i.e. turning the Lo Key knob for several selected zones will apply the edit focused low
key setting to all selected zones.
A low key can never be set higher than one semitone below the high key in a zone, so if the edit focused zone has
a low key above the high key of another zone, the other zone will be scaled to the minimum semitone range.

By Dragging the Zone Boundary Handles on the Tab Bar

As previously described, the area directly below the keyboard area is called the tab bar. This shows the key range for
the currently selected zone, and also contains boundary handles.

Dragging a boundary handle on the tab bar.

These handles can be used much to the same effect as when dragging the boundary handles in the key map display.
However, the handles on the tab bar can change the key range of multiple zones at the same time.
The following applies:
• The tab bar shows the key range for the zone with edit focus.

370 NN-XT SAMPLER

 • Dragging the boundary handles for that zone will also simultaneously change the key range for a number of
surrounding zones if:
D The high key or low key (depending on which handle you drag) of the other zones are the same as the zone
with edit focus.
D The other zones are adjacent to the zone with edit focus.
! Note that it doesn’t matter whether the other zones are selected or not. They will be affected anyway.

In the example in the picture above, the zone in the middle has edit focus. Its left handle (the low key) is placed dif-
ferently from any of the other zones, but all of the zones have the same high key setting. This means that...
• Dragging the left handle will only move the low key position of the zone with edit focus (the pictures show be-
fore and after dragging).

• Dragging the right handle will move the high key position for all of the zones at the same time, since they all
have the same high key position (again, the picture shows before and after dragging).

371 NN-XT SAMPLER

 Moving Zones by Dragging the Zone Boxes
You can also move entire zones horizontally, thereby changing their key ranges.
1. Select all the zones you want to move.
You can move several zones simultaneously.
2. Point on any of the selected zones, and press the mouse button.
3. Drag left/right and release the mouse button.

Dragging multiple zones.

Moving Zones by Dragging in the Tab Bar

You can also move a zone by dragging anywhere between the zone boundary handles on the tab bar. When you do,
the surrounding zones will be affected just as when dragging the boundary handles in the tab bar (see above).
This can be used to “slide” a zone in relation to its surrounding zones, as the picture example below shows (before
and after dragging).

About the Lock Root Keys function

Normally, when you move zones (as described above), the root note of the zone(s) you move will change accordingly.
In other words, the zone(s) will be transposed. If this is not desired, you can activate the Lock Root Keys function prior
to moving the zone(s) by clicking on the button above the key map display.
Moving zones without changing their root notes can be used for some interesting effects, since it will completely
change the timbre of the sample(s) as they are played back.

372 NN-XT SAMPLER

 About the Solo Sample function

The Solo Sample function lets you play a selected sample over the entire keyboard and disregarding any velocity
range assigned to the sample. All other loaded samples are temporarily muted.
This is useful if you for example want to check how far up and down from its root key a sample can be played on the
keyboard before starting to sound “unnatural”. The solo sample function can therefore be useful as a guide for setting
up key ranges, as described in “Setting up Key Ranges”.
1. Select one and only one zone, or - if you have a selection of multiple zones - make sure the one you want to
hear has edit focus.
2. Activate Solo Sample by clicking on the button so that it lights up.
3. Play the MIDI keyboard

373 NN-XT SAMPLER

 Sorting Zones by Note
The Edit menu and the NN-XT context menu contains an item called “Sort Zones by Note”. This option lets you auto-
matically sort the selected zones in descending order according to their key ranges.
When you invoke this option, the selected zones will be sorted from top to bottom in the display starting with the one
with the lowest range.
Note however, that the sorting is done strictly on a group basis. That is, only zones that belong to the same group can
be sorted in relation to each other.

Before sorting and after.

If two zones have the same key range, they are sorted by velocity range.

374 NN-XT SAMPLER

 Setting Root Notes and Tuning
About the Root Key
All instrument sounds have an inherent pitch. When playing a sample of such a sound on the keyboard, the keys you
play must correspond to that pitch. For example, you may have recorded a piano playing the key “C3”. When you map
this onto the NN-XT key map, you must set things up so that the sampler plays back the sample at original pitch
when you press the key C3.
This is done by adjusting the root note.
D Many samples files from different sources already have a set root key in the file. If they do, the root key will be
correctly set automatically when you load the sample into a zone.
D However if the sample doesn’t have a root note stored in the file, (if you for example have recorded it yourself)
you will need to adjust it

Setting the Root Note manually

To set the root key for a zone, proceed as follows:
D Make sure the zone has edit focus (for example by clicking on it), and do one of the following:
D Use the knob marked “Root” in the sample parameter area below the display.
Turning it to the right will raise the pitch of the root key. The selected key is displayed alphanumerically directly
above the knob, and you can also look at the keyboard area for a visual indication (see below).

D Press [Ctrl](Win) or [Cmd](Mac) and click on the desired root key in the keyboard area.
The set root key is shaded so you can easily distinguish it.

Tuning samples manually

In addition to setting the root note, you may need to fine tune your samples, in order for them to match other instru-
ments and/or each other:
D Make sure the zone has edit focus (for example by clicking on it).
D Use the knob marked “Tune” in the sample parameter area.
This allows you to tune each sample in a key map by +/– half a semitone (-50 – 0 – 50).

375 NN-XT SAMPLER

 Setting the Root Note and Tuning using pitch detection
The NN-XT features a pitch detection function to help you set the root keys. This is useful if you for example load a
sample that you haven’t recorded yourself, and you don’t have any information about its original pitch.
Proceed as follows:
1. Select all the zones you want to be subject to pitch detection.
2. Pull down the Edit menu or the NN-XT context menu and select “Set Root Notes from Pitch Detection”.
The samples in all the selected zones will now be analyzed, and the detected root keys will automatically be set for
you.
! Note that for this to work properly, the samples must have some form of perceivable pitch. If it is sampled
speech, or a snare drum for example, it probably doesn’t have any discernible pitch.

About changing the pitch of samples

The procedures above should be used to make sure the samples are consistently pitched across the keyboard, and
that they all match an absolute reference (for example A 440 tuning).
If you need to tune the samples to match other material, or to get a certain effect (for example detuning two sounds
against each other for a chorus effect) you should use the Pitch section among the synth parameters, not the sample
tuning parameters.

Using Automap
The automap function can be used as a quick way of creating a key map, or as a good starting point for further ad-
justments of a key map.
Automap works under the assumption that you intend to create a key map for a complete instrument, for example a
number of samples of a piano, all at different pitches.
1. Load the samples you want to Automap.
Now you have three options:
D Trust that the root note information in the files is already correct.
D Manually adjust the root notes (and tuning) for all the samples.
D Use “Set Root Notes from Pitch Detection” to automatically set up the root notes.
2. Select all zones you want to automap.
3. Select Automap Zones from the Edit menu or the NN-XT context menu.
All the selected zones will now be arranged automatically in the following way:
D The zones will be sorted in the display (from top to bottom - lowest key first) according to the root keys.
D The zones will be assigned key ranges according to the root keys.
The key ranges are set up so that the split between two zones is exactly in the middle between the zones’ root
notes. If two zones have the same root key they will be assigned the same key range.

Automapping zones chromatically

D This Edit menu item will give each zone a key range of one semitone (i.e. one key), starting from C2 and up-
wards.
The function does not take root key into account. It simply places each selected sample on successive keys ac-
cording the position in the sample list (from the top down).

376 NN-XT SAMPLER

 Layered, crossfaded and velocity switched sounds
Creating layered sounds
You can set things up so that two or more zones have overlapping key ranges - either completely or partially. This way
you can create layered sounds, i.e. different samples that are played simultaneously when you press a key on your
keyboard.

In the picture above, you can see a set of piano samples at the top, mapped across the key range.
Below these are a set of string samples that also span the entire key range.
Whenever you play a key within this keyboard range, the sound produced will be a combination of the piano and the
string sample.
In addition, in the example above, the user has arranged the piano samples into one group and the string samples in
another. This is convenient since it allows for quick selection of the entire piano map, for example for balancing its
level against the strings.

About velocity ranges

When zones are set up so that their key ranges overlap – completely or partially – you can use velocity switching and
crossfading to determine which zones should be played back depending on how hard or soft you play on your MIDI
keyboard.
This is done by setting up velocity ranges, with or without crossfading.
Each time you press a key on your MIDI keyboard, a velocity value between 1-127 is sent to Reason. If you press the
key softly, a low velocity value is sent and if you press it hard, a high velocity value is sent.
This velocity value determines which samples will be played and which will not.
Let’s say for example that you’ve mapped three different zones across the same key range:
• Zone 1 has a velocity range from 1-40.
This means that the sample in it will be triggered by velocity values between 1-40.
• Zone 2 has a velocity range of 41-80.
The sample in this zone will be played back by velocity values between 41-80.

377 NN-XT SAMPLER

 • Zone 3 has a velocity range of 81-127.
The sample in this zone will be triggered by all velocity values above 80.

127

100

80

60

40
Zone 3
20
Zone 2
Velocity 0 Zone 1

Overlapping velocity ranges

Let’s change the values above slightly:
• Zone 1 has a velocity range from 1-60.
• Zone 2 has a velocity range of 41-100.
• Zone 3 has a velocity range of 81-127.

127

100

80

60

40
Zone 3
20
Zone 2
Velocity 0 Zone 1
Now, velocity values between 41 and 60 will trigger samples from both Zone 1 and Zone 2. Likewise, velocity values
between 81 and 100 will trigger sounds from Zone 2 and Zone 3.

378 NN-XT SAMPLER

 About full and partial velocity ranges
You can see which zones have modified velocity ranges in the key map display:
• Zones with a full velocity range (0 - 127) are only shown with an outline.
• Zones with any other velocity range are shown as striped.

The top zone has a full velocity range (1-127), and the lower zone has a partial velocity range (any other range), which is indicated by
stripes

Sorting Zones by velocity values

The Edit menu and the NN-XT context menu contain an item called “Sort Zones by Velocity”. This option lets you au-
tomatically sort the selected zones in the display in descending order according to their set low or high velocity val-
ues.
When you invoke this option, the selected zones will be sorted from top to bottom starting with the one with the high-
est “Lo Vel “value.
Note however, that the sorting is done strictly on a group basis. That is, only zones that belong to the same group can
be sorted in relation to each other.
If two zones have the same velocity range, they are sorted by key range.

Setting velocity range for a Zone

To set up a velocity range for a zone, proceed as follows:
1. Select one or more zones that you want to adjust.
2. Use the knobs marked “Lo Vel” and “High Vel” in the sample parameter area to set the desired low- and high
velocity values.

Adjusting the “Lo Vel” value for a zone.

“Lo vel” is the lowest velocity value that should trigger the sample in the zone - i.e. if a key is pressed so softly that the
velocity is lower than this value, the sample will not be played.
“Hi vel” is the highest velocity value that should trigger the sample, which means that if a key is pressed so hard that
the velocity is higher than this value, the sample will not be played.

About Crossfading Between Zones

At the bottom right in the sample parameter area are two knobs marked “Fade In” and “Fade Out”. These are primarily
used for setting up velocity crossfades for smooth transitions between overlapping zones. In order to set up cross-
fades you adjust the fade out and fade in values for the overlapping zones.

379 NN-XT SAMPLER

 Crossfading Between two Sounds
An example:
• Two zones are both set to play in the full velocity range of 1-127.
• Zone 1 has a fade out value of 40.
This means that this zone will play at full level with velocity values below 40, With higher velocity values, it will grad-
ually fade out.
• Zone 2 has a fade in value of 80.
This has the effect that as you play velocity values up to 80, this zone will gradually fade in. With velocity values
above 80, it will play at full level.

127

100

80

60

40

20
Zone 2
Velocity 0 Zone 1
Another example:
Crossfading can be used to only fade in or fade out a certain sound. One common example is to set things up so that
one sound plays the entire velocity range and another is faded in only at high velocity values.
• Zone 1 is set to play the entire velocity range with no crossfade.
• Zone 2 is set to play the velocity range 80 to 127, with a fade in value of 110.
This means that this zone will start fading in from velocity values 80 and will play at full level in the velocity range
110 to 127.

127

100

80

60

40

20
Zone 2
Velocity 0 Zone 1
This can be used for example to add a rimshot to a regular snare sound or a harder attack to a softer violin sample.

380 NN-XT SAMPLER

 Setting crossfading for a Zone
Manually
To set up a crossfade for a zone, proceed as follows:
1. Select one or more zones that you want to adjust.
2. Use the knobs marked “Fade In” and “Fade Out” in the sample parameter area, to set the desired values.

q You can change the values with finer precision by pressing [Shift] while turning the knobs, and you can reset
the standard values by pressing [Command] (Mac)/[Ctrl] (Windows) and clicking on the knobs.

Automatically
If you find it tedious to manually set up crossfades between zones, NN-XT can do it for you! The Edit menu and the
NN-XT context menu contain an item called “Create Velocity Crossfades”.
1. Set up the zones so that their velocity ranges overlap, as desired.
2. Select the zones.
You can select as many zones as you wish, not just one pair of overlapping zones.
3. Select “Create Velocity Crossfades” from the Edit menu.
NN-XT will analyze the overlapping zones and automatically set up what it deems to be appropriate fade in and
fade out values for the zones.
D This operation will not work if both zones have full velocity ranges.
At least one of the zones must have a partial velocity range (see “About full and partial velocity ranges”).
D This operation will not work if the zones are completely overlapping.

Using Alternate
About the Alternate function

At the bottom right in the sample parameters area is a knob marked “Alt”. It only has two states - On and Off. This is
used for semi-randomly alternating between zones during playback.
There are several practical uses for this. Here follows two examples:
• Layering several recordings of the same snare drum. By alternating between them you get a more natural rep-
etition.
• Layering string up- and down strokes. By alternating you get the realistic effect of switching between the two
directions of the stroke.

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 You can layer as many sounds as you will and the algorithm switches between them in a way that provides as little
repetition as possible.
To set up an alternating set of zones, proceed as follows:
1. Set up the zones so that they overlap completely or partially.
2. Select them all.
3. Set “Alt” to On for all the zones.
Now, the program will automatically detect how to alternate between the zones, depending on their overlap.

Sample parameters

The Sample parameter area is found below the screen. They allow you to adjust parameters for one or several se-
lected zones. Adjusting a parameter with multiple zones selected, will set the parameter to the same value for all se-
lected zones. Below follows a run-down of the various parameters:

Root Note and Tune

These parameters are described in “Setting Root Notes and Tuning”.

Sample Start and End

By turning the knobs you offset the start and end positions, so that they will play back more or less of a sample’s
waveform. Typical examples of use for this would be:
• Removing unwanted portions from samples.
This could be anything from noise to “dead air” at the beginning or end of a sample.
• To create variations out of a single sample.
These controls can be used to pick out any section of a recording for use as a sample.
• Together with velocity sample start control.
You can for example increase Sample Start and then apply negative velocity modulation to Sample Start. Then, the
harder you play the more you will hear of the attack portion of the sound.
q If you hold down [Shift] when adjusting these parameters, the adjustment is in single frames (samples).

Loop Start and End

A sample, unlike the cycles of an oscillator for example, is a finite quantity. There is a sample start and end. To get
samples to play for as long as you press down the keys on your keyboard, they need to be looped. For this to work
properly, you have to first set up two loop points which determine the part of the sample that will be looped.
The instrument samples in the sound banks included with Reason are already looped. The same will be true for most
commercial sample libraries. However, if you need to, you can use these controls to adjust the looping.
• The size and position of the loop – in the sample – is determined by two parameters, Loop Start (the beginning
of the loop) and Loop End (the end point of the loop).
• The NN-XT then keeps repeating the section between the Loop Start and Loop end until the sound has de-
cayed to silence.

382 NN-XT SAMPLER

 Play Mode
By using this knob you can select one of the following loop modes for each zone:
• FW
The sample in the zone will play only once, without looping.
• FW-LOOP
The sample will play from the sample start point to the loop end point, jump back to the loop start point and then
loop infinitely between the start and end loop points. This is the most common loop mode.
• FW - BW
The sample will play from the sample start point to the loop end point, then from the loop end point to the loop start
point (backwards), and then loop infinitely forwards-backwards between the start and end loop points.
• FW-SUS
This works like FW-LOOP with the exception that it will only loop as long as the key is held down. As you release
the key, the sample will play to the absolute end of the sample, that is beyond the boundaries of the loop.
This means that the sound may have a short natural release even if the release parameter is raised to a high value
(which is not true for “FW-LOOP”, where the release parameter always controls the length of the sound after the
key is released).
• BW
The sample will play only once - from the end to the beginning - without looping.

Lo Key and Hi Key

These parameters are described in “Setting up Key Ranges”.

Lo Vel and Hi Vel

These parameters are described in “About velocity ranges”.

Fade In and Fade Out

These parameters are described in “About Crossfading Between Zones”.

Alt
This parameter is described in “About the Alternate function”.

Out
The NN-XT features eight separate stereo output pairs (see “Audio Output”). For each zone, you can decide which of
these output pairs to use. Thus, if you have created a key map consisting of eight zones, each of these can have a
separate stereo output from NN-XT, and can then be routed to a separate mixer channel if you so wish.
D To select which output a selected zone should be directed to, use the knob marked “Out” in the sample param-
eter area.
The output pairs are indicated above the button.
! Note that you still have to route the outputs the way you want them on NN-XT’s back panel. If you assign a
zone to an output pair other than 1-2 (which is the default) no connections or auto routing are made. You have
to do that manually.

383 NN-XT SAMPLER

 A Stereo example
One possible way of utilizing this would be to create a drum kit. In this case you could load up to eight different stereo
drum samples, assign them to separate outputs, route each to a separate mixer channel and then use the mixer to set
levels and pan, add send effects etc.

Using a stereo output as two mono outputs

If, on the other hand, you are using mono samples, you can use one stereo pair as a two separate outputs, effectively
giving you a total of 16 separate outputs.
1. Assign two zones to the same output.
2. Us the Pan control to pan one of the zones hard left and the other hard right.
3. Connect each of the two outputs in the stereo pair to a separate mixer channel.

Group parameters

The group parameters are located at the top left on the remote editor panel. These are parameters that in various
ways are directly related to playing style.
Group parameters apply to a group, that is they are settings that are shared by all zones in a group.
D To make adjustments to one group, select one or more zones that belong to the group, and adjust the param-
eter on the front panel.
D To set several groups to the same value, select at least one zone in each group you want to adjust, and adjust
the parameter on the front panel.

Key Poly
This setting determines the number of keys that you can play simultaneously (the polyphony). The maximum number
is 99 and the minimum is 1, in which case the group will be monophonic.
Users of other samplers may want to note that the polyphony often means setting the number of voices that should
be able to play. The NN-XT is different in this aspect, since the polyphony setting instead determines the number of
keys, regardless of how many voices each key plays.

The Group Mono button

The Group Mono button beside the Key Poly section can be used to quickly set a group to play monophonically re-
gardless of the polyphony setting. E.g. if you have a group with open and closed hi hats, you can switch this on so that
an open hi hat is automatically muted when you play a closed hi hat.
Group Mono overrides the Key Poly setting - except when playing the same note.

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 So you can play your open hi-hat repeatedly without the sound cutting itself off. When you play the closed hi-hat, this
cuts off the open hi-hat.
Note that activating this button is not the same as setting polyphony to 1. E.g., it can not be used for Legato or mono
Retrig (see “Legato and Retrig”).

Legato and Retrig

Legato
Legato works best with monophonic sounds. Set Key Poly (see above) to 1 and try the following:
D Hold down a key and then press another key without releasing the previous.
Notice that the pitch changes, but the envelopes do not start over. That is, there will be no new “attack”.
D If Key Poly is set to more voices than 1, Legato will only be applied when all the assigned keys are “used up”.
For example, if you had a polyphony setting of “4” and you held down a 4 note chord, the next note you played
would be Legato. Note, however, that this Legato key will “steal” one of the keys in the 4 note chord, as all the as-
signed keys were already used up!

Retrig
Retrig is the “normal” setting for playing polyphonic patches. That is, when you press a key without releasing the pre-
vious, the envelopes are triggered, like when you release all keys and then press a new one. In monophonic mode,
Retrig has an additional function; if you press a key, hold it, press a new key and then release that, the first note is
also retriggered.

LFO 1 Rate
This is used for controlling the rate of LFO 1 if it is used in “Group Rate” mode. In that case, this knob will take pre-
cedence over the rate parameter in the LFO 1 section. See “The LFOs” for detailed information about this.

Portamento
This is used for controlling portamento - a parameter that makes the pitch glide between the notes you play, rather
than changing the pitch instantly as soon as you hit a key on your keyboard. By turning this knob you set how long it
should take for the pitch to glide from one note to the next as you play them.
In legato mode, there will only be any portamento when actually playing legato (tied) notes.
With the knob turned all the way to the left, portamento is disabled.

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 Synth parameters
The Modulation controls

As previously described, the Modulation wheel (and the External Control wheel) can be used for controlling various
parameters. These controls allow you to define which parameters the wheels should modulate and to what extent.
D Below each of the knobs are the letters “W” and “X”.
These are used for selecting the source that should control the parameter, and represent the “Modulation Wheel”
and the “External Control wheel” respectively.
D By clicking on any of the letters, you decide which source should control the parameter.
You can select either, both or none. When a letter is “lit”, the corresponding source is set to control the parameter.
D By turning the knobs, you decide how much the modulation and/or external control wheel should modulate
the corresponding parameter.
Note that all of the control knobs are bi-polar, which means that they can be set to both positive and negative values.
Positive values are set by turning the knobs to the right, and negative values are thus set by turning the knobs to the
left:
• Setting them to positive values means that the value of the controlled parameter will be raised if the source
wheel is pushed forward.
• Setting them to negative values means that the value will be lowered when a wheel is pushed forward.
• Keeping the knobs in the center position means that no modulation control is applied.
There is one exception to these rules, and that is the LFO 1 Amt control, which works in a slightly different way. See
below for more information about this.
The following parameters can be modulated:

F.Freq
This sets modulation control of the Filter’s cutoff frequency (see “The Filter section”).

Mod Dec
This sets modulation control of the Decay parameter in the Modulation Envelope (see “The Modulation Envelope”).

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 LFO 1 Amt
This determines how much the amount of modulation from LFO 1 is affected by the Modulation wheel and/or the Ex-
ternal Controller wheel. It does this by “scaling” the amounts set with the three destination knobs in the LFO 1 section
(Pitch, Filter and Level, see “The LFOs”). We’ll explain this with an example:
To use the Modulation Wheel to increase pitch modulation (vibrato), proceed as follows:
1. Turn the Mod Wheel all the way down, so that no modulation is applied.
2. Activate the “W” button for LFO 1 Amt in the Modulation section.
3. Set the corresponding knob to “12 o’clock” (zero).
4. Set up LFO 1 so that as much vibrato is applied as you want it to be when the Modulation wheel is turned all
the way up.
5. Increase LFO 1 Amt until you hear as much vibrato as you want it to be when the wheel is turned all the way
down.
If you turn LFO 1 Amt all the way up, there will be no vibrato at all when the wheel is all the way down.
To instead use the Modulation wheel to decrease vibrato, process as follows:
1. Turn the Mod Wheel all the way down, so that no modulation is applied.
2. Activate the “W” button for LFO 1 Amt in the Modulation section.
3. Set the corresponding knob to “12 o’clock” (zero).
4. Set up LFO 1 so that as much vibrato is applied as you want it to be when the Modulation wheel is turned all
the way down.
5. Turn the Modulation wheel all the way up.
6. Decrease LFO 1 Amt until you hear as much vibrato as you want it to be when the wheel is turned all the way
up.
If you turn LFO 1 Amt all the way down, there will be no vibrato at all when the wheel is all the way up.

F.Res
This sets modulation control of the Resonance parameter in the Filter (see “The Filter section”).

Level
This sets the amount of amplitude envelope modulation of each zone’s level. The level set here will be the level of the
highest point of the Amp Envelope.

LFO 1 Rate
This sets modulation control of the Rate parameter in LFO 1 (see “The LFOs”).

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 The Velocity controls

Velocity is used for controlling various parameters according to how hard or soft you play notes on your keyboard. A
typical use of velocity control is to make sounds brighter and louder if you strike a key harder. By using the knobs in
this section, you can control if and how much the various parameters will be affected by velocity.
Just like the modulation controls, all of the velocity control knobs are bi-polar, and can be set to both positive and
negative values.
• Setting them to positive values means that the value of the controlled parameter will be raised the harder you
play.
• Setting them to negative values means that the value will be lowered the harder you play.
• Keeping the knobs in the center position means that no velocity control is applied.
The following parameters can be velocity controlled:

F.Freq
This sets velocity control of the Filter’s cutoff frequency (see “The Filter section”).

Mod Dec
This sets velocity control of the Decay parameter in the Modulation Envelope (see “The Modulation Envelope”).

Level
This sets velocity control of the Amp Envelope.

Amp Env Attack

This sets velocity control of the Attack parameter in the Amplitude Envelope (see “The Amplitude Envelope”).

Sample Start
This sets velocity control of the Sample Start parameter (see “Sample Start and End”), so that it will be offset for-
wards or backwards, according to how hard or soft you play.
This allows you to control how much of the attack portion of the sample you hear when playing harder or softer.
To be able to make use of negative values for this parameter, you must increase the sample parameter Sample Start.

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 The Pitch section

This section contains various parameters related to controlling the pitch, or frequency, of the zones.

Pitch Bend Range

This lets you set the amount of pitch bend, i.e. how much the pitch changes when your turn the pitch bend wheel fully
up or down. The maximum range is +/- 24 semitones (2 Octaves).

Setting the pitch

Use the three knobs marked “Octave”, “Semi” and “Fine” to change the pitch of the sample(s):
• Octave
This changes the pitch in steps of one full octave. The range is -5 – 0 – 5.
• Semi
This lets you change the pitch in semitone steps. The range is -12 – 0 – 12 (2 octaves).
• Fine
This changes the pitch in cents (hundredths of a semitone). The range is -50 – 0 – 50 (down or up half a semi-
tone).

K. Track
This knob controls Keyboard Tracking of the pitch.
• In the center position, each key represents a semitone This is the normal setting.
• When turned all the way down, all keys play the same pitch. This can be useful for percussion like timpani
where you might want to play the same pitch from a range of keys.
• When turned all the way up, each key on the keyboard shifts the pitch one octave.

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 The Filter section

Filters can be used for shaping the character of the sound. The filter in NN-XT is a multimode filter with six different
filter types.
D To activate/deactivate the filter, click the On/Off button in the top right corner.
When the filter is activated, the button is lit.

Filter mode
To select a filter mode, either click the Mode button in the bottom right corner or click directly on the desired filter
name so that it lights up:
• Notch
The notch filter is used for cutting off frequencies in a narrow frequency range around the set cutoff frequency,
while letting the frequencies below and above through.
• HP 12
This is a highpass filter with a 12 dB/Octave roll-off slope. A highpass filter cuts off low frequencies and lets high
frequencies pass. That is, frequencies below the cutoff frequency are cut off and frequencies above it pass
through.
• BP 12
This is a bandpass filter with a 12 dB/Octave roll-off slope. A bandpass filter could be viewed as the opposite of
a notch filter. It cuts off both the high and the low frequencies, while frequencies in the band range pass through.
• LP 6
This is a lowpass filter with a gentle, 6 dB/Octave slope. A lowpass filter is the opposite of a highpass filter. It lets
the low frequencies through and filters out the high frequencies. This filter has no Resonance.
• LP 12
This is a lowpass filter with a 12 dB/Octave roll-off slope.
• LP 24
This is a lowpass filter with a fairly steep roll-off slope of 24 dB/Octave.

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 Filter controls
The following filter controls are available:
• Freq
This is used for setting the filter cutoff frequency. The cutoff frequency determines the limit above or below which
frequencies will be cut off depending on the selected filter type. In the case of a lowpass filter for example, fre-
quencies below the cutoff frequency will be allowed to pass through, while frequencies above it will be cut off. The
farther to the right you turn the knob, the higher the cutoff frequency will be.
q It is very common to modulate filter frequency with the modulation envelope, as described in “The Modulation
Envelope”.
• Res
Technically, this knob controls feedback of the output signal from the filter, back to its input. Acoustically it empha-
sizes frequencies around the cutoff frequency. For a lowpass filter for example, increasing Res will make the
sound increasingly more hollow until the sound starts “ringing”. If you set a high value for the Res parameter and
then vary the filter frequency, this will produce a classic synthesizer filter sweep.
For the notch and bandpass filter types, the Resonance setting instead adjusts the width of the band. That is, the
higher the resonance setting, the narrower the band will be where frequencies are cut off (notch) or let through
(Bandpass).
• K. Track
This lets you activate and control keyboard tracking of the filter frequency. If keyboard tracking is activated, the set
cutoff frequency of the filter will change according to the notes you play on your keyboard. That is, if you play notes
higher up on the keyboard, the filter frequency will be raised and vice versa.
When the knob is set to its center position, filter frequency is adjusted so that the harmonic content remains con-
stant across the keyboard.
Keyboard tracking is deactivated by default (the knob all the way to the left). This means that the filter frequency
will remain unchanged regardless of where on the keyboard you play.

The Modulation Envelope

The Modulation Envelope parameters let you control how certain parameters, or destinations, should change over
time - from the moment a note is struck to the moment it is released again.
The destinations you can use are:
• Pitch
• Filter frequency

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 Parameters
The following are the available controlling parameters:
• Attack
When you press a key on your keyboard, the envelope is triggered. The attack parameter then controls how long
it should take before the controlled parameter (pitch or filter) reaches the maximum value, when you press a key.
By setting attack to a value of “0”, the destination parameter would reach the maximum value instantly. By raising
the attack parameter, the value will instead slowly “slide” up to its maximum.
• Hold
This is used for deciding how long the controlled parameter should stay at its maximum value before starting to
decrease again. This can be used in combination with the Attack and Decay parameters to make a value reach its
maximum level, stay there for a while (hold) and then start dropping gradually down to the sustain level.
• Decay
After the maximum value for a destination has been reached and the Hold time has expired, the controlled param-
eter will start to gradually drop down to the sustain level. How long it should take before it reaches the sustain level
is controlled with the Decay parameter. If Decay is set to “0”, the value will immediately drop down to the sustain
level.
• Sustain
The Sustain parameter determines the value the envelope should drop back to after the Decay. If you set Sustain
to full level however, the Decay setting doesn’t matter since the value will never decrease.
A combination of Decay and Sustain can be used for creating envelopes that rise up to the maximum value, then
gradually decrease to, and stay on a level somewhere in-between zero and maximum.
• Release
This works just like the Decay parameter, with the exception that it determines the time it takes for the value to fall
back to zero after the key is released.
• Delay
This is used for setting a delay between when a note is played and when the effect of the envelope starts. That is,
the sound will start unmodulated, and the envelope will kick in after you have kept the key(s) pressed down for a
while. Turn the knob to the right to increase the delay time. If the knob is set all the way to the left, there will be no
delay.
• Key To Decay
By using this, you can cause the value of the Decay parameter (see above) to be offset depending on where on
your keyboard you play. If you turn the knob to the right the decay value will be raised the higher up you play, and
turning the knob to the left will lower the decay value the higher up you play. With the knob in the center position,
this parameter is deactivated.

Destinations
The following are the available Mod Envelope destinations:
• Pitch
This will make the envelope modulate the pitch, as set in the Pitch section (see “The Pitch section”). Turn the knob
to the right to raise the pitch and to the left to lower the pitch. In the middle position, pitch will not be affected by
the envelope.
• Filter
This will make the envelope modulate the cutoff frequency of the Filter (see “The Filter section”). Turn the knob to
the right to increase the frequency and to the left to lower the frequency. In the middle position, the envelope will
have no effect on the cutoff frequency.

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 The Amplitude Envelope

The Amplitude Envelope parameters let you control how the volume of a sound should change over time - from the
moment a note is struck to the moment it is released again.

Parameters
Most of the Amplitude Envelope parameters are identical to those of the Modulation Envelope. So for a detailed de-
scription of the following parameters, please refer to the modulation envelope section in “The Modulation Envelope”:
• Attack
• Hold
• Decay
• Sustain
• Release
• Delay
• Key To Decay
The following are the parameters that are unique for the Amp Envelope section:
• Level
This knob sets the level of the zone. Turn it to the right to raise the level.
• Spread and Pan modes
These two parameters are used for controlling the stereo (pan) position of the sound. The Spread knob deter-
mines the sound’s width in the stereo image (how far left – right the notes will be spread out). If this is set to “0”,
no spread will take place. The Mode selector switch is used for choosing which type of spread you want to apply:

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 | Mode | Description
Key This will make the pan position shift gradually from left to right, the higher up on the keyboard you play.
Key 2 This will make the pan position shift from left to right and then back again from right to left in a sequence of eight keys.
Playing 4 adjacent semitones thus makes the pan position gradually go from left to right. The next 4 higher semitone notes
will then change the pan position from right to left in the same way, and this cycle will then be repeated.
Jump This will make the pan position jump between left and right each time a note is played.

• Pan
This controls the stereo balance of the output pair to which a zone is routed. In the middle position, the signal ap-
pears equally strong on the left and right channel in a stereo pair. By turning the knob to the left or right, you can
change the stereo balance.
Note that if you for instance turn the Pan knob all the way to the left, you cause the signal to be output from the left
channel of the stereo pair only.
You can use this to treat a stereo output as two independent mono outputs, if required.
See “Out” for information on routing zones to output pairs.

The LFOs

NN-XT features two Low Frequency Oscillators - LFO 1 and LFO 2. “Normal” oscillators generate a waveform and a
frequency, and produce sound. Low frequency Oscillators on the other hand, also generate a waveform and a fre-
quency, but there are two major differences:
• LFOs only generate sounds of a low frequency.
• LFOs don’t produce sound, but are instead used for modulating various parameters.
The most typical use of an LFO is to modulate the pitch of a sound (generated by an oscillator or - in the case of NN-
XT - a sample), to produce vibrato.

About the Difference between LFO 1 and LFO 2

There are two fundamental differences between LFO 1 and LFO 2:
• LFO 2 is always key synced, that is, each time you press a key, the LFO waveform starts over from scratch. LFO
1 can be switched between key synced and non-key synced modes.
• LFO 2 only has one waveform, triangle.
The following parameters are available for the LFOs:

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 Rate (LFO 1 and 2)
This knob controls the frequency of the LFO. For a faster modulation rate, turn the knob to the right. The Rate knob
of LFO 1 is also used for setting the timedivision when synchronizing the LFO to the song tempo (see below).

Delay (LFO 1 and 2)

This can be used for setting a delay between when a note is played and when the LFO modulation starts kicking in
(gradually). This way, you can make the sound start unmodulated, and then have the LFO modulation start after you
have kept the key(s) pressed down for a while.
Turn the knob to the right to increase the delay time.

Mode (LFO 1 only)

This lets you set the “operation mode” for the LFO. Click the button to switch between the available modes:
• Group Rate
In this mode, the LFO will run at the rate set for its group in the group section, rather than at the rate set here (see
“Group parameters”). This way, all zones in the group will have the exact same modulation rate.
• Tempo Sync
In this mode, the LFO will be synchronized to the song tempo, in one of 16 possible time divisions.
! When tempo sync is activated, the Rate knob is used for selecting the desired timedivision. Turn the Rate knob
and observe the tool tip for an indication of the timedivision.
• Free Run
In free run mode, the LFO simply runs at the rate set with the Rate parameter. Furthermore, if Key Sync is deacti-
vated, the modulation cycle will not be retriggered each time you press a key - it will run continuously.

Waveform (LFO 1 only)

Here, you select which type of waveform should be used for modulating the destination parameters.
Click the button to switch between the following waveforms:

| Waveform | Description
Triangle This is a smooth waveform, suitable for normal vibrato.

Inverted Sawtooth This produces a “ramp up” cycle. If applied to an oscillator’s frequency, the pitch would sweep up,
after which the cycle immediately starts over.
Sawtooth This produces a “ramp down” cycle, the same as above but inverted.

Square This produces cycles that abruptly change between two values, usable for trills etc.

Random Produces random stepped modulation to the destination. Some vintage analog synths called this
feature “sample & hold”.
Soft Random The same as above, but with smooth modulation.

! LFO 2 always uses a triangle waveform.

Key Sync (LFO 1 only)

By activating key sync, you “force” the LFO to restart its modulation cycle each time a key is pressed.
! Note that LFO 2 always uses Key Sync.

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 Destinations for LFO 1
The following parameters can be modulated by LFO 1:
• Pitch
This will make the LFO modulate the pitch, for vibrato, trills, etc. It can be set to -2400 – 0 – 2400 cents which
equals 4 octaves. The set pitch will change up and down by this amount, with each modulation cycle. Turning the
knob to the right will make the modulation cycle start above the set pitch, while turning it to the left will invert the
cycle. Keeping this in the middle position means that the pitch will not be affected by the LFO.
• Filter
This will make the LFO modulate the cutoff frequency of the Filter, for auto-wah effects, etc. The positive/negative
effect is the same as for pitch.
• Level
This will make the LFO modulate NN-XT’s output level, for tremolo effects, etc. The positive/negative effect is the
same as for pitch.

Destinations for LFO 2

The following parameters can be modulated by LFO 2:
• Pan
This makes the LFO modulate the pan position of a zone. The sound will move back and forth in the stereo field.
Turning the knob to the left makes the sound move from left to right, and turning it to the right thus makes it move
from right to left. The middle position provides no modulation at all.
• Pitch
Just like for LFO 1 (see above), this makes LFO 2 modulate the pitch. The range is also the same as for LFO 1.

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 Connections
On the back panel of NN-XT are a number of connectors. Many of these are CV/Gate related. Using CV/Gate is de-
scribed in the chapter “Routing Audio and CV”.

Sequencer Control
The Sequencer Control CV and Gate inputs allow you to play the NN-XT from another CV/Gate device (typically a
Matrix or a Redrum). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note
on/off along with velocity.

Modulation Input
These control voltage (CV) inputs (with associated voltage trim pots), can modulate various NN-XT parameters from
other devices. These inputs can control the following parameters:
• Oscillator Pitch
• Filter Cutoff Frequency
• Filter Resonance
• LFO 1 Rate
• Master Volume
• Pan
• Modulation Wheel
• Pitch Wheel

Gate Input
These inputs can receive a CV signal to trigger the following envelopes:
• Amplitude Envelope
• Modulation Envelope
Note that connecting to these inputs will override the normal triggering of the envelopes. For example, if you connect
a Matrix Gate Out to the Gate In Amp Envelope, you would not trigger the amp envelope by playing notes, as this is
now controlled by the Matrix Gate Out. In addition you would only hear the Gate Out triggering the envelope for the
notes that you hold down.

Audio Output
There are 16 audio output jacks on the NN-XT’s back panel - eight separate stereo pairs. When you create a new
NN-XT device, the first output pair (1L & 2R) is auto-routed to the first available outputs in the I/O device.
The other output pairs are never automatically routed. If you wish to use any of the other output pairs, you have to
manually connect them to the desired device. The basics on Routing is described in “Routing Audio and CV”.

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 ! Note that when you use any other output pair than the first, you also have to route one or more zones to it if
you want it to actually output sound, since all zones by default are routed to outputs 1 & 2. How to route zones
to other outputs is described in the “Out” section.

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Chapter 22
NN-19 Sampler
 Introduction
The NN-19 is a sampler capable of playing back - but not editing - sound files.
The program comes with numerous ready-made sample patches, covering all kinds of instrument types. In addition to
this there are plenty of single samples that can be used for creating your own patches.
There are also plenty of relatively inexpensive (and even free) audio editing software for both the Windows and the
macOS platforms, that will allow you to both record audio (via your computers or audio cards audio inputs), and to edit
the resulting audio file. Virtually every software that is capable of this, can create sound files which can be loaded di-
rectly into the NN-19.
Also, there are thousands of high quality sample libraries available, covering every conceivable musical style or direc-
tion ranging from professionally recorded orchestral samples to esoteric electronic noises.
! Please, note that this device is not available in Reason Lite Rack Plugin.

General sampling principles

Background
Before a sound can be used by a sampler, it must be converted to a digital signal. Hardware samplers and computer
audio cards provide audio inputs that can convert the analog signal to digital, by the use of an “A/D Converter” (ana-
log to digital). This “samples” the signal at very short time intervals and converts it to a digital representation of the an-
alog signal’s waveform. The sample rate and the bit depth of this conversion determines the resulting sound quality.
Finally the signal is passed through a digital to analog converter (D/A) which reconstructs the digital signal back to
analog, which can be played back.

Multisampling vs. single samples

Most of the included NN-19 patches are made up of a collection of several samples. This is because a single sam-
pled sound only sounds natural within a fairly narrow frequency range. If a single sample is loaded into an empty NN-
19, the sample will be playable across the whole keyboard. The pitch (frequency) of the original sample (called root-
key) will be automatically placed on the middle C key (C3).
Note that this has nothing to do with the actual pitch the sample itself produces! It may not even have a pitch as such,
it could be the sound of someone talking for example.
If you play any single sample about two octaves above or below its root key, it will most likely sound very “unnatural”.
In the case of it actually being a sample of someone talking, playing two octaves up will make the talking voice sam-
ple sound squeaky, short and most likely unintelligible. Two octaves down the voice will sound something like a
drawn-out gargle.
Thus, the range that most samples can be transposed without sounding unnatural is limited. To make a sampled pi-
ano, for example, sound good across the whole keyboard, you need to first have made many samples at close inter-
vals across the keyboard, and then define an upper and lower range for each sample, called a Key Zone. All the
keyzones in the piano sample patch then make up a Key Map.
How to create key zones is described in “About Key Zones and samples”.
To sample real instruments accurately requires a lot of hard work. Firstly, you need the original instrument, which
should be in perfect working order. For acoustic instruments you need a couple of good microphones, a mixer or
other device with high quality microphone preamps, and a room with good acoustics. You need to be meticulous
when recording the different samples, so that levels are smooth and even across the range etc.
Fortunately Reason provides a wide range of high quality multisampled instruments, so much of this hard work has
already been done for you.

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 In our experience, most people don’t use samplers only for playing sampled versions of “real” instruments. Very often,
single “stand alone” or single samples are used. Maybe you wish to use different sounds for every key zone. Or you
could have complete chorus and verse vocals plus variations assigned to several “one note” key zones. Or use sam-
ples of different chords that play rhythmic figures to the same tempo, and use these to build song structures etc. The
possibilities are endless. When you use samples in this way, the keys on your keyboard that play the samples do not
necessarily correspond to pitch at all, the keys are simply used to trigger the samples.

About audio file formats

The audio file format support differs depending on which computer OS you are using.
The NN-19 can read audio files in the following formats:
• In Windows:
.wav, .aif, .mp3, .aac, .m4a and .wma.
• In macOS:
.wav, .aiff, .3g2, .3gp, .mp1, .mp2, .mp3, .mpeg, .mpa, .snd, .au, .sd2, .ac3, .aac, .adts, .amr, .caf, .m4a .m4r and .mp4.
• SoundFonts (.sf2)
SoundFonts are an open standard for wavetable synthesized audio, developed by E-mu systems and Creative
Technologies.
• REX file slices (.rx2, .rex, .rcy)
REX files are music loops created in the ReCycle program. The NN-19 lets you either load REX files as patches
or separate slices from REX files as individual samples.
• Any sample rate and practically any bit depth.

About the Sample Patch format

Reason’s Sample Patch format (.smp), is based on either Wave or AIFF files, but includes all the NN-19 associated
parameter settings as well.
• The audio files may be stereo or mono. Stereo audio files are shown with a “S” symbol beside its name in the
display.

Loading a Sample Patch

When you create a new NN-19 device, it is automatically loaded with a default patch. If you want to start from
scratch, with no samples loaded, you can select “Reset Device” from the context menu or Edit menu. For NN-19 to
produce sound, you need to load either a sample patch, or a sample.
A patch contains “everything”. All the samples, assigned key zones, and associated panel settings will be loaded.
Loading a sample patch is done using the Browser, just like in all other devices that use Patches.
1. Click the Browse Patch button on the front panel to set browse focus to the NN-19 device.
2. Navigate to the folder that contains the NN-19 patch you wish to load, select it and click Load in the Browser.
D Alternatively, drag an NN-19 patch from the Browser and drop it on the NN-19 device in the rack.
The panel is dimmed in orange and the Patch Replace symbol appears in the center.

Loading REX Files as Patches

REX files are music loops created in the ReCycle program. In Reason, REX files are primarily used in the Dr. Octo
Rex loop player, but they can be used in the NN-19 as well. Possible extensions are “.rx2”, “.rcy” and “.rex”.
1. Click the Browse Patch button on the front panel to set browse focus to the NN-19 device.
2. Navigate to the folder that contains the REX loop you wish to load, select it and click Load in the Browser.

401 NN-19 SAMPLER

 D Alternatively, drag a REX loop from the Browser and drop it on the NN-19 device in the rack.
The panel is dimmed in orange and the Patch Replace symbol appears in the center.
When loading a REX file, each slice in the file is assigned to one key, chromatically, starting from C1. All parame-
ters are set to their default settings.
When using REX files in the Dr. Octo Rex loop player, it is possible to make a track play the slices in order to recreate
the original loop. To do the same in the NN-19 requires a few extra steps.
1. Use the Browser to load the REX file into an NN-19 sampler.
2. Create a Dr. Octo Rex loop player and load the same REX file into o Loop Slot of this device.
3. Use the “Copy Loop To Track” feature on the Dr. Octo Rex to create playback data (a group) on the track as-
signed to the Dr. Octo Rex.
4. Move that group to the track that plays the NN-19 and play it back from there.
5. Delete the Dr. Octo Rex loop player.

About Key Zones and samples

Loading a Sample into an empty NN-19
1. Create an NN-19 device and select “Reset Device” from the context menu or from the Edit menu.
2. Click on the Browse Sample button.
This is located above the keyboard display to the left.

q When you browse samples, you can preview them before loading using the Play button in the Browser. If you
select the “Autoplay” function, the samples play back once automatically when selected.
3. Select a sample in the Browser and click the Load button in the Browser to load it.
D Alternatively, drag a sample file from the Browser and drop it on the NN-19 device in the rack.
The panel is dimmed in blue and a Sample Replace symbol appears in the center.
When you load the first sample into an empty NN-19, this will be assigned a key zone that spans the entire range of
the keyboard, and the default Init Patch settings will be used.

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 Below the keyboard, the range, sample name, root key, tuning, level and loop status of the current key zone is dis-
played, each with a corresponding knob.

The light blue strip above the keyboard indicates the currently selected key zone, which is in this case the full range of the keyboard.
The inverted note on the keyboard indicates the “root key” of the sample. All samples contain a root key, tuning and
level setting. If NN-19 is empty, a sample will have its root key placed on the middle “C” (C3) key.
4. If desired, click on the keyboard to change the root key.
! You can audition a loaded sample patch or sample by holding down [Option] (Mac)/[Alt] (Windows) and click-
ing on a key in the Keyboard display. The mouse will take on the shape of a speaker symbol to indicate this.

Loading SoundFont samples

The SoundFont format was developed by E-mu systems in collaboration with Creative Technologies. It is a standard-
ized data format containing wavetable synthesized audio and information on how it should be played back in wavet-
able synthesizers - typically on audio cards. The SoundFont format is an open standard so there is a vast amount of
SoundFont banks and SoundFont compatible banks developed by third parties.
The samples in a SoundFont are stored hierarchically in different categories: User Samples, Instruments, Presets etc.
The NN-19 allows you to browse for and load single SoundFont samples, but not entire soundfonts.
1. Click the Browse Sample button, select a SoundFont file (.sf2) in the Browser and open it.
The Browser opens the SoundFont and displays the folders within it.
2. Select the folder “Samples” and open it.
This folder contains a number of samples which can be loaded like any other sample.
3. Select the desired sample and load it by clicking the Load button in the Browser.
The sample is loaded and assigned a key zone range that spans the entire keyboard. You can now make settings
for it as with any other sample.
D Alternatively, drag a sample file from the Browser and drop it on the NN-19 device in the rack.
The panel is dimmed in blue and a Sample Replace symbol appears in the center.

Loading REX slices as samples

A slice is a snippet of sound in a REX File.
1. To import a REX slice, click the Browse Sample button (see above), browse to a REX file and open it as if it
was a folder.
The Browser will then display the slices as files inside that “folder”.

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 2. Select the desired REX slice and load it by clicking the Load button in the Browser.
The REX slice is loaded and assigned a key zone range that spans the entire keyboard. You can now make set-
tings for it as with any other sample.
D Alternatively, drag REX slice from the Browser and drop it on the NN-19 device in the rack.
The panel is dimmed in blue and a Sample Replace symbol appears in the center.
In the rest of this manual, when we refer to importing samples, all that is said applies to REX slices as well.

Creating Key Zones

A “key zone” is a range of keys, that plays a sample. All key zones together make up a “key map”.
To create a new key zone, the following methods can be used:
D Select “Split Key Zone” from the Edit or context menus.
This splits the currently selected key zone in the middle. The new zone is the upper half of the split, and is empty.
The dividing point has a “handle” above it, see “Setting the Key Zone Range” below for a description.
D By [Alt]/[Option]-clicking at a point just above the key zone strip, a new empty key zone is created.
The point where you click becomes the lower limit (or boundary) for the original key zone, and the upper limit for
the new key zone.

The new empty key zone gets selected upon creation.

Selecting Key Zones

Only one key zone can be selected at a time. A selected key zone is indicated by a light blue (as opposed to dark
blue) strip above the keyboard in the display. There are two ways you can select key zones:
D By clicking on a non-selected key zone in the display.
D By activating the “Select Key Zone via MIDI” button.
Playing a note belonging to a non-selected key zone from your MIDI keyboard, will select the key zone it belongs
to.

Setting the Key Zone Range

! Key zones cannot overlap.
When you adjust the boundaries of a key zone, the surrounding boundaries are automatically adjusted accordingly.
You can change the key zone range in the following ways:

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 D By dragging the “handle(s)” which divides the key zones, you can change the range of the selected key zone.
In the case of having two key zones split in the middle, you could thus change the lower limit for the upper (new)
key zone and the upper limit for the original key zone.

D By using the “Lowkey” and “Highkey” knobs to set a lower and upper range, respectively.

Deleting a Key Zone

D To delete a key zone, select it and then select “Delete Key Zone” from the Edit menu.

About Key zones, assigned and unassigned samples

When you load samples and rearrange your key mapping, you will often end up with samples that are not assigned to
any key zone. In the following texts we refer to the samples as follows:
• Assigned samples are samples that are currently assigned to one ore more key zones.
• Unassigned samples are samples that reside in the sample memory, but that are currently not assigned to any
key zone.

Adding sample(s) to a Key Map

If the sample hasn’t been loaded yet
1. Select a key zone.
This can be empty, or contain a sample - it doesn’t matter for now.
2. Use the Browser to add one, or several (see below), sample(s).
The following will happen:

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 D If the zone contained a sample prior to loading, this will be replaced, both in the zone and in the sample mem-
ory, unless the sample was also used by another key zone, in which case it will be kept.
D If you loaded several samples, one of the samples will be assigned to the key zone, and the other samples will
be loaded but remain unassigned.

If the sample is already loaded but unassigned

1. Select a key zone.
This can be empty, or contain a sample - it doesn’t matter for now.
2. Use the Sample knob to dial in the sample you want the key zone to play.

The Sample knob.

Setting the Root Key

Once you have defined a key zone, and added a sample, you should set the root key for the sample.
D Select the key zone the sample belongs to, and click on the key you wish to set the root key to.
Which key to select is normally determined by the pitch of the sample. For example if the sample plays a F#2 gui-
tar note, click on F#2.
q Note that it is possible to select a root key outside the key zone, if required.

Removing sample(s) from a Key Map

D To remove a sample, select the zone it belongs to, and then select “Delete Sample” from the Edit or context
menus.
The sample is removed from the zone and from sample memory.
D To remove a sample from a key zone/map, without removing it from memory, you can either select “No Sam-
ple” with the Sample knob for that zone, or simply replace it with another sample in the same way.

Removing all unassigned samples

D To remove all samples that are not assigned to any key zone, select Delete Unused Samples from the Edit
menu.

Rearranging samples in a Key Map

There is no specific function for rearranging or trading places between samples and key zones. Simply select a key
zone and change the current sample assignment with the Sample knob.

Setting Sample Level

For each key zone you can set a volume level, using the Level button below the display. If the transition between two
key zones causes a noticeable level difference, this parameter can be used to balance the levels.

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 Tuning samples
Sometimes you might find that the samples you wish to use in a key map are slightly out of tune with each other. This
parameter allows you to tune each sample in a map by +/– half a semitone.
D Select the key zone(s) that contains the out of tune sample(s), and use the Tune knob below the keyboard dis-
play.
q If all samples originate from different sources, and all or most of them are pitched slightly different (a not un-
common sampling scenario), you could first tune them so that they all match each other, and then, if neces-
sary, use the Sample Pitch controls in the Osc section to tune them globally to the “song” you wish to use the
samples in.
D Note that if all the samples were slightly out of tune by the same amount in relation to the song you intend to
use the samples in, it would be much simpler to use the Sample Pitch controls in the Osc section directly.

Looping Samples

A sample, unlike the cycles of an oscillator for example, is a finite quantity. There is a sample start and end. To get
samples to play for as long as you press down the keys on your keyboard, they need to be looped.
For this to work properly, you have to first set up two loop points which determine the part of the sample that will be
looped, and make this a part of the audio file. You cannot set loop points in the NN-19, this has to be done in an ex-
ternal sample editor.
All included samples already have set loop points (if needed).
For each sample (or key zone), you can select the following Loop modes by using the Loop knob below the keyboard
display:
D OFF
No looping is applied to the sample.
D FWD
The part between the loop points plays from start to end, then the cycle is repeated. This is the most common loop
mode.
D FWD - BW
The part between the loop points plays from start to end, then from end to start, and then repeats the cycle.
! For samples without any loop points, the whole sample will be looped.

About the Solo Sample function

The Solo Sample button will allow you to listen to a selected sample over the entire keyboard range.
D Select the key zone the sample is assigned to, and then activate Solo Sample.
This can be useful for checking if the root key is set correctly or if the current range is possible to extend etc.
! For Solo Sample to work, “Select Key Zone via MIDI” must be disabled!

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 Automap Samples
If you have a number of samples that belong together, but haven’t mapped them to key zones you can use the “Au-
tomap Samples” function on the Edit menu. This is used in the following way:
1. Select all samples that belong together and load them in one go, using the sample browser.
One of the samples will be assigned to a key zone spanning the whole range, and the rest will be loaded in to
memory but remain unassigned.
2. Select Automap Samples from the Edit menu.
Now all samples currently in memory (assigned or unassigned) will be arranged automatically so that:
D Each sample will be placed correctly according to its root note, and will be tuned according to the information
in the sample file.
Most audio editing programs can save root key information as part of the file.
D Each sample will occupy half the note range to the next sample’s root note.
The root key will always be in the middle of each zone, with the zone extending both down and up in relation to the
root position.

Mapping samples without Root Key or Tuning information

Some samples may not have any information about root key or tuning stored in the file. If the file names indicate the
root key you can manually set it for each sample using the method described below. In a worst case scenario, i.e. no
tuning or root key information whatsoever, you can still make use of the Automap function:
1. Select all samples that belong together and load them in one go, using the sample browser.
One of the samples will be assigned to a key zone spanning the whole range, and the rest will be loaded in to
memory but remain unassigned.
2. Manually set the root key, and adjust the tune knob if the sample needs fine-tuning.
Without any information stored in the file, or if the file name doesn’t indicate the root key, you will have to use your
ears for this step. Play the sample and use another instrument or a tuner to determine its pitch.
3. Select the next sample using the Sample knob, and repeat the previous step.
Proceed like this until you have set a root key for all the samples in memory.
4. Select “Automap Samples” from the edit menu.
The samples will be mapped according to their set root key positions!

How Mapping Information is saved

All information about key zones, high and low range, root key etc. is stored as part of the Sampler Patch. The original
sample files are never altered!

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 NN-19 synth parameters
The NN-19 synth parameters are used to shape and modulate samples. These are mostly similar to the parameters
used to shape the oscillators in Subtractor - you have envelope generators, a filter, velocity control etc. Again, it is im-
portant to remember that these parameters do not alter the audio files in any way, only the way they will play back.
! These parameters are global, in the sense that they will affect all samples in a sample patch.

The Oscillator Section

For a sample patch, the actual samples are what oscillators are for a synthesizer, the main sound source. The follow-
ing settings can be made in the Osc section of the NN-19:

Sample Start
This changes the start position of samples in a sample patch. Turning the knob clockwise gradually offsets the sam-
ples’ start position, so that they will play back from a position further “into” the samples’ waveform. This is useful
mainly for two things:
D Removing “air” or other unwanted artefacts from the start of less than perfect samples.
Occasionally (although not in any samples supplied with Reason) you may come across samples where the start
point of the sample is slightly ahead of the start of the actual sound. There may be noise or silence in the begin-
ning which was not intended to be part of the sample. By adjusting the sample start position, this can be removed.
D Changing the start point as an effect.
For example, if you had a sample of someone saying “one, two, three”, you could change the start position so that
when you played the sample it would start on “three”.
q You can also assign velocity sample start allowing to use your playing to determine the exact sample start.
See later in this chapter.

Setting Sample Pitch - Octave/Semitone/Fine

By adjusting the corresponding knobs you can change the pitch of all samples belonging to a patch, in three ways:
D Octave steps
The range is 0 - 8. The default setting is 4.
D Semitone steps
Allows you to raise the frequency in 12 semitone steps (1 octave).

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 D Fine steps (100th of a semitone)
The range is -50 to 50 (down or up half a semitone).
! Note that the controls in this section cannot be used to tune samples against each other, as all samples will be
affected equally. To tune individual samples, you use the Tune parameter below the keyboard display (see
“Tuning samples”).

Keyboard Tracking
The Osc section has a button named “Kbd. Track”. If this is switched off, the sample’s pitch will remain constant, re-
gardless of any incoming note pitch messages, although the oscillator still reacts to note on/off messages. This could
be useful if you are using non-pitched samples, like drums for example. You could then play a sample in a zone using
several keys, allowing for faster note triggering if you wanted to play a drum roll, for example.

Osc Envelope Amount

This parameter determines to what degree the overall pitch of the samples will be affected by the Filter Envelope
(see “Filter Envelope”). You can set negative or positive values here, which determines whether an envelope param-
eter should raise or lower the pitch.

The Filter Section

Filters are used for shaping the overall timbre of the sound. The filter in NN-19 is a multimode filter with five filter
types.

Filter Mode
With this selector you can set the filter to operate as one of five different types of filter. These are as follows:
• 24 dB Lowpass (LP 24)
Lowpass filters lets low frequencies pass and cuts out the high frequencies. This filter type has a fairly steep roll-
off curve (24dB/Octave). Many classic synthesizers (Minimoog/Prophet 5 etc.) used this filter type.
• 12 dB Lowpass (LP 12)
This type of lowpass filter is also widely used in classic analog synthesizers (Oberheim, TB-303 etc.). It has a gen-
tler slope (12 dB/Octave), leaving more of the harmonics in the filtered sound compared to the LP 24 filter.
• Bandpass (BP 12)
A bandpass filter cuts both high and low frequencies, while midrange frequencies are not affected. Each slope in
this filter type has a 12 dB/Octave roll-off.
• High-Pass (HP12)
A highpass filter is the opposite of a lowpass filter, cutting out the lower frequencies and letting the high frequen-
cies pass. The HP filter slope has a 12 dB/Octave roll-off.

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 • Notch
A notch filter (or band reject filter) could be described as the opposite of a bandpass filter. It cuts off frequencies
in a narrow midrange band, letting the frequencies below and above through.

Filter Frequency
The Filter Frequency parameter (often referred to as “cutoff”) determines which area of the frequency spectrum the
filter will operate in. For a lowpass filter, the frequency parameter could be described as governing the “opening” and
“closing” of the filter. If the Filter Freq is set to zero, none or only the very lowest frequencies are heard, if set to max-
imum, all frequencies in the waveform are heard. Gradually changing the Filter Frequency produces the classic syn-
thesizer filter “sweep” sound.
! Note that the Filter Frequency parameter is usually controlled by the Filter Envelope (see “Envelope Section”
below) as well. Changing the Filter Frequency with the Freq slider may therefore not produce the expected re-
sult.

Resonance
The filter resonance parameter (sometimes called Q) is used to set the Filter characteristic, or quality. For lowpass fil-
ters, raising the filter Res value will emphasize the frequencies around the set filter frequency. This produces a gen-
erally thinner sound, but with a sharper, more pronounced filter frequency “sweep”. The higher the resonance value,
the more resonant the sound becomes until it produces a whistling or ringing sound. If you set a high value for the
Res parameter and then vary the filter frequency, this will produce a very distinct sweep, with the ringing sound being
very evident at certain frequencies.
• For the highpass filter, the Res parameter operates just like for the lowpass filters.
• When you use the Bandpass or Notch filter, the Resonance setting adjusts the width of the band. When you
raise the Resonance, the band where frequencies are let through (Bandpass), or cut (Notch) will become nar-
rower. Generally, the Notch filter produces more musical results using low resonance settings.

Envelope Section
Envelope generators are used to control several important sound parameters in analog synthesizers, such as pitch,
volume, filter frequency etc. Envelopes govern how these parameters should respond over time - from the moment a
note is struck to the moment it is released.
Standard synthesizer envelope generators have four parameters; Attack, Decay, Sustain and Release (ADSR).
There are two envelope generators in the NN-19, one for volume, and one for the filter frequency.
! Please refer to the Subtractor chapter for a description of the basic envelope parameters.

Amplitude Envelope

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 The Amp Envelope is used to adjust how the volume of the sound should change from the time you press a key until
the key is released. By setting up a volume envelope you sculpt the sound’s basic shape with the four Amplitude En-
velope parameters, Attack, Decay, Sustain and Release. This determines the basic character of the sound (soft, long,
short etc.). The Level parameter acts as a general volume control for the sample patch.

Filter Envelope

The Filter Envelope can be used to control two parameters; filter frequency and sample pitch. By setting up a filter
envelope you control the how the filter frequency and/or the sample pitch should change over time with the four Fil-
ter Envelope parameters, Attack, Decay, Sustain and Release.

Filter Envelope Amount

This parameter determines to what degree the filter will be affected by the Filter Envelope. Raising this knob’s value
creates more drastic results. The Envelope Amount parameter and the set filter frequency are related. If the Filter
Freq slider is set to around the middle, this means that the moment you press a key the filter is already halfway open.
The set Filter Envelope will then open the filter further from this point. The Filter Envelope Amount setting affects
how much further the filter will open.

Filter Envelope Invert

If this button is activated, the envelope will be inverted. For example, normally the Decay parameter lowers the filter
frequency, but after activating Invert it will instead raise it, by the same amount. Note that Invert does not affect the
Osc pitch parameter (this can be inverted by setting positive or negative values).

LFO Section

LFO stands for Low Frequency Oscillator. LFOs are oscillators in the sense that they generate a waveform and a fre-
quency. However, there are two significant differences compared to normal sound generating oscillators:
• LFOs only generate waveforms with low frequencies.

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 • The output of the two LFOs are never actually heard. Instead they are used for modulating various parameters.
The most typical application of an LFO is to modulate the pitch of a (sound generating) oscillator or sample, to pro-
duce vibrato.
The LFO section has the following parameters:

Waveform
LFO 1 allows you to select different waveforms for modulating parameters. These are (from the top down):

| Waveform | Description
Triangle This is a smooth waveform, suitable for normal vibrato.
Inverted This produces a “ramp up” cycle. If applied to an oscillator’s frequency, the pitch would sweep up to a set
Sawtooth point (governed by the Amount setting), after which the cycle immediately starts over.
Sawtooth This produces a “ramp down” cycle, the same as above but inverted.
Square This produces cycles that abruptly changes between two values, usable for trills etc.
Random Produces random stepped modulation to the destination. Some vintage analog synths called this fea-
ture “sample & hold”.
Soft Random The same as above, but with smooth modulation.

Destination
The available LFO Destinations are as follows:

| Destination | Description
Osc Selecting this makes LFO control the pitch (frequency) of the sample patch.
Filter Selecting this makes the LFO control the filter frequency.
Pan Selecting this makes the LFO modulate the pan position of samples, i.e. it will move the sound from left
to right in the stereo field.

Sync
By clicking this button you activate/deactivate LFO sync. The frequency of the LFO will then be synchronized to the
song tempo, in one of 16 possible time divisions. When sync is activated, the Rate knob (see below) is used for set-
ting the desired time division.
Turn the knob and check the tooltip for an indication of the time division.

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 Rate
The Rate knob controls the LFO’s frequency. Turn clockwise for a faster modulation rate.

Amount
This parameter determines to what degree the selected parameter destination will be affected by the LFO. Raising
this knob’s value creates more drastic results.

Play Parameters
This section deals with two things: Parameters that are affected by how you play, and modulation that can be applied
manually with standard MIDI keyboard controls.
These are:
• Velocity Control
• Pitch Bend and Modulation Wheel
• Legato
• Portamento
• Polyphony
• Voice Spread
• External Controllers

Velocity Control

Velocity is used to control various parameters according to how hard or soft you play notes on your keyboard. A com-
mon application of velocity is to make sounds brighter and louder if you strike the key harder. By using the knobs in
this section, you can control how much the various parameters will be affected by velocity. The velocity sensitivity
amount can be set to either positive or negative values, with the center position representing no velocity control.
The following parameters can be velocity controlled:

| Destination | Description
Amp This let’s you velocity control the overall volume of the sound. If a positive value is set, the volume will in-
crease the harder you strike a key. A negative value inverts this relationship, so that the volume de-
creases if you play harder, and increases if you play softer. If set to zero, the sound will play at a constant
volume, regardless of how hard or soft you play.
F. Env This sets velocity control for the Filter Envelope Amount parameter. A positive value will increase the
envelope amount the harder you play. Negative values invert this relationship.
F. Dec This sets velocity control for the Filter Envelope Decay parameter. A positive value will increase the De-
cay time the harder you play. Negative values invert this relationship.
S.Start This sets velocity control for the Sample Start parameter. A positive value will increase the Start Time
amount the harder you play. Negative values invert this relationship.
A. Attack This sets velocity control for the Amp Envelope Attack parameter. A positive value will increase the At-
tack time the harder you play. Negative values invert this relationship.

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 Pitch Bend and Modulation Wheels
The Pitch Bend wheel is used for “bending” notes, like bending the strings on a guitar. The Modulation wheel can be
used to apply various modulation while you are playing. Virtually all MIDI keyboards have Pitch Bend and Modulation
controls. NN-19 also has two functional wheels that could be used to apply real time modulation and pitch bend
should you not have these controllers on your keyboard, or if you aren’t using a keyboard at all. The wheels mirror the
movements of the MIDI keyboard controllers.

Pitch Bend Range

The Range parameter sets the amount of pitch bend when the wheel is turned fully up or down. The maximum range
is “24” (=up/down 2 Octaves).

Modulation Wheel
The Modulation wheel can be set to simultaneously control a number of parameters. You can set positive or negative
values, just like in the Velocity Control section. The following parameters can be affected by the modulation wheel:

| Destination | Description
F. Freq This sets modulation wheel control of the Filter Frequency parameter. A positive value will increase the
frequency if the wheel is pushed forward. Negative values invert this relationship.
F. Res This sets modulation wheel control of the Filter Resonance parameter. A positive value will increase the
resonance if the wheel is pushed forward. Negative values invert this relationship.
F. Dec This sets modulation wheel control for the Filter Envelope Decay parameter. A positive value will increase
the decay if the wheel is pushed forward. Negative values invert this relationship.
LFO This sets modulation wheel control of the LFO Amount parameter. A positive value will increase the
Amount if the wheel is pushed forward. Negative values invert this relationship.
Amp This sets modulation wheel control for the Amp level parameter. A positive value will increase the level if
the wheel is pushed forward. Negative values invert this relationship.

Legato
Legato works best with monophonic sounds. Set Polyphony (see “Setting Number of Voices - Polyphony”) to 1 and
try the following:
D Hold down a key and then press another key without releasing the previous.
Notice that the pitch changes, but the envelopes do not start over. That is, there will be no new “attack”.
D If polyphony is set to more voices than 1, Legato will only be applied when all the assigned voices are “used
up”.
For example, if you had a polyphony setting of “4” and you held down a 4 note chord, the next note you played
would be Legato. Note, however, that this Legato voice will “steal” one of the voices in the 4 note chord, as all the
assigned voices were already used up!

Retrig
This is the “normal” setting for playing polyphonic patches. That is, when you press a key without releasing the previ-
ous, the envelopes are retriggered, like when you release all keys and then press a new one. In monophonic mode,
Retrig has an additional function; if you press a key, hold it, press a new key and then release that, the first note is
also retriggered.

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 Portamento (Time)
Portamento is when the pitch “glides” between the notes you play, instead of instantly changing the pitch. The Porta-
mento knob is used to set how long it takes for the pitch to glide from one pitch to the next. If you don’t want any Por-
tamento at all, set this knob to zero.

Setting Number of Voices - Polyphony

This determines the polyphony, i.e. the number of voices a patch can play simultaneously. This can be used to make
a patch monophonic (=a setting of “1”), or to extend the number of voices available for a patch. The maximum num-
ber of voices you can set a patch to use is 99.
! Note that the Polyphony setting does not “hog” voices. For example, if you have a patch that has a polyphony
setting of ten voices, but the part the patch plays only uses four voices, this won’t mean that you are “wasting”
six voices. In other words, the polyphony setting is not something you need to consider if you want to conserve
CPU power - it is only the number of voices actually used that counts.

Voice Spread
This parameter can be used to control the stereo (pan) position of voices. The Spread knob determines the intensity
of the panning. If this is set to “0”, no panning will take place. The following pan modes can be selected:

| Mode | Description
Key This will shift the pan position gradually from left to right the higher up on the keyboard you play.
Key 2 This will shift the pan position from left to right in 8 steps (1/2 octave) for each consecutive higher note you play,
and then repeat the cycle.
Jump This will alternate the pan position from left to right for each note played.

Low Bandwidth
This will remove some high frequency content from the sound, but often this is not noticeable (this is especially true
if you have “filtered down” samples). Activating this mode will save you some extra computer power, if needed.

Controller Section
NN-19 can receive common MIDI controller messages, and route these to various parameters. The following MIDI
messages can be received:
• Aftertouch (Channel Pressure)
• Expression Pedal
• Breath Control
If your MIDI keyboard is capable of sending Aftertouch messages, or if you have access to an Expression Pedal or a
Breath controller, you can use these to modulate NN-19 parameters. The “Source” selector switch determines which
of these message-types should be received.
These messages can then be assigned to control the following parameters:
F. Freq This sets external modulation control of the filter frequency parameter. A positive value will increase the fre-
quency with higher external modulation values. Negative values invert this relationship.
LFO 1 This sets external modulation control of the LFO Amount parameter. A positive value will increase the LFO
amount with higher external modulation values. Negative values invert this relationship.
Amp This let’s you control the overall volume of the sound with external modulation. If a positive value is set, the vol-
ume will increase with higher external modulation values. A negative value inverts this relationship.

416 NN-19 SAMPLER

 Connections
On the back panel of the NN-19 you will find the connectors, which are mostly CV/Gate related.

Audio Outputs
These are the main left and right audio outputs. When you create a new NN-19 device, these are auto-routed to the
first available outputs in the I/O device.

Mono Sequencer Control

These are the main CV/Gate inputs. CV controls the note pitch. Gate inputs trigger note on/off values plus a level,
which can be likened to a velocity value. If you want to control the NN-19 from a Matrix Pattern Sequencer for exam-
ple, you would normally use these inputs. The inputs are “mono”, i.e. they control one voice in the sampler.

Modulation Inputs
! Remember that CV connections will not be stored in the sample patch, even if the connections are to/from the
same NN-19 device!
These control voltage (CV) inputs (with associated voltage trim pots), can modulate various NN-19 parameters from
other devices, or from the modulation outputs of the same NN-19 device. These inputs can control the following pa-
rameters:
• Osc (sample) Pitch
• Filter Cutoff
• Filter Resonance
• Amp Level
• Mod Wheel

Modulation Outputs
The Modulation outputs can be used to voltage control other devices, or other parameters in the same NN-19 device.
The Modulation Outputs are:
• Filter Envelope
• LFO

417 NN-19 SAMPLER

 Gate Inputs
These inputs can receive a CV signal to trigger the envelopes. Note that connecting to these inputs will override the
“normal” triggering of the envelopes. For example, if you connected a LFO output to the Gate Amp input, you would
not trigger the amp envelope by playing notes, as this is now controlled by the LFO. In addition you would only hear
the LFO triggering the envelope for the notes that you hold down.
• Amp Envelope
• Filter Envelope

418 NN-19 SAMPLER

Chapter 23
Pangea
World Instruments
 Introduction

Pangea World Instruments features a unique assortment of rare instruments from all over the world - perfect for
spicing up any music style. Each of the multi-sampled instruments can also be tailored and processed in the high-
quality filter, amp, delay and reverb sections.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel overview
The Pangea front panel contains the following sections:

1 2

3 4 5 6 7 8

The Pangea front panel sections.

• 1. Patch Selector (for browsing, loading and saving patches).
• 2. Master Volume.
• 3. Global performance and “play” controls.
• 4. Instruments section.
• 5. Filter section.
• 6. Amp Envelope section.
• 7. Delay effect section.
• 8. Reverb effect section.

420 PANGEA WORLD INSTRUMENTS

 Using Pangea
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Global performance and “play” controls

Note
The Note LED lights up each time Pangea receives a MIDI Note On.

Range
D Set the desired Pitch Bend range for the “Pitch” wheel with the up/down buttons, or by click-holding on the
display and dragging up/down.
Range: +/-12 semitones (+/-1 octave) in steps of +/-1 semitone.

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Pangea also responds to Pitch Bend MIDI
data from a connected MIDI master keyboard. You set the desired Pitch bend Range with the “Range” control to the
right of the Mod wheel.

Mod
The Mod wheel can be used for controlling the three predefined parameters to the right of the Mod wheel:
• S. Start
Here you set how/if the Mod wheel should affect the Sample Start position of the currently selected instrument.
The parameter is bi-polar, with zero modulation at the 12 o’clock position. A negative value moves the sample start
back and a positive value moves it forward.
! Note that if the S.Start parameter in the Instruments section (see “S. Start (Sample Start)”) is at 0 ms, the sam-
ple start cannot be moved back any further. Similarly, if the S.Start parameter in the Instruments section is at
150 ms, the sample start cannot be moved forward any further.
• F. Freq
Here you set how/if the Mod wheel should affect the Filter Cutoff parameter. The parameter is bi-polar, with zero
modulation at the 12 o’clock position.

421 PANGEA WORLD INSTRUMENTS

 ! Note that if the Cutoff parameter in the Filter section (see “Cutoff”) is at 20 Hz, the frequency cannot be low-
ered any further. Similarly, if the Cutoff parameter in the Filter section is at 25 kHz, the frequency cannot be
raised any further.
• Level
Here you set how/if the Mod wheel should affect the Master Volume parameter. The parameter is bi-polar, with
zero modulation at the 12 o’clock position.
! The volume can never be modulated louder than the maximum +12 dB limit of the Master Volume control.

Panel controls
The Instruments section

Instrument selector

D Click the Instrument name selector to bring up a menu of the available instruments - and then select the de-
sired instrument from the menu.
Alternatively, click and drag up/down in the display above the selector to scroll through the instruments.
! Depending on the instrument size (in MB), it could take a short moment before the entire instrument is loaded
into RAM.
! Also note that the note ranges of the instruments extend outside their “natural” ranges, which could produce
nice artificial effects.
The available instruments are:

422 PANGEA WORLD INSTRUMENTS

 • Acoustic Saz

This is a 5-string electro-acoustic hybrid saz baglema from Turkey, also known as Turkish guitar. It was played with
a hard pick and the strings were recorded with external close mics (stereo) in a studio, dry. The samples are from
Soundiron's Acoustic Saz library.
• Angklung

This is an18-piece tuned bamboo rattle instrument from Indonesia. It was recorded with close mics (stereo) in a
large hall. The samples are from Soundiron's Angklung library.
• Bizarre Sitar

This is a small 8-string sitar from India. It was played with a hard pick and was recorded with close mics (stereo) in
a studio, dry. The samples are from Soundiron's Bizarre Sitar library.

423 PANGEA WORLD INSTRUMENTS

 • Harp Guitar

This is a custom instrument designed by Brad Hoyt. It was played with a hard pick and was recorded with close
mics (stereo) in a studio, dry. The samples are from Soundiron's Brad Hoyt Harp Guitar library.
• Kinderklavier

This is a children’s toy steel tine piano from Germany. It was recorded with close mics (stereo) in a studio, dry. The
samples are from Soundiron's Kinderklavier library.
• Lakeside Pipe Organ

This is a large church pipe organ recorded with close mics (stereo) in a large hall, wet. The samples are from
Soundiron's Lakeside Pipe Organ library.

424 PANGEA WORLD INSTRUMENTS

 • Little Wooden Flutes

This is a Native American walnut 6-hole flute. It was recorded with close mics (stereo) in a studio, dry. The samples
are from Soundiron's Little Wooden Flutes library.
• Little Pump Reeds

This is a pumped reed instrument related to a harmonium, from India. It was recorded with close mics (stereo) in a
studio, dry. The samples are from Soundiron's Little Pump Reeds library.
• Struck Grand Piano

This is a grand piano, with the strings being struck with a small metal hammer. It was recorded with close mics
(stereo) in a large hall. The samples are from Soundiron's Struck Grand Piano library.

425 PANGEA WORLD INSTRUMENTS

 • Traveler Organ

This is a mechanically operated antique organ, also known as a traveling organ. It is operated by pumping in air
using the two pedals and then playing the keyboard. It was recorded with close mics (stereo) in a studio, dry. The
samples are from Soundiron's Traveler Organ library.
• Zitherette

This is an 8 string fretless zither played with a hard pick. It was recorded with a close mic (mono) in a studio, dry.
The samples are from Soundiron's Zitherette library.

S. Start (Sample Start)

D Turn the S.Start knob to set where in the sample the playback should start.
Note that the effect could be different depending on the selected instrument.

Oct

D Set the octave transposition for the instrument.

Range: 5 (+/-2) octaves.
! Note that the note ranges of the instruments can be transposed far outside their “natural” ranges, which could
produce nice artificial effects.

426 PANGEA WORLD INSTRUMENTS

 Semi

D Set the pitch in semitone steps.

Range: +/-12 semitones (two octaves).

Fine

D Set the pitch in steps of 1 cent.

Range: +/- 50 cents (down or up half a semitone).

The Filter section

Filter On/Off

D Click the On/Off LED button to switch on/off the Filter section.

Filter Type selector

D Click and drag up/down on the Filter Type selector to select one of the available filter types - or step through
the filter types by clicking the Up/Down arrow buttons.
The available filter types are:
• LP
This is a lowpass filter with 12db/octave slope.
• HP
This is a highpass filter with 12db/octave slope.
• BP
This is a bandpass filter with 6db/octave slopes.
• Comb
This is a comb filter for phaser/flanger type of effects.

427 PANGEA WORLD INSTRUMENTS

 Cutoff

D Set the cutoff/center frequency with the Cutoff knob.

The cutoff parameter sets where in the frequency range you want the resonance and attenuation to appear.
Range: 20 Hz to 25 kHz.

Reso

D Set the resonance amount with the Reso knob.

The resonance parameter amplifies the frequencies at, and around the cutoff/center frequency.

Env

D With the Env knob you set how much you want the Filter Envelope (see below) to affect the Cutoff frequency.
Range: 0% to 100%.

Filter Envelope

The standard ADSR type envelope controls the filter cutoff frequency modulation over time. The ADSR envelope
characteristics are described in detail in “Amp Envelope”.

Vel

D Turn the Vel knob to set how much the cutoff/center frequency should be modulated by Keyboard Velocity.
Range: 0% to 100%.

428 PANGEA WORLD INSTRUMENTS

 Kbd

D Turn the KBD (Keyboard Track) knob to set how much the cutoff/center frequency should track incoming MIDI
Notes.
Range: 0% (no tracking (constant frequency)) to 100% (1 semitone per key).

The Amp section

Vel

D Turn the Vel knob to set how much the amplitude should be modulated by Keyboard Velocity.
Range: 0% to 100%.

Amp Envelope

429 PANGEA WORLD INSTRUMENTS

 The Amp Envelope is a standard ADSR envelope which controls the amplitude of the signal over time. The picture
below shows the various stages of the ADSR envelope:

Level

Volume
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

• A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value
set with the Master Volume knob. How long this should take, depends on the Attack setting. If the Attack is set to
“0”, the Volume value is reached instantly. If the Attack value is raised, it will take longer time before the Master
Volume value is reached.
• D(ecay)
After the Master Volume value has been reached, the level starts to drop. How long this should take is governed
by the Decay parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to
“0”, the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the
volume gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to
drop to some other value than zero, you raise the Sustain parameter.
• S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain
to full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain pa-
rameter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and
stays there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Volume value, then
gradually decreases to finally land to rest on a level somewhere in-between zero and the Master Volume value.
Note that Sustain represents a level, whereas the other envelope parameters represent times.
• R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume
to drop back to zero after you release the key.

430 PANGEA WORLD INSTRUMENTS

 The Delay section

This is a stereo delay which affects all voices globally.

Delay On/Off

D Click the On/Off LED button to switch on/off the Delay section.

Time

D Set the time between the delay repeats.

If Sync is active (see below), the Time parameter controls the time divisions.

Feedback

D Set the number of delay repeats with the Feedback knob.

Sync

D Click the Sync button to sync the delay time to the main sequencer Tempo.
When active, the Time knob (see above) controls the time divisions.

431 PANGEA WORLD INSTRUMENTS

 Ping Pong

D Activate this to get the delay repeats alternating between the left and right channels.
Note that this also doubles the delay tempo.

Damp

D Raise the Damp value to gradually cut off the high frequencies of the delay repeats.

Amount

D Set the delay amount with the Amount knob.

Note that the Delay effect is routed as a “send effect”, with the Amount knob working as an “effect return” control.
This means that if you play a short note with the Amount knob set to zero, you might still get a delay effect if you
turn up the Amount afterwards (depending on the Feedback setting, see “Feedback”).

The Reverb section

This is a stereo reverb which affects all voices globally.

Reverb On/Off

D Click the On/Off LED button to switch on/off the Reverb section.

432 PANGEA WORLD INSTRUMENTS

 Time

D Set the reverberation duration time.

In practice this sets the “size” of the reverberation “chamber/room/hall”.

Pre-Delay

D Set the pre-delay time of the reverb.

Range: 0-200 ms.

Hi Damp

D Raise the Hi Damp value to cut off the high frequencies of the reverb and thereby create a smoother, warmer
effect.

Lo Damp

D Raise the Lo Damp value to cut off the low frequencies of the reverb signal, to make the reverb effect less
“muddy”.

Amount

D Set the reverb amount with the Amount knob.

Note that the Reverb effect is routed as a “send effect”, with the Amount knob working as an “effect return” con-
trol. This means that if you play a short note with the Amount knob set to zero, you might still get a reverb effect if
you turn up the Amount afterwards (depending on the Time setting, see “Time”).

433 PANGEA WORLD INSTRUMENTS

 Connections

! Remember that CV connections are NOT stored in the Pangea patches! If you want to store CV connections be-
tween devices, put them in a Combinator device and save the Combi patch.

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play Pangea from another CV/Gate device (typically a Matrix
or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/
off along with velocity. There are also inputs for modulating the Pitch Bend and Mod Wheel parameters.

Modulation Inputs
These control voltage (CV) inputs can be used for modulating the Filter Cutoff and Resonance parameters, as well as
the Master Volume level.

Audio Out
These are the main audio outputs. When you create a new Pangea device, these outputs are auto-routed to the first
available outputs in the I/O device.

434 PANGEA WORLD INSTRUMENTS

Chapter 24
Polytone Dual-Layer
Synthesizer
 Introduction

Polytone Dual-Layer Synthesizer is a synth-lover’s synth. Inspired by the great analog poly synths of the past, we
designed Polytone Dual-Layer Synthesizer so that you could quickly create the warm analog sounds you know and
love. With a familiar and immediate interface, you’ll feel right at home.
However, it wouldn’t be a Reason Studios instrument if we didn’t tweak the formula a little. The oscillators include
waveshaping and unique flavors of noise for more tonal variety than the classics. Best of all, Polytone Dual-Layer
Synthesizer has two layers that can be stacked or morphed between for super fat and evolving sounds.
Polytone Dual-Layer Synthesizer is more than just a replica of the past. It’s a loving tribute!

436 POLYTONE DUAL-LAYER SYNTHESIZER

 Panel overview
The Polytone Dual-Layer Synthesizer front panel contains the following sections:

1 2

4 5 6 7

8 9 10 11

12

The Polytone Dual-Layer Synthesizer front panel sections.

• 1. Patch selector (for browsing, loading and saving patches).
• 2. Age and Master Volume controls.
• 3. Layer controls.
• 4. Layer specific Keyboard, Mod Wheel and Pitch Bend controls.
• 5. Layer specific Oscillator and Noise section.
• 6. Layer specific Filter section.
• 7. Layer specific Amplifier section.
• 8. Layer specific Modulation LFO section.
• 9. Layer specific Modulation Envelope section.
• 10. Layer specific Filter Envelope section.
• 11. Layer specific Amplifier Envelope section.
• 12. Global controls, LFO and Effects sections for both layers.

437 POLYTONE DUAL-LAYER SYNTHESIZER

 Signal flow
The picture below show the signal flow in Polytone Dual-Layer Synthesizer:

Shape Pitch Wave KEY

LAYER A (per Voice)
MOD
(Shape)
Type Freq Reso KEY A D S R
OSCILLATOR 1 NOISE
MOD
(Pitch)

FM AMT FILTER AMP+AMP ENV

MOD SYNC 1-MIX-2 Level
(FM)

OSCILLATOR 2 MOD ENV Spread VEL Gain

(Freq) (Freq)
VEL
MOD Delay
(Pitch)
Wave A
Shape Pitch Wave KEY MOD LFO MOD ENV
Rate R
FILTER ENV
MOD (Shape)

VEL
Wave Rate
A D S R

GLOBAL
AGE
LFO

LAYER B (per Voice)

Identical configuration as Layer A Level Target

Single/
Mix/
Morph

Amount Amount

CHORUS REVERB
Master
Volume
: audio signal
Type Decay Type : control signal

Polytone Dual-Layer Synthesizer signal flowchart.

438 POLYTONE DUAL-LAYER SYNTHESIZER

 Playing and using Polytone Dual-Layer Synthesizer
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Global output controls

Age
This controls the “age” of the synthesizer and simulates the “degradation” of the analog components. This degrada-
tion could result in pitch and filter fluctuations, for example, and is a little different for each individual voice. Pitch fluc-
tuation was never a desired behavior of course, but has later become kind of a “nostalgic” effect. Use it if you like!
Range: 2024 to 1970

Volume
This is the main stereo output volume control.

Global performance and “play” controls

Pitch
The Pitch bend wheel can be used for bending note pitches up and down. Polytone Dual-Layer Synthesizer also re-
sponds to Pitch Bend MIDI data from a connected MIDI master keyboard. You can then set the desired Pitch Bend
Range with the “P.Bend Range” control individually for each layer.

Mod
The Mod wheel can be used for controlling FM Amount, Filter Frequency and LFO Depth individually for each layer in
Polytone Dual-Layer Synthesizer.

439 POLYTONE DUAL-LAYER SYNTHESIZER

 Voices
Here you set the desired maximum polyphony of your patch, from 1 to 20 voices. This control is mainly intended for
deliberately restricting the polyphony of a sound. If you just want to play a polyphonic patch you can leave the Voices
setting at 20 at all times. The DSP Load won’t increase with higher voice number settings - only if you play a lot of
notes simultaneously.
q If you want monophonic playback you could use the “Retrig” and “Legato” modes instead of lowering the
Voices parameter to 1.

Key Mode
Here you choose how Polytone Dual-Layer Synthesizer should respond to MIDI Note data:
• Poly
Select this if you want to play Polytone Dual-Layer Synthesizer polyphonically. The maximum number of voices is
20. The number of voices is set in the Voices control at the center right of the Polytone Dual-Layer Synthesizer
panel, see “Voices”.
• Retrig
Select this if you want to play Polytone Dual-Layer Synthesizer in monophonic mode and always retrigger the en-
velopes as soon as you play a new note.
• Legato
The Mono Legato mode is also monophonic. However, if you play a new note without having released the previous
one, the envelopes won’t start over.

Layer controls

Like the name suggests Polytone Dual-Layer Synthesizer consists of two individual layers - Layer A and Layer B.
Each layer has the same configuration and features the same parameters.
You can edit Layer A and Layer B separately and then define how you want the layers to play back: individually, mixed
together, or by morphing between the layers.
Mixing the layers opens up for really nice and fat sounds, whereas morphing between the layers can generate really
complex evolving sounds. The mixing/morphing can be controlled from the Mod Wheel, Keyboard Velocity, Global
LFO, or from a fader on the front panel. The mix/morph balance can also be controlled from the “Mix/Morph” CV in-
put on the rear panel.

440 POLYTONE DUAL-LAYER SYNTHESIZER

 A/B

D Click the A or B button to select which layer to edit (or play, depending on the “Mode” setting).
When Layer A is selected, the front panel has a blue background and when Layer B is selected the background is
red:

Layer A selected (to the left) and Layer B (to the right)

Mode

D Select how the layers should play back.

In Single mode each layer plays back individually, depending on if you have clicked the A or B button (see above).
In Mix mode the layers play back mixed together. The mix relation is defined by the “Mix/Morph” setting.
In Morph mode all the settings of the layer-specific parameters in each layer are morphed between, depending on
the “Mix/Morph” setting. In practice, morphing means that the parameters continuously move between the values
they have in the two layers.
In the picture below the Reso, Freq and Type parameters in the Filter section have different values in Layer A and
B. At a Morph value of 50% these parameters have morphed halfway to the values of the other Layer:

A B A B A B

FILTER FILTER FILTER

DRIVE DRIVE DRIVE

LFO OSC2 LFO OSC2 LFO OSC2

MOD MOD MOD
RESO RESO RESO
FREQ FREQ FREQ
TYPE TYPE TYPE
24 24 24
12 12 12 12 12 12
BP KBD BP KBD BP KBD

LP HP LP HP LP HP

24 24 24 24 24 24
ENV VEL ENV VEL ENV VEL

Three parameters with different values in Layer A and Layer B, and the (50%) morphed result in the image to the right.

441 POLYTONE DUAL-LAYER SYNTHESIZER

 Mix/Morph

Here you select how the layers should play back when Mix or Morph mode has been selected (see “Mode”).
D Select Wheel to control the layer mix/morph relation from the Mod Wheel.
A low Mod Wheel value will play back more of Layer A and vice versa.
D Select Vel to control the layer mix/morph relation from Keyboard Velocity.
Low Keyboard Velocity values will play back more of Layer A and vice versa.
D Select G. LFO to control the layer mix/morph relation from the Global LFO, see “The Global LFO section”.
When mixing/morphing, the Global LFO always uses the full LFO range, independently of the Global LFO Level.
D Select Fader to control the layer mix/morph relation from the fader to the right of the buttons.
The center position of the fader represents a 50/50 Mix/Morph of Layer A and B.

Copy

Here you choose if you want to copy or swap all the layer-specific parameter settings of a layer to/with the other
layer.
D Click the desired button to copy or swap the layer(s).
If you should accidentally hit the wrong button, there is always the [Cmd]/[Ctrl]+[Z] (Undo) function to the rescue.

Panel reference
! Note that all sections below that have pictures with blue panel color describe parameters and functions that
are individual for each of the two Layers.

Glide

Glide (portamento) makes note pitches glide from previous notes to new ones, at the time set with the Time knob.
Glide can be used in all Key Modes (see “Key Mode”).
• When On in Poly Key Mode, the pitches will glide from any of the available voices.
The results will be unpredictable since there is no way of controlling from which note(s) the glide(s) will com-
mence. The effect is very nice, though.
• When On in Retrig or Legato Key Mode, the pitch will glide between consecutive notes.
• If Auto is selected, the pitch will glide between consecutive monophonic notes only when you play legato. If
you have selected Poly Key Mode, Auto will have no effect at all.
If you release the previous key before hitting the new key, there will be no glide effect.

442 POLYTONE DUAL-LAYER SYNTHESIZER

 Wheels

• With the FM AMT knob you control how much the Mod Wheel should affect the “FM Amt” parameter in the Os-
cillator section.
• With the FILTER FREQ knob you control how much the Mod Wheel should affect the “Freq” parameter of the
Filter.
• With the LFO DEPTH knob you control how much the Mod Wheel should scale the MOD parameters - if they
are set to a value in the “LFO range”.
The MOD parameters can be found in the Oscillator and Filter sections:

! To have any effect, the MOD knobs have to be set somewhere in the “LFO range”, i.e. turned counter-clockwise
from the 12 o’clock position.
If you set the LFO DEPTH knob to maximum and have some other MOD knobs set to an “LFO range” value, the
Mod Wheel will gradually introduce the LFO modulation the further you move it. With the Mod Wheel at zero, there
will be no LFO modulation at all.
See “The Mod LFO section” for more information about the LFO.

P.Bend Range

D Set the desired range for the Pitch Bend wheel (see “Pitch”) with the up/down buttons, or by dragging up/
down in the display.
Range: +/-24 semitones (+/-2 octaves) in steps of +/-1 semitone.

443 POLYTONE DUAL-LAYER SYNTHESIZER

 The Oscillator section

The Oscillator section consists of two separate Oscillators, plus a Noise oscillator and a Mixer. The two Oscillators
can also be cross-modulated (Oscillator 2 frequency modulates Oscillator 1) or synced (Oscillator 1 is synced to Os-
cillator 2).
The parameter configuration is identical for the two Oscillators.

KBD
D Set how much the pitch should track incoming MIDI Notes.
Range: 0% (no tracking (constant pitch)) to 100% (1 semitone per note).

Fine
D Change the pitch in steps of 1 cent.
Range: +/- 50 cents (down or up half a semitone).

Pitch
D Set the pitch of the respective Oscillator with the Pitch knob.
Range: +/-24 semitones (+/-2 octaves) in steps of 1 semitone.

Pitch Mod
D Set a desired pitch modulation amount with the Mod knob.
Turning the knob counter-clockwise from the 12 o’clock position will make the MOD LFO modulate the pitch.
Turning the knob clockwise from the 12 o’clock position will modulate the pitch from the MOD ENV.
Range: 100-0% LFO modulation and 0-100% MOD ENV modulation.

Waveform selector
D Click the spin control buttons, or click the display to select the desired oscillator waveform.
You can then change the waveform shape with the “Shape” knob.
The following waveforms are available:
• Saw-Pulse
A sawtooth wave at Shape=0 gradually transformed to a 50% duty cycle square wave at Shape=100%.
• Pulse
A ~5% duty cycle pulse wave at Shape=0 gradually transformed via a 50% duty cycle square wave towards a
~95% duty cycle pule wave at Shape=100%.
• Triangle
A pure triangle (odd partials only) wave at Shape=0 with gradual introduction of even partials up to 50%, then
gradual reduction of the odd partials, resulting in only even partials (one octave up) at Shape=100%.

444 POLYTONE DUAL-LAYER SYNTHESIZER

 • Sine
A pure sine wave at Shape=0. Increasing the Shape value will sweep through the harmonic overtones, resulting in
a pure sine wave 3 octaves above the root note at Shape=100%.
• Digital
A selection of very nice and useful wave tables which you could crossfade between with the Shape knob.
• Band Noise (Oscillator 1 only)
Noise, where Shape controls the bandwidth. With Shape at 0% the result is close to a sine wave, while higher
Shape settings adds progressively more noise.
• Chaos (Oscillator 2 only)
This produces a wide range of noises and heavily modulated signals, great for adding a bit of chaos to the sound!

Shape
D Turn the Shape knob to transform the Oscillator waveforms, as described above.

Shape Mod
D Set a desired Shape modulation amount with the Shape Mod knob. Turning the knob counter-clockwise from
the 12 o’clock position will make the MOD LFO modulate the Shape. Turning the knob clockwise from the 12
o’clock position will modulate the Shape from the MOD ENV.
Range: 100-0% LFO modulation and 0-100% MOD ENV modulation.

FM Amt
D Set the amount of linear frequency modulation of Oscillator 1 by Oscillator 2 with the FM Amt knob.

FM Amt Mod
D Set a desired FM Amt modulation amount with the FM Amt Mod knob.
Turning the knob counter-clockwise from the 12 o’clock position will make the MOD LFO modulate the FM Amt.
Turning the knob clockwise from the 12 o’clock position will modulate the FM Amt from the MOD ENV.
Range: 100-0% LFO modulation and 0-100% MOD ENV modulation.

Osc Sync
D Click the Osc Sync button to have Oscillator 2 sync the restart of Oscillator 1.
Oscillator sync is when one oscillator will restart the period of another oscillator, so that they will have the same
base frequency. In Polytone Oscillator 2 is the syncing oscillator, and thus controls the pitch. If you change or mod-
ulate the pitch of the synced oscillator (Oscillator 1 in Polytone) you get that characteristic sound associated with
oscillator sync. To hear the sync effect, make sure you listen to Oscillator 1.

Syncing oscillator
(Oscillator 2)

Synced oscillator
(Oscillator 1)

q Modulate the pitch of Oscillator 1 to get the characteristic “sync sound”, see “Pitch Mod”.

445 POLYTONE DUAL-LAYER SYNTHESIZER

 The Mixer

The Oscillator Mixer is where you mix and set the signal level from the Oscillator section. You can also add Noise to
the signal.

1-Mix-2
D Set the mix between the Oscillator 1 and 2 signals.

Level
D Set the level of the mixed oscillator signals.
High level settings will overdrive the Filter input, as indicated by the Drive LED in the Filter section (see “Drive”).

Noise
D Set the level of white noise you want to add to the mixed oscillator signals.
White noise contains all frequencies at equal energy levels.

The Filter section

The Filter section features a very advanced (continuous) state variable filter. The filter types are:
• 24 dB/Oct Lowpass
Amplitude

RESO

Frequency
FREQ

446 POLYTONE DUAL-LAYER SYNTHESIZER

 • 12 dB/Oct Lowpass
Amplitude

RESO

Frequency
FREQ
• 12 dB/Oct Bandpass
Amplitude

RESO

Frequency
FREQ
• 12 dB/Oct Highpass
Amplitude

RESO

Frequency
FREQ
• 24 dB/Oct Highpass
Amplitude

RESO

Frequency
FREQ
The really clever thing with the Polytone filter is that you can continuously sweep through all filter types, i.e. the filter
types “morph” into each other seamlessly from 24 dB LP to 24 dB HP.

Drive
• The Drive LED lights up when the signal in the Filter is overdriven (e.g. by a high “Level” and/or by a high
“Reso” setting).

447 POLYTONE DUAL-LAYER SYNTHESIZER

 Reso
D Set the resonance amount with the Reso knob.
A high Resonance setting will amplify the signal level around the cut-off Frequency, as described in this picture:
Amplitude

Amplitude Amplitude

Time Time
RESONANCE

Frequency
FREQ
(Cutoff Frequency)

Amplitude

Amplitude Amplitude

Time Time
RESONANCE

Frequency
FREQ
(Cutoff Frequency)

Amplitude

Amplitude Amplitude

Time Time
RESONANCE

Frequency
FREQ
(Cutoff Frequency)

! Be careful when using high Resonance values as this could generate quite loud audio levels!

Freq
D Set the cutoff frequency (or center frequency for the Bandpass filter type) with the Freq knob.

Filter type
D Select the desired filter type with the knob.
Range: continuous between 24 dB LP and 24 dB HP

Env
D Set how much you want the Filter Envelope to affect the filter cutoff Frequency.
Range: -100 (inverted modulation) to 100 (positive modulation)

Vel
D Set how much you want the Filter Envelope Amount to be affected by Keyboard Velocity.
Range: 0-100%

448 POLYTONE DUAL-LAYER SYNTHESIZER

 KBD
D Set how much the cutoff Frequency should track incoming MIDI Notes.
Range: 0% (no tracking (constant Frequency)) to 100% (1 semitone per note).

Freq Mod
D Set a desired cutoff Frequency modulation amount with the Freq Mod knob.
Turning the knob counter-clockwise from the 12 o’clock position will make the MOD LFO modulate the cutoff Fre-
quency. Turning the knob clockwise from the 12 o’clock position will modulate the cutoff Frequency from the Os-
cillator 2 signal, resulting in the popular “Filter FM” effect found on some great analog synths from the past.
Range: 100-0% LFO modulation and 0-100% Oscillator 2 modulation.

Amplitude
FREQ MOD (by LFO or OSC2)

Frequency (linear)
FREQ

The Filter Envelope section

The Filter Envelope can be used for modulating the Filter cutoff frequency over time. You set the modulation amount
with the “Env” knob in the Filter section. It features the standard Attack, Decay, Sustain and Release parameters of an
ADSR envelope.
The various envelope stages work exactly like those of the Amp Envelope, see “The Amp Envelope section”.

Level

Gain
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
D Set the time it should take to reach from the set “Freq” value to the maximum Filter Frequency value.

449 POLYTONE DUAL-LAYER SYNTHESIZER

 D(ecay)
D Set the time it should take to go from maximum Filter Frequency value to the Sustain value (see below).

S(ustain)
D Set the Filter Frequency value the envelope should rest at, after the Decay stage (see above).

R(elease)
D Set the time it should take to go from the Sustain level back to the set “Freq” value, after you have released
the key.

The Amp section

The Amp section is where you control the output gain of the signal, as well as any Keyboard Velocity and stereo
Spread.

Gain
D Set the desired maximum level for the Amplifier with the Gain knob.
This is the maximum level the Amp Envelope will reach after the Attack stage is completed (see Amp Envelope).

Vel
D If you want the Gain level to be controlled from Keyboard Velocity, turn up the Vel knob.

Spread
D Set the desired stereo spread of the output signal from the Amplifier.
Spread pans each alternate voice left-right (with Spread controlling the amount of panning). This means that if you
play a chord, the individual notes in the chord will be panned differently.
In Mix mode, the Spread panning will be mirrored for Layer A and B. By mixing two detuned or slightly different lay-
ers with Spread, you can create a really wide stereo sound.
! Note that the Spread function doesn’t work in Legato mode, see “Legato”.

The Amp Envelope section

450 POLYTONE DUAL-LAYER SYNTHESIZER

 The Amp Envelope section contains a standard ADSR envelope, which controls the amplitude of the signal.
The picture below shows the various stages of the ADSR envelope:

Level

Gain
(level)
Sustain
(level)
Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope stages.

A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value
set with the “Gain” knob in the Amp section. How long this should take, depends on the Attack setting. If the Attack
is set to “0”, the Gain level is reached instantly. If the Attack value is raised, it will take longer time before the Gain
level is reached.

D(ecay)
After the Gain level has been reached, the level starts to drop. How long this should take is governed by the Decay
parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to “0”,
the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the volume
gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to drop to
some other value than zero, you raise the Sustain parameter.

S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain to
full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain param-
eter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays
there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Gain level, then grad-
ually decreases to finally land to rest on a level somewhere in-between zero and the Gain level. Note that Sustain
represents a level, whereas the other envelope parameters represent times.

R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume to
drop back to zero after you release the key.

451 POLYTONE DUAL-LAYER SYNTHESIZER

 The Mod LFO section

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO
modulate the pitch of a signal to produce vibrato, but there are also other applications for LFOs.
The Mod LFO is used for modulating parameters in the selected Layer that feature MOD knobs with an LFO sector,
i.e. the Oscillator and Filter sections. Here is also where you set the desired modulation amount for the parameters, by
turning the MOD knobs counter-clockwise from the 12 o’clock positions:

q You can also gradually introduce the LFO modulation amount by using the Mod Wheel, see “Wheels”.

Waveform selector
D Select an LFO waveform by clicking the spin controls to the right of the waveform display, or by click-holding
in the display and dragging up or down.
Besides the standard waveforms (sine, triangle, pulse, etc.) there are stepped and smooth random waveforms. The
shape of the waveforms are shown in the display.
! Note that the MOD LFO waveforms are switched between - not continuously morphed between - in Morph
Mode.

Rate
D Set the LFO frequency with the Rate knob.
Range: 0.16-57.0 Hz

Delay
D Turn the Delay knob to introduce a delay before the LFO modulation kicks in after you play a note.
Range: 0-2.0 s

452 POLYTONE DUAL-LAYER SYNTHESIZER

 The Mod Envelope section

The Mod Envelope section consists of a simple AR (Attack, Release) envelope, which can be used for modulating pa-
rameters in the selected Layer that feature MOD knobs with an ENV sector, i.e. the Oscillator section. Here is also
where you set the desired modulation amount for the parameters, by turning the MOD knobs clockwise from the 12
o’clock positions:

The Mod Envelope has two stages - Attack and Release, according to the following principle:
Level

Level

Time
Attack Release
(time) (time)

Key Down Key Up

The ADSR envelope stages.

As you can see in the picture above, Key Down triggers the envelope, but the Key Up doesn’t matter if it happens af-
ter the envelope has finished its two stages.
If Key Up happens before the Attack stage has finished, the Release stage takes over at the current Level.

Attack
D Set the time it should take to reach from zero to the maximum modulation value.

Release
D Set the time it should take to go from maximum modulation value back to zero.
if Release if set to max, it will be infinite, acting as sustain.

Vel
D Turn up the Vel knob if you want the Mod Envelope level to be controlled from Keyboard Velocity.

453 POLYTONE DUAL-LAYER SYNTHESIZER

 The Global LFO section

The Global LFO affects parameters in both Layers globally. It can also be used for modulating the Mix or Morph be-
tween the Layers. Note that the Global LFO is also global for all voices, meaning it modulates all voices equally and
simultaneously.

Waveform selector
D Select an LFO waveform by clicking the spin controls to the right of the waveform display, or by clicking the
display and dragging up or down.
Besides the standard waveforms (sine, triangle, pulse, etc.) there are random, slope and stepped waveforms. The
shape of the waveforms are shown in the display.

Rate
D Set the LFO frequency with the Rate knob.
Range: 0.16-57.0 Hz

Sync
D Click the Sync button to have the Rate synced to time divisions of the main sequencer tempo.

1-Shot
D Click the 1-Shot button to have the LFO only run one single cycle of the waveform.
This is useful if you want the LFO to work as kind of an “extra” envelope generator for controlling a desired target
parameter.
When Polytone is in Retrig Mode (see “Key Mode”), the LFO will retrigger with each new note, regardless of how
you play. In all other Key Modes, the LFO will require that you release all notes before it can retrigger (it will not re-
trigger if you play legato).

Level
D Set the desired modulation amount with the Level knob.

Target selector
D Select the desired target (destination) parameter for the LFO modulation, by clicking the spin controls to the
right of the waveform display, or by clicking the display and selecting from the pop-up menu.
q Note that you can also control the Morph “position” from the Global LFO. You can set this up by clicking the G.
LFO button in the “Mix/Morph” section.

454 POLYTONE DUAL-LAYER SYNTHESIZER

 The Chorus section

The Chorus section features a stereo chorus effect with two selectable speeds, and a fixed-speed stereo flanger ef-
fect.
D Set the effect level with the Amount knob.
D Click one of the buttons for the desired effect.

The Reverb section

The Reverb section features a stereo echo effect and two types of stereo reverbs.
D Set the echo/reverb level with the Amount knob.
D Set the echo/reverb decay time with the Decay knob.
D Click one of the buttons to select the desired effect type.

455 POLYTONE DUAL-LAYER SYNTHESIZER

 Connections

Sequencer Control
The Sequencer Control CV and Gate inputs allow you to play Polytone Dual-Layer Synthesizer from another CV/
Gate device (typically a Matrix or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the
Gate input delivers note on/off along with velocity.

Mix/Morph
This CV input allows for control of the Mix/Morph balance.

CV Modulation inputs
These control voltage (CV) inputs and can be used for modulating the corresponding parameters from external mod-
ulations sources. All inputs also have a corresponding attenuation knob for adjusting the input CV level.

Audio Output
These are the main audio outputs.

Wheels
These are CV inputs for modulating the Pitch Bend and Mod Wheel parameters. The inputs also have a correspond-
ing attenuation knob for adjusting the input CV signal level.

456 POLYTONE DUAL-LAYER SYNTHESIZER

Chapter 25
Radical Piano
 Introduction

Radical Piano is designed to be a straightforward, awesome sounding and very flexible piano. Radical Piano com-
bines sample playback technology with physical modeling to give you great sound quality and seamless dynamic re-
sponse as well as great freedom to tweak your sounds.
The combination of sample playback and physical modeling also makes it possible to keep each piano sound set
down to a minimum size. This allows for very quick loading times when you switch between instruments.
Radical Piano also features sympathetic resonance, which means that any undamped strings will ring along with the
played notes (strings), just like on acoustic pianos. This makes Radical Piano sound extremely realistic and alive.
There are also a number of other controls for further shaping the sound the way you want it.
As a bonus, we also included an audio input so you can route external audio and process it in Radical Piano!
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

The pianos
Radical Piano holds complete sound sets recorded from these three pianos:
• Home Grand
A Bechstein grand piano with a nice “not perfectly tuned” home grand character.
• Deluxe Grand
A Steinway Model D grand piano - one of the greatest grand pianos out there. This particular one belongs to
Sveriges Radio (Swedish Radio Ltd).
• Upright
A Futura upright piano with a distinct “living room” character.

458 RADICAL PIANO

 The microphone configurations

The microphone configurations for the grand pianos and the upright piano respectively.
The pianos were recorded using up to nine microphones per instrument, placed at various critical positions inside and
outside of the pianos. The different microphone recordings were then stored in Radical Piano as separate sound sets.
The following microphone configurations were used:
• Vintage Mono
A single microphone placed outside the waist of the grand piano (or behind the upright piano). For the Steinway
grand piano we used an old school mono ribbon mic with vintage characteristics and a narrow frequency response
with the emphasis in the mid range. For the Futura upright piano we used a vintage tube microphone.
• Ambience
Two microphones in AB configuration* placed quite far away from the piano to capture the room ambience.
• Floor
Two pressure zone microphones that lay flat on the floor just behind the front legs of the grand piano (and behind
the upright piano). They add depth and richness to the sound and are best used as a complement to the other
mics.
• Jazz
Two microphones in AB configuration* placed just outside/in front of the piano. This gives a full bodied sound with
a wide stereo image and a less pronounced attack.
• Close
Two microphones in XY configuration** placed close to the hammers. The close mics produce a distinct sound with
a sharp attack, ideal for uptempo pop/rock.
* AB configuration: Two mics in stereo configuration placed several feet apart and tilted slightly away from each
other.
** XY configuration: Two mics in stereo configuration placed close together in 'V' shape at a 90° coincidence.

459 RADICAL PIANO

 Using Radical Piano
Loading and saving patches
Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Selecting piano sound sets

A patch in Radical Piano can consist of a mix between two piano sound sets. The mix could be between two sound
sets from the same piano, or from different pianos. You could, for example, blend a Close mic’ed upright piano with
the Floor microphones from a grand piano to create your own custom piano sound. The piano sound sets can be se-
lected in the Piano Select section:

The Piano Select section.

1. Select desired piano sound set(s) by clicking the corresponding LED button(s).
You can select one sound set to the left of the Blend knob and one to the right.
2. Set the mix between the sound sets with the Blend Microphones knob.
If you only want to use a single sound set for your sound, set the Microphone Blend knob to min or max.

Character

D Set the character of the sound with the Character knob.

Range: Subdued to Agitated, in 24 steps, with natural sound at the 12 o’clock position.
Subdued produces a warm and mellow tone whereas Agitated generates a brighter and significantly more pro-
nounced tone.
! Changing the Character value temporarily mutes the audio outputs.

460 RADICAL PIANO

 Volume

The master volume control for Radical Piano.

Velocity Response

Most sample-based piano instruments and sound libraries on the market use a predefined number of velocity layers.
Depending on how soft or hard you play the keys, samples from a specific velocity layer play back. Due to memory
limitations, the number of velocity layers aren’t often that many. This can make the velocity response feel and sound
unnatural. Thanks to the combination of samples and physical modeling in Radical Piano, all sound sets feature very
wide and completely seamless velocity ranges.
With the Velocity Response knobs you can tailor the dynamic response of your piano sound.
• With the High knob you set the timbre for the highest velocity.
Note that the High parameter can go far beyond the natural range of an acoustic piano, which is great for experi-
mental sounds.
• With the Low knob you set the timbre for the lowest velocity.
With the Low knob set to zero (marked with an ‘S’) playing really soft won’t play back any sound at all. This can be
useful if you, for example, want to hold down a chord and then play other keys to introduce the sympathetic reso-
nance effect, see “Resonance”.
• With the Curve knob you set the shape of the velocity curve - from exponential, via linear to logarithmic.
Set this parameter where it feels the best to play. There is no “perfect” position since most MIDI keyboards re-
spond differently to velocity.
q If you want a natural dynamic range, set the Low knob to around the 9 o’clock position and the High knob to
around the 12 o’clock position. Adjust the Curve setting to your liking.
q If you want a dynamic range that stretches beyond the range of an acoustic piano, set the Low knob to zero
and the High knob past the 12 o’clock position.
q If you want a static response (with the same timbre no matter how soft or hard you play), set the Low knob to
max and the High knob to zero. Note that there will still be some velocity sensitivity left for controlling the vol-
ume.

461 RADICAL PIANO

 Tune

Cent
D Set the overall master tune of your sound with the Cent knob.
Range: +/-1 semitone (+/-100 cents).

Drift
The Drift parameter can be used for introducing a slow irregular pitch variation to your sound. It’s perfect for adding
kind of a “scary” or melancholic touch to your piano sound.

Sustain

The Sustain parameter is a special feature in Radical Piano. It lets you control the piano sustain continuously from
pedal up to pedal down. As on acoustic pianos, the sustain pedal is not either “on” or “off - it can be “somewhere in
between” as well. The Sustain function in Radical Piano simulates this behavior.
The Sustain parameter can be controlled either from the Pedal LED strip control on the front panel or from a Sustain
pedal connected to the Sustain Pedal input of your MIDI master keyboard.
• When you use a standard (switch type) sustain pedal connected to a standard sustain pedal input on your MIDI
keyboard, this will control the Sustain function in Radical Piano as either Off (‘0’) or On (‘127’).
You could record using the standard sustain pedal and then manually edit the Sustain Pedal performance control-
ler data in the note clip in Reason afterwards and adjust the “in between” Sustain levels.
! The Sustain parameter value (and LED bar) will always adjust to the latest incoming Sustain Pedal data, be it
from the Pedal LED strip control or from a sustain pedal connected to your MIDI keyboard.

462 RADICAL PIANO

 Resonance

Sympathetic resonance is a physical phenomenon that can occur in acoustic instruments, like in pianos for example.
It means that any undamped strings will ring along with the played strings. For example, if you play a key with the sus-
tain pedal down, all other strings in the piano will also vibrate at various intensities. Similarly, if you hold down a num-
ber of keys (so that the dampers are off the strings) and then play additional keys, the strings for the held keys will
resonate.
With the Resonance controls you set the amount of sympathetic resonance in your piano sound.

Level
D Set the amount of overall sympathetic resonance in your sound.

Release Time
D Set the time it should take for the sympathetic resonance to fade to silence.

Envelope

Radical Piano features a special type of envelope generator which is used for shaping the character of the piano
sound.

Attack
D Set the attack time for the piano sound, from immediate to (unnaturally) slow.
The range is 0-200 ms.

Decay Curve
D Set the shape of the decay curve.
This control determines how the sound should decay when you play and hold the keys.
The range is from exponential, via linear, to logarithmic. Exponential settings will make the sound decay faster,
which simulates a piano with little body sustain. Logarithmic settings makes the sound sustain more slowly and
simulates a piano with a lot of body sustain.

463 RADICAL PIANO

 Release
D Set the time it should take for the sound to fade to silence once you release the keys.
This simulates the behavior of the dampers. For example, worn out dampers could result in somewhat longer re-
lease times.

Mechanics

The Mechanics section features controls for the mechanical noise.

Key Down
• Key Down controls the level - and character - of the noise that occurs when the keys are pressed/hit.
At the 12 o’clock position the noise is the most natural. Above the 12 o’clock position the noise is more pro-
nounced and below the 12 o’clock position the noise is suppressed.

Key Up
• Key Up controls the level of the noise that occurs when the keys are released and the hammers and dampers
return to their initial positions.
At the 12 o’clock position the noise level is natural. Above the 12 o’clock position the noise is louder and below the
12 o’clock position the noise is quieter.

Pedal
• Pedal controls the level of the noise that occurs when you press and release the sustain pedal.
At the 12 o’clock position the noise level is natural. Above the 12 o’clock position the noise is louder and below the
12 o’clock position the noise is quieter.

EQ

The built-in equalizer is a powerful 3-band EQ with gain controls for the Low, Mid and High bands. The EQ charac-
teristics have been fine tuned and optimized for piano sounds. The gain range is +/-18dB for each of the bands,
which makes it easy to quickly achieve great sonic results.
The EQ can be switched on/off by clicking the LED button at the top.

464 RADICAL PIANO

 Ambience

The Ambience section features four different reverb types and a Level control. The reverb types are:
• Small Room
This simulates the acoustic reflections in a small room.
• Large Room
This simulates the acoustic reflections in a large room.
• Hall
This simulates the acoustic reflections in a medium size hall.
• Theater
This simulates the acoustic reflections in a large hall/theater.

Output

Comp(ression)
This controls the amount of compression of your piano sound.

Width
This lets you set the stereo width of the piano sound.
! Note that the Width control does not have any effect on the sound if you use only the “Vintage Mono” piano
sound set(s), see “Selecting piano sound sets”.

465 RADICAL PIANO

 Connections
! Remember that CV connections will not be stored in the Radical Piano patch!

Sequencer Control
The Sequencer Control CV and Gate inputs allow you to play Radical Piano from another CV/Gate device (typically a
Matrix or an RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers
note on/off along with velocity.

Modulation In
These control voltage (CV) inputs (with associated trim pots) can modulate following parameters in Radical Piano:
• Pitch
The Pitch can be modulated at a maximum range of +/-1 octave.
! Note that +/- 1 octave is the maximum range a piano sound can be pitch shifted in Radical Piano. This as-
sumes that no Pitch Bend performance controller is used (see “Pitch Bend”) and that the Character knob is set
to Natural (see “Character”).
• Master Volume

Audio In
Route an external audio signal to this input to process it with the Resonance, EQ, Ambience and Compression effects
in Radical Piano.
q Routing a vocal signal and processing it with the sympathetic resonance effect (with the sustain pedal down)
could generate really interesting results. It would be like singing into a piano body!

Audio Out
These are the main audio outputs. When you create a new Radical Piano device, these outputs are auto-routed to the
first available outputs in the I/O device.

466 RADICAL PIANO

 Additional external control
The Radical Piano responds to the following standard Performance Controllers:
• Pitch Bend
Radical Piano responds to Pitch Bend data from the pitch bend control of your MIDI master keyboard.
The range is fixed at +/-7 semitones.
! Note that the Character setting (see “Character”) as well as any Pitch CV modulation (see “Pitch”) can reduce
the Pitch Bend range.
• Sustain Pedal
If you have a standard (switch type) sustain pedal connected to a standard Sustain Pedal input of your MIDI mas-
ter keyboard, this can be used for controlling Sustain On/Off. You can then edit the Sustain values in your note
clips in the sequencer afterwards and set continuous values all the way between 0-127, see “Sustain” for more
details.

467 RADICAL PIANO

468 RADICAL PIANO
Chapter 26
Redrum Drum
Computer
 Introduction

At first glance, Redrum looks styled after pattern-based drum machines, like the legendary Roland 808/909 units. In-
deed, it does have a row of 16 step buttons that are used for step programming patterns, just like the aforementioned
classics. There are significant differences, however. Redrum features ten drum “channels” that can each be loaded
with an audio file, allowing for completely open-ended sound possibilities. Don’t like the snare - just change it. Com-
plete drum kits can be saved as Redrum Patches, allowing you to mix and match drum sounds and make up custom
kits with ease.

470 REDRUM DRUM COMPUTER

 About file formats
Redrum reads two basic types of files:

Redrum Patches
A Redrum patch (Windows extension “.drp”) contains all settings for all ten drum sound channels, including file refer-
ences to the used drum samples (but not the actual drum samples themselves). Switching patches is the same as se-
lecting a new drum kit.

Drum Samples
The audio file format support differs depending on which computer OS you are using.
Redrum can read audio files in the following formats:
• In Windows:
.wav, .aif, .mp3, .aac, .m4a and .wma.
• In macOS:
.wav, .aiff, .3g2, .3gp, .mp1, .mp2, .mp3, .mpeg, .mpa, .snd, .au, .sd2, .ac3, .aac, .adts, .amr, .caf, .m4a .m4r and .mp4.
• SoundFonts (.sf2)
SoundFonts are an open standard for wavetable synthesized audio, developed by E-mu systems and Creative
Technologies.
• REX file slices (.rx2, .rex, .rcy)
REX files are music loops created in the ReCycle program or when editing audio clips inline in Reason. Redrum
lets you load separate slices from REX files as individual samples.
• Any sample rate and practically any bit depth.

471 REDRUM DRUM COMPUTER

 Using patches
When you create a new Redrum device it is loaded with a default kit. If you like you can program a pattern and play
back using the default kit - or you can load another Redrum patch (or create one from scratch, by loading individual
drum samples). A Redrum patch contains settings for the ten drum sound channels, complete with file references to
the drum samples used.
! Redrum patterns are not part of the patch! If you want to save Redrum patches complete with patterns, create
a Combinator containing the Redrum and save the Combi patch.

Loading a patch
To load a patch, use one of the following methods:
D Use the Browser to locate and load the desired patch.
To open the browser and set browse focus to the Redrum device, select “Browse Redrum Patches” from the de-
vice context menu, or click the folder button in the patch section on the device panel.

D Once you have selected a patch, you can step between all the patches in the same folder by using the arrow
buttons next to the patch name display.
D If you click on the patch name display on the device panel, a pop-up menu will appear, listing all patches in the
current folder.
This allows you to quickly select another patch in the same folder, without having to step through each one in turn.
D Drag a Redrum (.drp) patch from the Browser and drop on the device panel.
The panel is dimmed in orange and a Patch Replace symbol appears in the center.

Checking the sounds in a patch

There are two ways you can listen to the sounds in a patch without programming a pattern:
D By clicking the Trigger (arrow) button at the top of each drum sound channel.

D By playing the keys C1 to A1 on your MIDI keyboard.

C1 plays drum sound channel 1 and so on. See also “Using Redrum as a sound module”.
Both these methods play back the drum sample for the corresponding drum sound channel, with all settings for the
sound applied.

472 REDRUM DRUM COMPUTER

 Creating a new patch
To create a patch of your own (or modify an existing patch), you use the following basic steps:
1. Click the folder button for a drum sound channel.
The Redrum sample browser opens.

2. Locate and load a drum sample.

You will find a large number of drum samples in the Factory Sound Bank (in the folder Redrum Drum Kits/xclusive
drums-sorted). You can also load other samples in any supported format.
D Alternatively, drag a sample file from the Browser and drop on the desired sound channel section.
The sound channel is dimmed in blue and a Sample Replace symbol appears in the center.
3. Make the desired settings for the drum sound channel.
The parameters are described in “Redrum parameters”.
4. Repeat steps 1 and 3 for the other drum sound channels.
5. When you’re satisfied with the drum kit, you can save the patch by clicking the Floppy Disk button in the patch
section on the device panel.
Note however, that you don’t necessarily need to save the patch - all settings are included when you save the
song.

Loading REX file slices

Loading slices from within a REX file is done much in the same way as loading “regular” samples:
1. Open the Browser as described above.
2. Browse to a REX file.
Possible extensions are “.rx2”, “.rex” and “.rcy”.
3. Unfold the REX file.
The browser will now display a list of all the separate slices within the REX file.
4. Select the desired slice and click the Load button in the Browser.
The slice is loaded into the Redrum.
D Alternatively, drag a REX slice file from the Browser and drop on the desired sound channel section.
The sound channel is dimmed in blue and a Sample Replace symbol appears in the center.

Creating an empty patch

To “initialize” the settings in the Redrum, select “Reset Device” from the device context menu. This removes all sam-
ples for all drum sound channels, and sets all parameters to their default values.

473 REDRUM DRUM COMPUTER

 Programming patterns
Pattern basics
Redrum contains a built-in pattern sequencer. Unlike the main sequencer in Reason, the Redrum sequencer repeat-
edly plays back a pattern of a specified length. The typical analogy in the “real world” is a drum machine which plays
drum patterns, usually one or two bars in length.
Having the same pattern repeat throughout a whole song may be fine in some cases, but most often you want some
variations. The solution is to create several different patterns and program pattern changes (automatic switching from
one pattern to another) at the desired positions in the song.

How the Redrum pattern sequencer integrates with the main sequencer
The built-in pattern sequencer in the Redrum interacts with the main sequencer in the following ways:
D The tempo set on the transport panel is used for all playback.
If tempo automation is used in the main sequencer, Redrum will follow this.
D If you start playback for the main sequencer (on the transport panel), Redrum will automatically start as well
(provided the pattern sequencer hasn’t been disabled - see below).
D You can mute and solo Redrum tracks in the sequencer.
If the Redrum has a track in the sequencer and you mute this track, Redrum will automatically be muted as well.
This is indicated by a Mute indicator on the device panel.

This Redrum device is muted.

D You can also run Redrum separately (without starting the main sequencer) by clicking the Run button on the
device panel.
This starts the built-in pattern sequencer in the device. To stop playback, click the Run button again or click the
Stop button on the Transport panel.

The Run button on the Redrum.

D If you are running Redrum separately and start playback of the main sequencer, the pattern device will auto-
matically restart in sync with the sequencer.
D Pattern changes can be controlled by pattern automation in the main sequencer.

474 REDRUM DRUM COMPUTER

 D The sound sources can also be played by the main sequencer, or via MIDI.
You can combine the built-in pattern playback with playback from the main sequencer or via MIDI. For example,
this allows you to add variations or fills to a basic pattern.
It is also possible to disable the pattern sequencer totally, converting the device to a pure sound module. This is
done by deactivating the Enable Pattern Section switch.

Selecting patterns
The Redrum has 32 pattern memories, divided into four banks (A, B, C, D).

The Bank and Pattern buttons for the Redrum pattern sequencer.
D To select a pattern in the current bank, click on the desired Pattern button (1-8).
If you like, you can assign computer key commands and/or MIDI messages to pattern selection.
D To select a pattern in another bank, first click the desired Bank button (A, B, C, D) and then click the Pattern
button.
Nothing happens until you click the Pattern button.
D If you select a new pattern during playback, the change will take effect on the next downbeat (according to the
time signature set in the transport panel).
If you automate pattern changes in the main sequencer, you can make them happen at any position.
D Note that you cannot load or save patterns - they are only stored as part of a song.
However, you can move patterns from one location to another (even between songs) by using the Cut, Copy and
Paste Pattern commands on the context menu.

Pattern tutorial
If you are unfamiliar with step programming patterns, the basic principle is very intuitive and easy to learn. Proceed as
follows:
1. Load a Redrum patch, if one isn’t already loaded.
2. Make sure an empty pattern is selected.
If you like, use the Clear Pattern command on the device context menu to be sure.

475 REDRUM DRUM COMPUTER

 3. Make sure that the “Enable Pattern Section” and the “Pattern” buttons are activated (lit).

4. Press the “Run” button.

There will be no sound, as no pattern steps have been recorded yet. But as you can see, the LEDs over the Step
button light up consecutively, moving from left to right, and then starts over. Each Step button represents one
“step” in the Pattern.
5. Select a Redrum channel, by clicking the “Select” button at the bottom of the channel.
The button lights up, indicating that this channel and the drum sound it contains is selected.

6. While in Run mode, press Step button 1, so that it lights up.

The selected sound will now play every time Step 1 is “passed over”.
7. Clicking other Step buttons so they light up will play back the selected sound as the sequencer passes those
steps.
Clicking on a selected (lit) step button a second time removes the sound from that step and the button goes dark
again. You can click and drag to add or remove steps quickly.
8. Select another Redrum channel to program steps for that sound.
Selecting a new sound or channel also removes the visual indications (static lit buttons) of step entries for the pre-
viously selected sound. The step buttons always show step entries for the currently selected sound.
9. Continue switching between sounds, and programming steps to build your pattern.
Note that you can erase or add step entries even if Run mode isn’t activated.

Setting pattern length

You may want to make settings for Pattern length, i.e the number of steps the pattern should play before repeating:

D Use the “Steps” spin controls to set the number of steps you wish the pattern to play.
The range is 1 to 64. You can always extend the number of steps at a later stage, as this will merely add empty
steps at the end of the original pattern. You could also make it shorter, but that would (obviously) mean that the
steps “outside” the new length won’t be heard. These steps aren’t erased though; if you raise the Steps value
again, the steps will be played back again.

476 REDRUM DRUM COMPUTER

 About the “Edit Steps” Switch
If you set the pattern length to more than 16 steps, the pattern steps following after the 16th won’t be visible, al-
though they will play back. To view and be able to edit the next 16 steps, you have to set the Edit Steps switch to 17-
32. To see and edit steps beyond 32 you set the switch to 33-48, and so on.

Setting pattern resolution

Redrum always follows the tempo setting on the transport panel, but you can also make Redrum play in different
“resolutions” in relation to the tempo setting. Changing the Resolution setting changes the length of each step, and
thereby the “speed” of the pattern.

Step dynamics
When you enter step notes for a drum sound, you can set the velocity value for each step to one of three values:
Hard, Medium or Soft. This is done by setting the Dynamic switch before entering the note.

The color of the step buttons reflect the dynamics for each step. Soft notes are light yellow, Medium notes are orange and Hard are red.
D When the Medium value is selected, you can enter Hard notes by holding down [Shift] and clicking.
In the same way, you can enter Soft notes by holding down [Option] (Mac) or [Alt] (Windows) and clicking. Note
that this doesn’t change the Dynamic setting on the device panel - it only affects the notes you enter.
D When you use different dynamics, the resulting difference in the sound (loudness, pitch, etc.), is governed by
the “VEL” knob settings for each drum channel (see “Redrum parameters”).
If no velocity amount is set for a drum channel, it will play back the same, regardless of the Dynamic setting.
D To change the dynamics for an already programmed step, set the switch to the dynamic value you wish to
change it to and click on the step.
! Note that if you are triggering Redrum via MIDI or from the main sequencer, the sounds will react to velocity
like any other audio device. The Dynamic values are there to offer velocity control when using the built-in pat-
tern sequencer.

477 REDRUM DRUM COMPUTER

 Pattern Shuffle
Shuffle is a rhythmic feature, that gives the music a more or less pronounced swing feel. It works by delaying all six-
teenth notes that fall in between the eighth notes. You can activate or deactivate shuffle individually for each Redrum
pattern by clicking the Shuffle button on the device panel.

The amount of shuffle is set globally with the Shuffle control on the I/O device - see “Shuffle”.

Flam

A flam is when you double-strike a drum, to create a rhythmic or dynamic effect. Applying flam to a step entry will add
a second “hit” to a drum sound. The flam amount knob determines the delay between the two hits.
To add a flam drum note, proceed as follows:
1. Activate flam by clicking the Flam button.
2. Click on a step to add a note (taking the Dynamic setting into account as usual).
A red LED is lit above the step to indicate that flam will be applied to that step.
3. Use the Flam knob to set the desired amount of flam.
The flam amount is global for all patterns in the device.
D To add or remove flam to or from an existing step note, click directly on the corresponding flam LED.
You can also click and drag on the LEDs to add or remove several flam steps quickly.
D Applying flam to several consecutive step entries is a quick way to produce drum rolls.
By adjusting the Flam knob you can create 1/32 notes even if the step resolution is 1/16, for example.

The Pattern Enable switch

If you deactivate the “Pattern” button the pattern playback will be muted, starting at the next downbeat (exactly as if
you had selected an empty (silent) pattern). For example, this can be used for bringing different pattern devices in
and out of the mix during playback.

478 REDRUM DRUM COMPUTER

 The Enable Pattern Section switch

If this is off, Redrum will function as a pure “sound module”, i.e. the internal Pattern sequencer is disengaged. Use this
mode if you wish to control Redrum exclusively from the main sequencer or via MIDI (see “Using Redrum as a sound
module”).

Pattern functions
When a Redrum device is selected, you will find some specific pattern functions on the Edit menu (and on the device
context menu):

| Function | Description
Shift Pattern Left/Right These functions move all notes in the pattern one step to the left or right.
Shift Drum Left/Right The Shift Drum functions move all notes for the selected drum channel (the channel for
which the Select button is lit) one step to the left or right.
Randomize Pattern Creates a random pattern. Random patterns can be great starting points and help you get
new ideas.
Randomize Drum Creates a random pattern for the selected drum sound only - the notes for the other
drum sound channels are unaffected.
Alter Pattern The Alter Pattern function modifies the selected pattern by “shuffling” the current pattern
notes and redistributing them among the drum sounds at random. This creates a less
chaotic pattern than the “Randomize Pattern” function.
Note that there must be something in the pattern for the function to work on - using an
Alter function on an empty pattern will not do anything.
Alter Drum Works like the “Alter Pattern” function, but affects the selected drum sound only.

Chaining patterns
When you have created several patterns that belong together, you most probably want to make these play back in a
certain order. This is done by recording pattern automation into the main sequencer.

479 REDRUM DRUM COMPUTER

 Copy MIDI Notes to a sequencer track
It’s possible to copy the MIDI Notes of the currently selected Pattern to a track in your DAW’s sequencer:
1. Click to select the desired Pattern.
2. Click the “Drag MIDI Notes to track in sequencer” icon and drag to the destination track in the sequencer:

! Be sure to disable the Enable Pattern Section function on the Redrum panel afterwards, to avoid “doubled
notes” during playback.

480 REDRUM DRUM COMPUTER

 Redrum parameters
Drum sound settings
Redrum features ten drum sound channels that can each be loaded with a sample. Although they are basically simi-
lar, there are three “types” of drum sound channels, with slightly different features. This makes some channels more
suitable for certain types of drum sounds, but you are of course free to configure your drum kits as you like.
On the following pages, all parameters will be listed. If a parameter is available for certain drum sound channels only,
this will be stated.

Mute & Solo

At the top of each drum sound channel, you will find a Mute (M) and a Solo (S) button. Muting a channel silences its
output, while Soloing a channel mutes all other channels. Several channels can be muted or soloed at the same time.
You can also use keys on your MIDI keyboard to mute or solo individual drum sounds in real time.
D The keys C2 to E3 (white keys only) will mute individual drum channels starting with channel 1.
The sounds are muted for as long as you hold the key(s) down.
D The keys C4 to E5 (white keys only) will solo individual drum channel, starting with channel 1.
The sounds are soloed for as long as you hold the key(s) down.
C2 C3 C4 C5

1 2 3 4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10

Mute Solo

This is a great way to bring drum sounds in and out of the mix when playing Reason live. You can also record the
drum channel Mutes in the main sequencer, just like any other controller (see “Recording parameter automation”).

The Effect Sends (S1 & S2)

On the back panel of Redrum you will note two audio connections labeled “Send Out” 1 and 2. When you create a
Redrum device, these will by default be auto-routed to the first two “Chaining Aux” inputs on the Mixer device (pro-
vided that these inputs aren’t already in use).
This feature allows you to add effects to independent drum sounds in the Redrum.

481 REDRUM DRUM COMPUTER

 D Raising the S1 knob for a drum sound channel will send the sound to the first send effect connected to the
mixer.
Similarly, the S2 knob governs the send level to the second send effect in the mixer.
D Note that there must be send effects connected to the AUX Sends and Returns in the mixer for this to work.

D Also note that if Redrum is soloed in the Mixer the effect sends will be muted.

D Another way to add independent effects to drum sounds is to use the independent drum outputs.
See “Connections”.

Pan

Sets the Pan (stereo position) for the channel.

D If the LED above the Pan control is lit, the sound uses a stereo sample.
In that case, the Pan control serves as a stereo balance control.

Level and Velocity

The Level knob sets the volume for the channel. However, the volume can also be affected by velocity (as set with the
Dynamic value, or as played via MIDI). How much the volume should be affected by velocity is set with the “Vel” knob.
D If the Vel knob is set to a positive value, the volume will become louder with increasing velocity values.
The higher the Vel value, the larger the difference in volume between low and high velocity values.
D A negative value inverts this relationship, so that the volume decreases with higher velocity values.

D If the Vel knob is set to zero (middle position), the sound will play at a constant volume, regardless of the ve-
locity.
When Vel is set to zero, the LED above the knob goes dark.

Length and the Decay/Gate switch

The Length knob determines the length of the drum sound, but the result depends on the setting of the Decay/Gate
switch:
D In Decay mode (switch down), the sound will decay (gradually fade out) after being triggered. The decay time is
determined by the Length setting.
In this mode, it doesn’t matter for how long a drum note is held (if played back from the main sequencer or via
MIDI) - the sound will play the same length for short notes as for long notes. This is the traditional “drum machine”
mode.

482 REDRUM DRUM COMPUTER

 D In Gate mode (switch up), the sound will play for the set Length, and then be cut off.
Furthermore, if a sound set to Gate mode is played from the main sequencer, from a CV/Gate device or via MIDI,
the sound will be cut off when the note ends or after the set Length, depending on which comes first. Or in other
words, the sound plays for as long as you hold the note, but the Length setting serves as the maximum length for
the sound.
There are several uses for the Gate mode:
• For “gated” drum sounds, when the tail of the sound is abruptly cut off as an effect.
• For when you want to use very short sounds, and don’t want them to “lose power” by being faded out.
• For when you play the Redrum from the sequencer or via MIDI, with sounds for which the length is important
(e.g. when using the Redrum as a sound effects module).
! Audio samples sometimes contain a “loop”, which is set by editing the audio in a sample editor. This loop re-
peats a part of the sample to produce sustain as long as you hold down a note. Drum samples usually don’t
contain loops, but who is to say that Redrum should only play drum samples?
Note that if a sample contains a loop, and Length is set to maximum, the sound will have infinite sustain, in
other words it will never become silent, even if you stop playback. Decreasing the Length setting solves this
problem.

Pitch

Sets the pitch of the sound. The range is +/- 1 octave.

D When the pitch is set to any other value than 0, the LED above the knob lights up to indicate that the sample
isn’t played back at its original pitch.

Pitch Bend

By setting the Bend knob to a positive or negative value, you specify the start pitch of the sound (relative to the Pitch
setting). The pitch of the sound will then be bent to the main Pitch value. Thus, selecting a positive Bend value will
cause the pitch to start higher and bend down to the original Pitch, and vice versa.
D The Rate knob determines the bend time - the higher the value, the slower the bend.
D The Vel knob determines how the Bend amount should be affected by velocity.
With a positive Vel value, higher velocity results in wider pitch bends.
D The Bend and Vel knobs have LEDs that light up when the functions are activated (i.e. when a value other than
zero is selected).
! Pitch bend is available for drum sound channels 6 and 7 only.

483 REDRUM DRUM COMPUTER

 Tone

The Tone knob determines the brightness of the drum sound. Raising this parameter results in a brighter sound. The
Vel knob determines whether the sound should become brighter (positive Vel value) or darker (negative Vel value)
with higher velocity.
D The Tone and Vel knobs have LEDs that light up when the functions are activated (i.e. when a value other than
zero is selected).
! The Tone controls are available for drum sound channels 1, 2 and 10 only.

Sample Start

The Start parameter allows you to adjust the start point of the sample. The higher the Start value, the further the start
point is moved “into” the sample. If you set the Start Velocity knob to a positive amount, the sample start point is
moved forward with higher velocities. A negative Start Velocity amount inverts this relationship.
D When Start Velocity is set to any other value than zero, the LED above the knob lights up.
D A negative Start Velocity amount is only useful if you have set the Start parameter to a value higher than 0.
By raising the Start value a bit and setting Start Velocity to a negative value, you can create rather realistic velocity
control over some drum sounds. This is because the very first transients in the drum sound will only be heard when
you play hard notes.
! The Sample Start settings are available for drum sound channels 3-5, 8 and 9.

Global settings
Channel 8 & 9 Exclusive

If this button is activated, the sounds loaded into drum channels 8 and 9 will be exclusive. In other words, if a sound
is played in channel 8 it will be silenced the moment a sound is triggered in channel 9, and vice versa.
The most obvious application for this feature is to “cut off” an open hi-hat with a closed hi-hat, just like a real one
does.

484 REDRUM DRUM COMPUTER

 High Quality Interpolation

When this is activated, the sample playback is calculated using a more advanced interpolation algorithm. This results
in better audio quality, especially for drum samples with a lot of high frequency content.
D High Quality Interpolation uses more computer power - if you don’t need it, it’s a good idea to turn it off!
Listen to the drum sounds in a context and determine whether you think this setting makes any difference.

Master Level
The Master Level knob in the top left corner of the device panel governs the overall volume from Redrum.

Using Redrum as a sound module

The drum sounds in Redrum can be played via MIDI notes. Each drum sound is triggered by a specific note number,
starting at C1 (MIDI note number 36):
C1 C2

2 4 7 9

1 3 5 6 8 10

This allows you to play Redrum live from a MIDI keyboard or a MIDI percussion controller, or to record or draw drum
notes in the main sequencer. If you like, you can combine pattern playback with additional drum notes, such as fills
and variations. However:
! If you want to use Redrum purely as a sound module (i.e. without pattern playback) you should make sure that
the “Enable Pattern Section” button is deactivated (see “The Enable Pattern Section switch”), otherwise the
Redrum pattern sequencer will start as soon as you start the main sequencer.

485 REDRUM DRUM COMPUTER

 Connections

On the back of the Redrum you will find the following connections:

For each drum sound channel:

| Connection | Description
Audio Outputs There are individual audio outputs for each drum sound channel, allowing you to route a drum sound to
a separate channel in the mixer, possibly via insert effects, etc.
For mono sounds, use the “Left (Mono)” output (and pan the sound using the Pan control in the mixer).
When you use an individual output for a sound, the sound is automatically excluded from the master
stereo output.
Gate Out This sends out a gate signal when the drum sound is played (from a pattern, via MIDI or by using the
Trigger button on the device panel). This lets you use the Redrum as a “trig sequencer”, controlling
other devices.
The length of the gate signal depends on the Decay/Gate setting for the sound: In Decay mode, a
short “trig pulse” is sent out, while in Gate mode, the gate signal will have the same length as the drum
note.
Gate In Allows you to trigger the sound from another CV/Gate device. All settings apply, just as when playing
the drum sound conventionally.
Pitch CV In Lets you control the pitch of the drum sound from another CV device.

Other
| Connection | Description
Send Out 1-2 Outputs for the send signals controlled with the S1 and S2 knobs.
Stereo Out This is the master stereo output, outputting a mix of all drum sounds (except those for which you use
individual outputs).

486 REDRUM DRUM COMPUTER

Chapter 27
Rytmik Drum Machine
 Introduction

The Rytmik Drum Machine device is an eight-channel drum sample player. Rytmik features a Distortion effect and a
Low Cut + Hi Cut filter per drum channel. There are also two send effects - Reverb and Delay - that are shared
among the eight drum channels, plus a master section with a Master Compressor, Master Pitch and a Master Filter.
! Please, note that this device is not available in Reason Lite Rack Plugin.

488 RYTMIK DRUM MACHINE

 Panel overview
The Rytmik front panel contains the following sections:

1
2 3

4 4 4 4 4 4 4 4 5

The Rytmik front panel sections.

• 1. Patch selector (for browsing, loading and saving patches).
• 2. Sample playback and editing section (for the currently selected Drum Channel).
• 3. Distortion and Low Cut + Hi Cut Filter (for the currently selected Drum Channel).
• 4. Drum Channel sections (for playing back samples and for selecting Drum Channel to edit).
• 5. Send Effects and Compressor section (global for all eight Drum Channels).

489 RYTMIK DRUM MACHINE

 Signal flow
The picture below shows the basic signal flow in Rytmik:

Reverb
Send (x8) Lvl (x1)

Sample Playback (x8) Distortion (x8) Filter (x8)

Delay
Pan (x8) Send (x8)
Clean

Vol (x8)

Compressor

EQ/Filter

Volume

Out

Rytmik signal flowchart.

• Rytmik features eight Sample Playback engines (one per Drum Channel).
Each Sample Playback engine features a Sample section, where you can select any of the built-in samples. Here,
you can also set the Sample Start, Sample End, Pan, Pitch, Fade In and Fade Out parameters.
• The audio from the Sample Playback engines are then routed to a Distortion insert effect (one per Drum Chan-
nel).
• The signal is then routed to a Filter section (one per Drum Channel).
The signal can then be processed with a Low Cut and Hi Cut filter.
• The signal from each Drum Channel can then be panned in stereo - and have its own individual volume.
• Each Drum Channel can also use the two global Send Effects (Reverb and Delay).
The signal levels to each of these send effects can be set individually for each Drum Channel.
• Finally, the signals of all eight Drum Channels, plus the Send Effects are mixed and output as a stereo signal
via a Master Compressor and a Master Filter.
It’s also possible to output the desired Drum Channel signals individually via separate audio outputs - if you want
to process the signals outside of the Rytmik device. If you do that, the signal will be output after the Filter section
of the Drum Channel, bypassing the Send Effects and the Master FX.

Global controls
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

490 RYTMIK DRUM MACHINE

 The Drum Channel sections

Auditioning samples
D Click a Drum Channel button to play back the sample of the corresponding Drum Channel.
By clicking the Drum Channel button you also automatically select the Drum Channel (see below).

• If you are using a MIDI Keyboard/On-screen Piano Keys you can play back the samples of the Drum Channels
from Key C0 to G0.

Selecting a Drum Channel

D Click a Drum Channel button to select the desired Drum Channel:

Muting and soloing Drum Channels

D Click the M(ute) or S(olo) button in a Drum Channel to mute or solo the desired Drum Channel:

Setting the Drum Channel volumes

D Drag the Volume slider up/down - or just click - to set the volume of a Drum Channel:

491 RYTMIK DRUM MACHINE

 Setting the Send Effect levels
D Turn the Delay and Reverb knobs to set the send levels from the Drum Channel to the respective Send Effect:
See “The Master FX section” for more information about the Send Effects.

The Sample Playback section

The Sample Playback section contains all sample related controls and parameters for the currently selected Drum
Channel. The currently selected Drum Channel is indicated by the lit Drum Channel selection button (see “Selecting a
Drum Channel”).
The Sample Playback section features the following parameters and controls:

Selecting Samples

D Select and load a sample either by clicking the triangular arrow buttons on either side of the sample name - or
by clicking the sample name and selecting from the pop-up menu.
The pop-up menu features eight sub-groups with different types of samples.
! All samples in Rytmik are embedded in the device itself, so when you save a Rytmik patch the samples are al-
ways included (as opposed to other sampler devices in Reason).

492 RYTMIK DRUM MACHINE

 Setting the Sample Start and End

D Click and drag the Sample Start handle sideways to change the where in the sample playback should begin.
D Click and drag the Sample End handle sideways to change the where in the sample playback should stop.

Setting the Panning

D Click and drag up/down in the Pan box to place the sample in the stereo panorama.
Range: 100L to 100R.

Setting the Pitch

D Click and drag up/down in the Pitch box to set the pitch of the sample.
Range: +/- 1200 cents.

493 RYTMIK DRUM MACHINE

 Setting Fade In and Fade Out

D Click and drag up/down in the Fade In and/or Fade Out boxes to apply a fade in/out of the sample.
Any fade in/out is shown graphically in the sample display.

The Insert Effects section

The Insert Effects section consists of a Distortion effect and a Low Cut and Hi Cut Filter.

Distortion

The Distortion effect is a transistor type of distortion.

D Click and drag up/down in the Distortion box to adjust the input gain to the distortion effect.
A high value will overdrive the pre-amp and generate more distortion.
Range: 0-100%, where “0” is completely dry/off.

494 RYTMIK DRUM MACHINE

 Low Cut and Hi Cut Filter

The Filter features a Low Cut (Highpass) and a Hi Cut (Lowpass) filter. Perfect for removing any rumble (Lo Cut)
and/or hiss (Hi Cut), for example.
D Click and drag up/down in the Low Cut box to set the cutoff frequency for the highpass filter.
Alternatively click the left part of the filter curve in the display and drag sideways.
Range: 20 Hz to 25 kHz.
D Click and drag up/down in the Hi Cut box to set the cutoff frequency for the lowpass filter.
Alternatively click the right part of the filter curve in the display and drag sideways.
Range: 20 Hz to 25 kHz.
! Note that the Low Cut and Hi Cut cutoff frequencies can also be on opposite sides of each other, which means
that the level of the sample could eventually drop to zero with extreme settings.

The Master FX section

The Master FX section features controls for the Delay and Reverb send effects, as well as for the master bus Com-
pressor. The Send Effects can be used by all Drum Channels simultaneously, and the effects are active simultane-
ously.

495 RYTMIK DRUM MACHINE

 Delay

This is a delay with two different modes - Stereo and Ping Pong. The delay repeats are tempo synced to the main se-
quencer and you can select the desired time division.

Mode

D Click the Mode box and select the desired mode from the pop-up.
“Stereo” repeats the delay in stereo in a fixed centered position.
“Ping Pong” repeats the delays, alternating between the left and right channels.

Time

D Click and drag the Time box up/down to set the time division of the tempo-synced delay repeats.
The tempo is hard-wired to the main sequencer tempo.
Time divisions: 1/1, 1/2D, 1/1T, 1/2, 1/4D, 1/2T, 1/4, 1/8D, 1/4T, 1/8, 1/16D, 1/8T and 1/16

Feedback

D Click and drag the Feedback box up/down to set the number of delay repeats.
Alternatively, click and drag in the display to set the Feedback amount.
Range: 0-100%, where “0” is one single delay repeat.

496 RYTMIK DRUM MACHINE

 Reverb

This is a stereo reverb with six different Modes (reverb algorithms).

Mode

D Click the Mode box and select the desired reverb algorithm from the pop-up.
The following reverb types can be selected:
• Room
• Large Room
• Culvert
• Plate
• Gated
• Hall

Decay

D Click and drag the Decay box up/down to set the length of the reverb effect.
Alternatively, click and drag in the display to set the decay length.
The Decay amount is also shown graphically in the display.

497 RYTMIK DRUM MACHINE

 Low Cut and Hi Cut

This is essentially a combination of a highpass and a lowpass filter.

D Raise the Low Cut value to cut the low frequencies of the reverb signal and make the reverb effect less
“muddy”.
Range: 20 Hz to 25 kHz.
D Lower the Hi Cut value to cut off the high frequencies of the reverb, thereby creating a smoother, warmer ef-
fect.
Range: 20 Hz to 25 kHz.
! Note that the Low Cut and Hi Cut cutoff frequencies can also be on opposite sides of each other, which means
that the reverb level could eventually drop to zero with extreme settings.

Compressor

This is a stereo compressor, which acts on the total mix of all Drum Channels. It is always active, but if you don’t want
any compression effect you can set the controls so that no compression is produced. The gain reduction is shown by
the meter.

Threshold

This is the threshold level above which the compression sets in. Signals with levels above the threshold will be af-
fected, signals below it will not. In practice, this means that the lower the Threshold setting, the more the compression
effect.
D Click and drag the Threshold box up/down to set the threshold level.
Range: -60 dB to 0 dB
q For no compression effect at all, set the Threshold to “0 dB”.

498 RYTMIK DRUM MACHINE

 Ratio

This specifies the amount of gain reduction applied to the signal above the set threshold. A high Ratio value makes
the sound less dynamic and more “even” in level.
D Click and drag the Ratio box up/down to set the compression ratio.
Range: 1:1 to 20:1
q For no compression effect at all, set the Ratio to “1.00”.

Attack
This governs how quickly the compressor applies the gain reduction when signals rise above the set Threshold (see
above). If you raise the Attack value, the response will be slower, allowing more of the signal to pass through the
compressor unaffected. Typically, this is used for preserving the attacks of the sounds.

D Click and drag the Attack box up/down to set the compressor attack time.
Range: 0-200 ms

Release
When the signal level drops below the set Threshold (see above), this determines how long it takes before the com-
pressor lets the sound through unaffected. Set this to short values for intense, “pumping” compressor effects, or to
longer values for a smoother change of the dynamics.

D Click and drag the Release box up/down to set the compressor release time.
Range: 0-200 ms

499 RYTMIK DRUM MACHINE

 Master Pitch

D Turn the Master Pitch knob to adjust the pitches of all Drum Channels equally.
Range: +/-1200 cents.

Master Reverb

D Turn the Master Reverb knob to adjust the Reverb return level.
Range: +/-100%.
! Note that this control is bipolar, i.e. you could attenuate or amplify the reverb return level. This means that if
any of the Reverb Amount knobs for the Drum Channels are < 0 dB, raising the Master Reverb knob to a posi-
tive value will increase the Reverb level for these Drum Channels. The level can never exceed 0 dB, though.

Master Filter

The Master Filter is a combined highpass and lowpass filter, which can be used for cutting out low or high frequen-
cies in the total mix signal. At the default 0% setting the output signal is completely unaffected (not filtered at all).
D Turn the Master Filter knob to adjust the Low Cut and Hi Cut effect.
Range: +/-100%.

500 RYTMIK DRUM MACHINE

 Master Volume

D Drag the Master Volume slider up/down - or just click - to set the output volume of the total mix.
Range: -inf to 6.00 dB

501 RYTMIK DRUM MACHINE

 Connections

! Remember that CV connections are NOT stored in the Rytmik patches! If you want to store CV connections be-
tween devices, put them in a Combinator device and save the Combi patch.

Drum Gate In/Out

The Drum Gate inputs allow you to play Rytmik from another CV/Gate device. The Drum Gate inputs respond to
“Note On/Off” along with Velocity.
The Drum Gate outputs allow you to control other CV/Gate equipped devices from Rytmik. The Drum Gate outputs
send out “Note On/Off” along with Velocity.

Separate Outputs
The separate outputs can be used for routing individual Drum Channels to external destinations, for further process-
ing.
! Note that Drum Channels routed via separate outputs are automatically disconnected from the Master FX sec-
tion. Note, though, that the signal could still be sent to (and heard via) the Send Effects (Delay and Reverb) on
the Main Outputs.

Main Output L & R

These are the main audio outputs. When you create a new Rytmik device, these outputs are auto-routed to the first
available outputs in the I/O device.

502 RYTMIK DRUM MACHINE

Chapter 28
Subtractor Synthesizer
 Introduction
Subtractor is an analog-type polyphonic synthesizer based on subtractive synthesis, the method used in analog syn-
thesizers. This chapter will go through all parameters of each section of Subtractor. In addition to the parameter de-
scriptions, the chapter also includes a few tips and tricks to help you get the most out of the Subtractor synthesizer.
! It is recommended that you start with default settings (an “Init Patch”) if you intend to follow the examples in
this chapter, unless otherwise is stated. An Init Patch is created by selecting “Reset Device” from the device’s
context menu, or from the Edit menu. If you wish to keep the current settings, save them before initializing.
The Subtractor has the following basic features:
D Up to 99 Voice Polyphony.
You can set the number of voices for each Patch.
D Dual Filters.
A combination of a multimode filter and a second, linkable, lowpass filter allows for complex filtering effects. See
“The Filter Section”.
D Two Oscillators, each with 32 waveforms.
See “The Oscillator Section”.
D Frequency Modulation (FM).
See “Frequency Modulation (FM)”.
D Oscillator Phase Offset Modulation.
This is an unique Subtractor feature that generates waveform variations. See “Phase Offset Modulation”.
D Two Low Frequency Oscillators (LFO’s)
See “LFO Section”.
D Three Envelope Generators.
See “Envelopes - General”.
D Extensive Velocity Control.
See “Velocity Control”.
D Extensive CV/Gate Modulation possibilities.
See “Connections”.

Loading and Saving Patches

Loading and saving patches is done in the same way as with any other Reason device - see “Loading patches” and
“Saving patches”.

504 SUBTRACTOR SYNTHESIZER

 The Oscillator Section

Subtractor provides two oscillators. Oscillators are the main sound generators in Subtractor, the other features are
used to shape the sound of the oscillators. Oscillators generate two basic properties, waveform and pitch (frequency).
The type of waveform the oscillator produces determines the harmonic content of the sound, which in turn affects the
resultant sound quality (timbre). Selecting a oscillator waveform is usually the starting point when creating a new
Subtractor Patch from scratch.

Oscillator 1 Waveform

Oscillator 1 provides 32 waveforms. The first four are standard waveforms, and the rest are “special” waveforms,
some of which are suitable for emulating various musical instrument sounds.
q It is worth noting here that all waveforms can be radically transformed using Phase offset modulation (see
“Phase Offset Modulation”).
D To select a waveform, click the spin controls to the right of the “Waveform” LED display.
The first 4 basic waveforms are shown as standard symbols, and the special waveforms are numbered 5 - 32.

505 SUBTRACTOR SYNTHESIZER

 Here follows a brief description of the Subtractor waveforms:
! Please note that the descriptions of the waveforms sound or timbre is merely meant to provide a basic guide-
line, and shouldn’t be taken too literally. Given the myriad ways you can modulate and distort a waveform in
Subtractor, you can produce extremely different results from any given waveform.

| Waveform | Description
Sawtooth This waveform contains all harmonics and produces a bright and rich sound. The Sawtooth is perhaps the
most “general purpose” of all the available waveforms.

Square A square wave only contains odd number harmonics, which produces a distinct, hollow sound.

Triangle The Triangle waveform generates only a few harmonics, spaced at odd harmonic numbers. This produces
a flute-like sound, with a slightly hollow character.

Sine The sine wave is the simplest possible waveform, with no harmonics (overtones). The sine wave produces
a neutral, soft timbre.

5 This waveform emphasizes the higher harmonics, a bit like a sawtooth wave, only slightly less bright-sound-
ing.
6 This waveform features a rich, complex harmonic structure, suitable for emulating the sound of an acoustic
piano.
7 This waveform generates a glassy, smooth timbre. Good for electric piano-type sounds.
8 This waveform is suitable for keyboard-type sounds such as harpsichord or clavinet.
9 This waveform is suitable for electric bass-type sounds.
10 This is a good waveform for deep, sub-bass sounds.
11 This produces a waveform with strong formants, suitable for voice-like sounds.
12 This waveform produces a metallic timbre, suitable for a variety of sounds.
13 This produces a waveform suitable for organ-type sounds.
14 This waveform is also good for organ-type sounds. Has a brighter sound compared to waveform 13.
15 This waveform is suitable for bowed string sounds, like violin or cello.
16 Similar to 15, but with a slightly different character.
17 Another waveform suitable for string-type sounds.
18 This waveform is rich in harmonics and suitable for steel string guitar-type sounds.
19 This waveform is suitable for brass-type sounds.
20 This waveform is suitable for muted brass-type sounds.
21 This waveform is suitable for saxophone-like sounds.
22 A waveform suitable for brass and trumpet-type sounds.
23 This waveform is good for emulating mallet instruments such as marimba.
24 Similar to 23, but with a slightly different character.
25 This waveform is suitable for guitar-type sounds.
26 This is a good waveform for plucked string sounds, like harp.
27 Another waveform suitable for mallet-type sounds (see 23-24), but has a brighter quality, good for vibra-
phone-type sounds.

506 SUBTRACTOR SYNTHESIZER

 | Waveform | Description
28 Similar to 27, but with a slightly different character.
29 This waveform has complex, enharmonic overtones, suitable for metallic bell-type sounds.
30 Similar to 29, but with a slightly different character. By using FM and setting the Osc Mix to Osc 1, this and
the following two waveforms can produce noise.
31 Similar to 30, but with a slightly different character.
32 Similar to 30, but with a slightly different character.

Setting Oscillator 1 Frequency - Octave/Semitone/Cent

By clicking the corresponding up/down buttons you can tune, i.e. change the frequency of Oscillator 1 in three ways:
D In Octave steps
The range is 0 - 9. The default setting is 4 (where “A” above middle “C” on your keyboard generates 440 Hz).
D In Semitone steps
Allows you to raise the frequency in 12 semitone steps (1 octave).
D In Cent steps (100th of a semitone)
The range is -50 to 50 (down or up half a semitone).

Oscillator Keyboard Tracking

Oscillator 1 has a button named “Kbd. Track”. If this is switched off, the oscillator pitch will remain constant, regard-
less of any incoming note pitch messages, although the oscillator still reacts to note on/off messages. This can be
useful for certain applications:
D When Frequency Modulation (FM - see “Frequency Modulation (FM)”) or Ring Modulation (see “Ring Modula-
tion”) is used.
This produces enharmonic sounds with very varying timbre across the keyboard.
D For special effects and non-pitched sounds (like drums or percussion) that should sound the same across the
keyboard.

Using Oscillator 2

You activate Osc 2 by clicking the button next to the text “Osc 2“. Setting oscillator frequency and keyboard tracking
is identical to Oscillator 1.

507 SUBTRACTOR SYNTHESIZER

 Adding a second oscillator enables many new modulation possibilities which can produce richer timbres. A basic ex-
ample is to slightly detune (+/– a few cents) one of the oscillators. This slight frequency offset causes the oscillators
to “beat” against each other, producing a wider and richer sound. Also, by combining two different waveforms, and
adding frequency or ring modulation, many new timbres can be created.

Oscillator Mix

The Osc Mix knob determines the output balance between Osc 1 and Osc 2. To be able to clearly hear both oscilla-
tors, the “Osc Mix” knob should be set somewhere around the center position. If you turn the Mix knob fully to the left,
only Osc 1 will be heard, and vice versa. [Command]/[Ctrl]-clicking the knob sets the Mix parameter to center posi-
tion.

Oscillator 2 Waveform
The waveform alternatives for Oscillator 2 are identical to those of Oscillator 1.
However, the Noise Generator provides a third sound generating source (in addition to the two oscillators) in Sub-
tractor, and could be regarded as an “extra” waveform for Oscillator 2, as it is internally routed to the Oscillator 2 out-
put. See below for a description of the Noise Generator.

Noise Generator

The Noise Generator could be viewed as an oscillator that produces noise instead of a pitched waveform. Noise can
be used to produce a variety of sounds, the classic example being “wind” or “rolling wave” sounds, where noise is
passed through a filter while modulating the filter frequency. Other common applications include non-pitched sounds
like drums and percussion, or simulating breath noises for wind instruments. To use the Noise Generator, select an
Init Patch and proceed as follows:
1. Turn Osc 2 off.
2. Click the button (in the Noise Generator section) to activate the Noise Generator.
If you play a few notes on your MIDI instrument you should now hear Osc1 mixed with the sound of the Noise
Generator.
3. Turn the Mix knob fully to the right, and play a few more notes.
Now just the Noise Generator will be heard.
D Thus, the output of the Noise Generator is internally routed to Osc 2.
If you switch Osc 2 on, the noise will be mixed with the Osc 2 waveform.

508 SUBTRACTOR SYNTHESIZER

 There are three Noise Generator parameters. These are as follows:

| Parameter | Description
Noise Decay This controls how long it takes for the noise to fade out when you play a note. Note that this is inde-
pendent from the Amp Envelope Decay parameter, allowing you to mix a short “burst” of noise at the
very beginning of a sound, i.e. a pitched sound that uses oscillators together with noise.
Noise Color This parameter allows you to vary the character of the noise. If the knob is turned fully clockwise, pure
or “white” noise (where all frequencies are represented with equal energy) is generated. Turning the
knob anti-clockwise produces a gradually less bright sounding noise. Fully anti-clockwise the noise
produced is an earthquake-like low frequency rumble.
Level Controls the level of the Noise Generator.

Phase Offset Modulation

A unique feature of the Subtractor oscillators is the ability to create an extra waveform within one oscillator, to offset
the phase of that extra waveform, and to modulate this phase offset. By subtracting or multiplying a waveform with a
phase offset copy of itself, very complex waveforms can be created. Sounds complicated? Well, the theory behind it
might be, but from a user perspective it is just a method of generating new waveforms from existing waveforms.
A seasoned synth programmer using Subtractor for the first time may wonder why the Subtractor oscillators (seem-
ingly) cannot provide the commonly used pulse waveform and the associated pulse width modulation (PWM). Or os-
cillator sync, another common feature in analog synthesizers. The simple answer is that Subtractor can easily create
pulse waveforms (with PWM) and oscillator sync-sounds, and a lot more besides, partly by the use of phase offset
modulation.

Each oscillator has it's own Phase knob and a selector button. The Phase knob is used to set the amount of phase
offset, and the selector switches between three modes:
• Waveform multiplication (x)
• Waveform subtraction (–)
• No phase offset modulation (o).

509 SUBTRACTOR SYNTHESIZER

 When phase offset modulation is activated, the oscillator creates a second waveform of the same shape and offsets
it by the amount set with the Phase knob. Depending on the selected mode, Subtractor then either subtracts or mul-
tiplies the two waveforms with each other. The resulting waveforms can be seen in the illustration below.
1. The two offset Ampl.
waveforms:

t.

2. The result of Ampl.

subtraction:

t.

3. The result of Ampl.

multiplication:

t.

• In example 1, we see two sawtooth waves with a slight offset.

• Example 2 shows that subtracting one slightly offset sawtooth wave from the other, produces a pulse wave. If
you modulate the Phase offset parameter (with for example an LFO), the result will be pulse width modulation
(PWM).
• Example 3 shows the resulting waveform when multiplying the offset waves with each other. As you can see
(and hear if you try it), multiplying waveforms can produce very dramatic and sometimes unexpected results.
Using phase offset modulation can create very rich and varied timbres, especially when used along with LFO or En-
velopes to modulate the phase offset.
q To get a “feel” for this concept, you could study Patches that use phase offset modulation, and maybe tweak
some of the Phase Offset parameters to find out what happens. Try “SyncedUp” in the Polysynth category in
the Factory Soundbank for an example of osc sync or “Sweeping Strings” (in the Pads category) for an exam-
ple of PWM.
! Note that if you activate waveform subtraction with a Phase offset set to “0” for an oscillator, the second wave-
form will cancel out the original waveform completely, and the oscillator output will be silent. If you set the
Phase Offset knob to any other value than zero, the sound returns.

510 SUBTRACTOR SYNTHESIZER

 Frequency Modulation (FM)

In synthesizer-speak, Frequency Modulation, or FM, is when the frequency of one oscillator (called the “carrier”) is
modulated by the frequency of another oscillator (called the “modulator”). Using FM can produce a wide range of har-
monic and non harmonic sounds. In Subtractor, Osc 1 is the carrier and Osc 2 the modulator. To try out some of the
effects FM can produce, proceed as follows:
1. Select an Init Patch by selecting “Initialize Patch” from the Edit menu.
2. Activate Osc 2.
As you need both a carrier and a modulator to produce FM, turning the FM knob will not produce any effect unless
you first activate Osc 2. For classic FM sounds, use sine wave on oscillator 1 and triangle wave on oscillator 2.
3. Use the FM knob to set the FM amount to a value of about 50.
As you can hear, the timbre changes, but the effect isn’t very pronounced yet.
4. Turn the Osc Mix knob fully to the left, so that only the sound of Osc 1 is heard.
The modulator (Osc 2) still affects Osc 1, even though the Osc 2 output is muted.
5. Now, hold down a note on your MIDI keyboard and tune Osc 2 a fifth up from the original pitch by setting the
Osc 2 frequency “Semi” parameter to a value of 7.
As you can hear, for each semitone step you vary the Osc 2 frequency, the timbre changes dramatically. Setting
Osc 2 frequency to certain musical intervals (i.e. fourth, fifth or octave semitone steps) produces harmonic, rich
timbres, almost like tube distortion. Setting Osc 2 to non-musical intervals usually results in complex, enharmonic
timbres.
q Experiment with different oscillator parameters such as phase offset modulation, changing the waveforms etc.
and listen to how they affect the sound of frequency modulation.

Using the Noise Generator as the Modulator source

As explained earlier, the Noise Generator is internally routed to the Osc 2 output. Hence, if you deactivate Osc 2, and
activate the Noise Generator while using FM, the noise will be used to frequency modulate Osc 1.
q With the Noise Generators default settings, this will sound much like colored noise. But by changing (lower-
ing) the Noise Generator Decay parameter, so that the noise modulates only the attack portion of the sound
can produce more interesting results. You could also use a combination of noise and Osc 2.

511 SUBTRACTOR SYNTHESIZER

 Ring Modulation

Ring Modulators basically multiply two audio signals together. The ring modulated output contains added frequencies
generated by the sum of, and the difference between, the frequencies of the two signals. In the Subtractor Ring Mod-
ulator, Osc 1 is multiplied with Osc 2 to produce sum and difference frequencies. Ring modulation can be used to
create complex and enharmonic, bell-like sounds.
1. Select an Init Patch by selecting “Initialize Patch” from the Edit menu.
Save any current settings you wish to keep before initializing.
2. Activate Ring Modulation with the button in the lower right corner of the oscillator section.
3. Activate Osc 2.
You need to activate Osc 2 before any ring modulation can happen.
4. Turn the Osc Mix knob fully to the right, so that only the sound of Osc 2 is heard.
Osc 2 provides the ring modulated output.
5. If you play a few notes while varying the frequency of either oscillator, by using the Semitone spin controls,
you can hear that the timbre changes dramatically.
If the oscillators are tuned to the same frequency, and no modulation is applied to either the Osc 1 or 2 frequency,
the Ring Modulator won’t do much. It is when the frequencies of Osc 1 and Osc 2 differ, that you get the “true”
sound of ring modulation.

512 SUBTRACTOR SYNTHESIZER

 The Filter Section

In subtractive synthesis, a filter is the most important tool for shaping the overall timbre of the sound. The filter sec-
tion in Subtractor contains two filters, the first being a multimode filter with five filter types, and the second being a
low-pass filter. The combination of a multimode filter and a lowpass filter can be used to create very complex filter ef-
fects.

Filter 1 Type
With this multi-selector you can set Filter 1 to operate as one of five different types of filter. The five types are illus-
trated and explained on the following pages:
• 24 dB Lowpass (LP 24)
Lowpass filters lets low frequencies pass and cuts out the high frequencies. This filter type has a fairly steep roll-
off curve (24dB/Octave). Many classic synthesizers (Minimoog/Prophet 5 etc.) use this filter type.

The darker curve illustrates the roll-off curve of the 24dB Lowpass Filter. The lighter curve in the middle represents the filter
characteristic when the Resonance parameter is raised.

513 SUBTRACTOR SYNTHESIZER

 • 12 dB Lowpass (LP 12)
This type of lowpass filter is also widely used in analog synthesizers (Oberheim, early Korg synths etc.). It has a
gentler slope (12 dB/Octave), leaving more of the harmonics in the filtered sound compared to the LP 24 filter.

The darker curve illustrates the roll-off curve of the 12dB Lowpass Filter. The lighter curve in the middle represents the filter
characteristic when the Resonance parameter is raised.
• Bandpass (BP 12)
A bandpass filter cuts both high and low frequencies, while midrange frequencies are not affected. Each slope in
this filter type has a 12 dB/Octave roll-off.

The darker curve illustrates the roll-off curve of the Bandpass Filter. The lighter curve in the middle represents the filter characteristic
when the Resonance parameter is raised.

514 SUBTRACTOR SYNTHESIZER

 • Highpass (HP12)
A highpass filter is the opposite of a lowpass filter, cutting out lower frequencies and letting high frequencies pass.
The HP filter slope has a 12 dB/Octave roll-off.

The darker curve illustrates the roll-off curve of the Highpass Filter. The lighter curve in the middle represents the filter characteristic
when the Resonance parameter is raised.
• Notch
A notch filter (or band reject filter) could be described as the opposite of a bandpass filter. It cuts off frequencies
in a narrow midrange band, letting the frequencies below and above through. On its own, a notch filter doesn’t re-
ally alter the timbre in any dramatic way, simply because most frequencies are let through. However, by combining
a notch filter with a lowpass filter (using Filter 2 - see “Filter 2”), more musically useful filter characteristics can be
created. Such a filter combination can produce soft timbres that still sound “clear”. The effect is especially notice-
able with low resonance (see “Resonance”) settings.

The darker curve illustrates the roll-off curve of the Notch Filter. The lighter curve in the middle represents the filter characteristic
when the Resonance parameter is raised.

515 SUBTRACTOR SYNTHESIZER

 Filter 1 Frequency
The Filter Frequency parameter (often referred to as “cutoff”) determines which area of the frequency spectrum the
filter will operate in. For a lowpass filter, the frequency parameter could be described as governing the “opening” and
“closing” of the filter. If the Filter Freq is set to zero, none or only the very lowest frequencies are heard, if set to max-
imum, all frequencies in the waveform are heard. Gradually changing the Filter Frequency produces the classic syn-
thesizer filter “sweep” sound.
! Note that the Filter Frequency parameter is usually controlled by the Filter Envelope (see “Filter Envelope”) as
well. Changing the Filter Frequency with the Freq slider may therefore not produce the expected result.

Resonance
The filter resonance parameter is used to set the Filter characteristic, or quality. For lowpass filters, raising the filter
Res value will emphasize the frequencies around the set filter frequency. This produces a generally thinner sound, but
with a sharper, more pronounced filter frequency “sweep”. The higher the filter Res value, the more resonant the
sound becomes until it produces a whistling or ringing sound. If you set a high value for the Res parameter and then
vary the filter frequency, this will produce a very distinct sweep, with the ringing sound being very evident at certain
frequencies.
• For the highpass filter, the Res parameter operates just like for the lowpass filters.
• When you use the Bandpass or Notch filter, the Resonance setting adjusts the width of the band. When you
raise the Resonance, the band where frequencies are let through (Bandpass), or cut (Notch) will become nar-
rower. Generally, the Notch filter produces more musical results using low resonance settings.

Filter Keyboard Track (Kbd)

If Filter Keyboard Track is activated, the filter frequency will increase the further up on the keyboard you play. If a low-
pass filter frequency is constant (a Kbd setting of “0”) this can introduce a certain loss of “sparkle” in a sound the
higher up the keyboard you play, because the harmonics in the sound are progressively being cut. By using a degree
of Filter Keyboard Tracking, this can be compensated for.

516 SUBTRACTOR SYNTHESIZER

 Filter 2
A very useful and unusual feature of the Subtractor Synthesizer is the presence of an additional 12dB/Oct lowpass
filter. Using two filters together can produce many interesting filter characteristics, that would be impossible to create
using a single filter, for example formant effects.
The parameters are identical to Filter 1, except in that the filter type is fixed, and it does not have filter keyboard
tracking.
D To activate Filter 2, click the button at the top of the Filter 2 section.
Filter 1 and Filter 2 are connected in series. This means that the output of Filter 1 is routed to Filter 2, but both fil-
ters function independently. For example, if Filter 1 was filtering out most of the frequencies, this would leave Filter
2 very little to “work with”. Similarly, if Filter 2 had a filter frequency setting of “0”, all frequencies would be filtered
out regardless of the settings of Filter 1.

q Try the “Singing Synth” patch (in the Monosynth category of the Factory Sound Bank) for an example of how
dual filters can be used.

Filter Link

When Link (and Filter 2) is activated, the Filter 1 frequency controls the frequency offset of Filter 2. That is, if you
have set different filter frequency values for Filter 1 and 2, changing the Filter 1 frequency will also change the fre-
quency for Filter 2, but keeping the relative offset.
q Try the “Fozzy Fonk” patch (in the Polysynth category of the Factory Sound Bank) for an example how linked
filters can be used.
! Caution! If no filter modulation is used, and the filters are linked, pulling down the frequency of Filter 2 to zero
will cause both filters to be set to the same frequency. If combined with high Res settings, this can produce
very loud volume levels that cause distortion!

517 SUBTRACTOR SYNTHESIZER

 Envelopes - General
Envelope generators are used to control several important sound parameters in analog synthesizers, such as pitch,
volume, filter frequency etc. Envelopes govern how these parameters should respond over time - from the moment a
note is struck to the moment it is released.
Standard synthesizer envelope generators have four parameters; Attack, Decay, Sustain and Release (ADSR).
There are three envelope generators in the Subtractor, one for volume, one for the Filter 1 frequency, and one mod-
ulation envelope which has selectable modulation destinations.
Level

Sustain
(level)

Time
Attack Decay Release
(time) (time) (time)

Key Down Key Up

The ADSR envelope parameters.

Attack
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the maxi-
mum value. How long this should take, depends on the Attack setting. If the Attack is set to “0”, the maximum value
is reached instantly. If this value is raised, it will take time before the maximum value is reached.
For example, if the Attack value is raised and the envelope is controlling the filter frequency, the filter frequency will
gradually rise up to a point each time a key is pressed, like an “auto-wha” effect.

Decay
After the maximum value has been reached, the value starts to drop. How long this should take is governed by the
Decay parameter.
If you wanted to emulate the volume envelope of a note played on a piano for example, the Attack should be set to
“0” and the Decay parameter should be set to a medium value, so that the volume gradually decreases down to si-
lence, even if you keep holding the key down. Should you want the decay to drop to some other value than zero, you
use the Sustain parameter.

Sustain
The Sustain parameter determines the level the envelope should rest at, after the Decay. If you set Sustain to full
level, the Decay setting is of no importance since the volume of the sound is never lowered.

518 SUBTRACTOR SYNTHESIZER

 If you wanted to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain pa-
rameter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays
there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the maximum value, then
gradually decreases to finally land to rest on a level somewhere in-between zero and maximum. Note that Sustain
represents a level, whereas the other envelope parameters represent times.

Release
Finally, we have the Release parameter. This works just like the Decay parameter, except it determines the time it
takes for the value to fall back to zero after releasing the key.

Amplitude Envelope

The Amplitude Envelope is used to adjust how the volume of the sound should change from the time you press a key
until the key is released. By setting up a volume envelope you sculpt the sound’s basic shape with the four Amplitude
Envelope parameters, Attack, Decay, Sustain and Release. This determines the basic character of the sound (soft,
long, short etc.).

Filter Envelope

The Filter Envelope affects the Filter 1 Frequency parameter. By setting up a filter envelope you control the how the
filter frequency should change over time with the four Filter Envelope parameters, Attack, Decay, Sustain and Re-
lease.

Filter Envelope Amount

This parameter determines to what degree the filter will be affected by the Filter Envelope. Raising this knob’s value
creates more drastic results. The Envelope Amount parameter and the set Filter Frequency are related. If the Filter
Freq slider is set to around the middle, this means that the moment you press a key the filter is already halfway open.
The set Filter Envelope will then open the filter further from this point. The Filter Envelope Amount setting affects
how much further the filter will open.

519 SUBTRACTOR SYNTHESIZER

 Filter Envelope Invert

If this button is activated, the envelope will be inverted. For example, normally the Decay parameter lowers the filter
frequency, but after activating Invert it will instead raise it, by the same amount.

Mod Envelope

The Mod Envelope allows you to select one of a number of parameters, or Destinations, to control with the envelope.
By setting up a modulation envelope you control the how the selected Destination parameter should change over
time with the four Mod Envelope parameters, Attack, Decay, Sustain and Release.
The available Mod Envelope Destinations are as follows:

| Destination | Description
Osc 1 Selecting this makes the Mod Envelope control the pitch (frequency) of Osc 1.
Osc 2 Same as above, but for Osc 2.
Osc Mix Selecting this makes the Mod Envelope control the oscillator Mix parameter. Both oscillators must be acti-
vated for this to have any effect.
FM Selecting this makes the Mod Envelope control the FM Amount parameter. Both oscillators must be acti-
vated for this to have any effect.
Phase Selecting this makes the Mod Envelope control the Phase Offset parameter for both Osc 1 and 2. Note
that Phase Offset Modulation (Subtraction or Multiplication) must be activated for this to have any effect.
Freq 2 Selecting this makes the Mod Envelope control the Frequency parameter for Filter 2.

520 SUBTRACTOR SYNTHESIZER

 LFO Section

LFO stands for Low Frequency Oscillator. LFO’s are oscillators, just like Osc 1 & 2, in that they also generate a wave-
form and a frequency. However, there are two significant differences:
• LFOs only generate waveforms with low frequencies.
• The output of the two LFO’s are never actually heard. Instead they are used for modulating various parameters.
The most typical application of an LFO is to modulate the pitch of a (sound generating) oscillator, to produce vibrato.
Subtractor is equipped with two LFO’s. The parameters and the possible modulation destinations vary somewhat be-
tween LFO 1 and LFO 2.

LFO 1 Parameters
Waveform
LFO 1 allows you to select different waveforms for modulating parameters. These are (from top to bottom):

| Waveform | Description
Triangle This is a smooth waveform, suitable for normal vibrato.
Inverted Sawtooth This produces a “ramp up” cycle. If applied to an oscillator’s frequency, the pitch would sweep up to a set point
(governed by the Amount setting), after which the cycle immediately starts over.
Sawtooth This produces a “ramp down” cycle, the same as above but inverted.
Square This produces cycles that abruptly changes between two values, usable for trills etc.
Random Produces random stepped modulation to the destination. On some vintage synths, this is called “sample & hold”.
Soft Random The same as above, but with smooth modulation.

Destination
The available LFO 1 Destinations are as follows:

| Destination | Description
Osc 1&2 Selecting this makes LFO 1 control the pitch (frequency) of Osc 1 and Osc 2.
Osc 2 Same as above, but for Osc 2.
Filter Freq Selecting this makes the LFO 1 control the filter frequency for Filter 1 (and Filter 2 if linked).
FM Selecting this makes the LFO 1 control the FM Amount parameter. Both oscillators must be activated for this to
have any effect.
Phase Selecting this makes the LFO 1 control the Phase Offset parameter for both Osc 1 and 2. Note that Phase Off-
set Modulation (Subtraction or Multiplication) must be activated for this to have any effect.
Osc Mix Selecting this makes the LFO 1 control the oscillator Mix parameter.

521 SUBTRACTOR SYNTHESIZER

 Sync
By clicking this button you activate/deactivate LFO sync. The frequency of the LFO will then be synchronized to the
song tempo, in one of 16 possible time divisions. When sync is activated, the Rate knob (see below) is used for set-
ting the desired time division.
Turn the knob and check the tooltip for an indication of the time division.

Rate
The Rate knob controls the LFO’s frequency. Turn clockwise for a faster modulation rate.

Amount
This parameter determines to what degree the selected parameter destination will be affected by LFO 1. Raising this
knob’s value creates more drastic results.

LFO 2 Parameters
LFO 2 is polyphonic. This means that for every note you play, an independent LFO cycle is generated, whereas LFO
1 always modulates the destination parameter using the same “cycle”. This can be used to produce subtle cross-
modulation effects, with several LFO cycles that “beat” against each other. This also enables LFO 2 to produce mod-
ulation rates that vary across the keyboard (see the “Keyboard Tracking” parameter below).

Destination
The available LFO 2 Destinations are as follows:

| Destination | Description
Osc 1&2 Selecting this makes LFO 2 modulate the pitch (frequency) of Osc 1 and Osc 2.
Phase Selecting this makes the LFO 2 modulate the Phase Offset parameter for both Osc 1 and 2. Note that
Phase Offset Modulation (Subtraction or Multiplication) must be activated for this to have any effect.
Filter Freq 2 Selecting this makes the LFO 2 modulate the filter frequency for Filter 2.
Amp Selecting this makes the LFO 2 modulate the overall volume., to create tremolo-effects.

LFO 2 Delay
This parameter is used to set a delay between when a note is played and when the LFO modulation “kicks in”. For ex-
ample, if Osc 1 & 2 is selected as the destination parameter and Delay was set to a moderate value, the sound would
start out unmodulated, with the vibrato only setting in if you hold the note(s) long enough. Delayed LFO modulation
can be very useful, especially if you are playing musical instrument-like sounds like violin or flute. Naturally it could
also be used to control more extreme modulation effects and still retain the “playability” of the sound.

LFO 2 Keyboard Tracking

If LFO keyboard tracking is activated, the LFO rate will progressively increase the higher up on the keyboard you play.
Raising this value creates more drastic results.

522 SUBTRACTOR SYNTHESIZER

 q If the LFO is set to modulate the phase offset, LFO keyboard tracking can produce good results. For example,
synth string pads and other sounds that use PWM (see “Phase Offset Modulation”) can benefit from this.

Rate
The Rate knob controls the LFO’s frequency. Turn clockwise for a faster modulation rate.

Amount
This parameter determines to what degree the selected parameter destination will be affected by LFO 2. Raising this
knob’s value creates more drastic results.

Play Parameters
This section deals with two things: Parameters that are affected by how you play, and modulation that can be applied
manually with standard MIDI keyboard controls.
These are:
• Velocity Control
• Pitch Bend and Modulation Wheel
• Legato
• Portamento
• Polyphony

Velocity Control

Velocity is used to control various parameters according to how hard or soft you play notes on your keyboard. A com-
mon application of velocity is to make sounds brighter and louder if you strike the key harder. Subtractor features
very comprehensive velocity modulation capabilities. By using the knobs in this section, you can control how much the
various parameters will be affected by velocity. The velocity sensitivity amount can be set to either positive or negative
values, with the center position representing no velocity control.

523 SUBTRACTOR SYNTHESIZER

 The following parameters can be velocity controlled:

| Destination | Description
Amp This let’s you velocity control the overall volume of the sound. If a positive value is set, the volume will in-
crease the harder you strike a key. A negative value inverts this relationship, so that the volume decreases
if you play harder, and increases if you play softer. If set to zero, the sound will play at a constant volume,
regardless of how hard or soft you play.
FM This sets velocity control for the FM Amount parameter. A positive value will increase the FM amount the
harder you play. Negative values invert this relationship.
M. Env This sets velocity control for the Mod Envelope Amount parameter. A positive value will increase the enve-
lope amount the harder you play. Negative values invert this relationship.
Phase This sets velocity control for the Phase Offset parameter. This applies to both Osc 1 & 2, but the relative
offset values are retained. A positive value will increase the phase offset the harder you play. Negative val-
ues invert this relationship.
Freq 2 This sets velocity control for the Filter 2 Frequency parameter. A positive value will increase the filter fre-
quency the harder you play. Negative values invert this relationship.
F. Env This sets velocity control for the Filter Envelope Amount parameter. A positive value will increase the en-
velope amount the harder you play. Negative values invert this relationship.
F. Dec This sets velocity control for the Filter Envelope Decay parameter. A positive value will increase the Decay
time the harder you play. Negative values invert this relationship.
Osc Mix This sets velocity control for the Osc Mix parameter. A positive value will increase the Osc 2 Mix amount
the harder you play. Negative values invert this relationship.
A. Attack This sets velocity control for the Amp Envelope Attack parameter. A positive value will increase the Attack
time the harder you play. Negative values invert this relationship.

Pitch Bend and Modulation Wheels

The Pitch Bend wheel is used for “bending” notes, like bending the strings on a guitar. The Modulation wheel can be
used to apply various modulation while you are playing. Virtually all MIDI keyboards have Pitch Bend and Modulation
controls. Subtractor features not only the settings for how incoming MIDI Pitch Bend and Modulation wheel mes-
sages should affect the sound. Subtractor also has two functional wheels that could be used to apply real time mod-
ulation and pitch bend should you not have these controllers on your keyboard, or if you aren’t using a keyboard at all.
The Subtractor wheels mirror the movements of the MIDI keyboard controllers.

Pitch Bend Range

The Range parameter sets the amount of pitch bend when the wheel is turned fully up or down. The maximum range
is “24” (=up/down 2 Octaves).

524 SUBTRACTOR SYNTHESIZER

 Modulation Wheel
The Modulation wheel can be set to simultaneously control a number of parameters. You can set positive or negative
values, just like in the Velocity Control section. The following parameters can be affected by the modulation wheel:

| Parameter | Description
F. Freq This sets modulation wheel control of the Filter 1 Frequency parameter. A positive value will increase the
frequency if the wheel is pushed forward. Negative values invert this relationship.
F. Res This sets modulation wheel control of the Filter 1 Resonance parameter. A positive value will increase the
resonance if the wheel is pushed forward. Negative values invert this relationship.
LFO 1 This sets modulation wheel control of the LFO 1 Amount parameter. A positive value will increase the
Amount if the wheel is pushed forward. Negative values invert this relationship.
Phase This sets modulation wheel control of the Phase Offset parameter for both Osc 1 and 2. Note that Phase
Offset Modulation (Subtraction or Multiplication) must be activated for this to have any effect.
FM This sets modulation wheel control of the FM Amount parameter. A positive value will increase the FM
amount if the wheel is pushed forward. Negative values invert this relationship. Both oscillators must be
activated for this to have any effect.

Legato
Legato works best with monophonic sounds. Set Polyphony (see below) to 1 and try the following:
D Hold down a key and press another key without releasing the previous.
Notice that the pitch changes, but the envelopes do not start over. That is, there will be no new “attack”.
D If polyphony is set to more voices than 1, Legato will only be applied when all the assigned voices are “used
up”.
For example, if you had a polyphony setting of “4” and you held down a 4 note chord, the next note you played
would be Legato. Note, however, that this Legato voice will “steal” one of the voices in the 4 note chord, since all
the assigned voices were already used up!

Retrig
This is the “normal” setting for playing polyphonic patches. That is, when you press a key without releasing the previ-
ous, the envelopes are retriggered, like when you release all keys and then press a new one. In monophonic mode,
Retrig has an additional function; if you press a key, hold it, press a new key and then release that, the first note is
also retriggered.

Portamento (Time)
Portamento is when the pitch “glides” between the notes you play, instead of instantly changing the pitch. The Porta-
mento knob is used to set how long it takes for the pitch to glide from one pitch to the next. If you don’t want any Por-
tamento at all, set this knob to zero.

525 SUBTRACTOR SYNTHESIZER

 Setting Number of Voices - Polyphony

This determines the polyphony, i.e. the number of voices a Subtractor Patch can play simultaneously. This can be
used to make a patch monophonic (=a setting of “1”), or to extend the number of voices available for a patch. The
maximum number of voices you can set a Subtractor Patch to use is 99. In the (unlikely) event you should need more
voices, you can always create another Subtractor!
! Note that the Polyphony setting does not “hog” voices. For example, if you have a patch that has a polyphony
setting of ten voices, but the part the patch plays only uses four voices, this won’t mean that you are “wasting”
six voices. In other words, the polyphony setting is not something you need to consider much if you want to
conserve CPU power - it is the number of voices actually used that counts.

About the Low Bandwidth button

This can be used to conserve CPU power. When activated, this function will remove some high frequency content
from the sound of this particular device, but often this is not noticeable (this is especially true for bass sounds).

External Modulation

Subtractor can receive common MIDI controller messages, and route these to various parameters. The following MIDI
messages can be received:
• Aftertouch (Channel Pressure)
• Expression Pedal
• Breath Control
If your MIDI keyboard is capable of sending Aftertouch messages, or if you have access to an Expression Pedal or a
Breath controller, you can use these to modulate parameters. The “Ext. Mod” selector switch sets which of these
message-types should be received.
These messages can then be assigned to control the following parameters:

| Destination | Description
F. Freq This sets External modulation control of the Filter 1 Frequency parameter. A positive value will increase
the frequency with higher external modulation values. Negative values invert this relationship.
LFO 1 This sets External modulation control of the LFO 1 Amount parameter. A positive value will increase the
LFO 1 amount with higher external modulation values. Negative values invert this relationship.

526 SUBTRACTOR SYNTHESIZER

 | Destination | Description
Amp This let’s you control the overall volume of the sound with external modulation. If a positive value is set,
the volume will increase with higher external modulation values. A negative value inverts this relation-
ship.
FM This sets External modulation control of the FM Amount parameter. If a positive value is set, the FM
amount will increase with higher external modulation values. A negative value inverts this relationship.
Both oscillators must be activated for this to have any effect.

Connections

Flipping the Subtractor around reveals a plethora of connection possibilities, most of which are CV/Gate related. Us-
ing CV/Gate is described in the chapter “Routing Audio and CV”.

Audio Output
This is Subtractor’s main audio output. When you create a new Subtractor device, this is auto-routed to the first avail-
able channel on the audio mixer.

Sequencer Control
The Sequencer Control CV and Gate inputs allow you to play the Subtractor from another CV/Gate device (typically
a Matrix or a Redrum). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers
note on/off along with velocity.
! For best results, you should use the Sequencer Control inputs with monophonic sounds.

527 SUBTRACTOR SYNTHESIZER

 Modulation Inputs
! Remember that CV connections will not be stored in the Subtractor patch, even if the connections are to/from
the same Subtractor device!
These control voltage (CV) inputs (with associated voltage trim pots), can modulate various Subtractor parameters
from other devices, or from the modulation outputs of the same Subtractor device. These inputs can control the fol-
lowing parameters:
• Oscillator Pitch (both Osc 1 & 2).
• Oscillator Phase Offset (both Osc 1 & 2).
• FM Amount
• Filter 1 Cutoff
• Filter 1 Res
• Filter 2 Cutoff
• Amp Level
• Mod Wheel

Modulation Outputs
The Modulation outputs can be used to voltage control other devices, or other parameters in the same Subtractor de-
vice. The Modulation Outputs are:
• Mod Envelope
• Filter Envelope
• LFO 1

Gate Inputs
These inputs can receive a CV signal to trigger the following envelopes. Note that connecting to these inputs will
override the normal triggering of the envelopes. For example, if you connected an LFO output to the Gate Amp input,
you would not trigger the amp envelope by playing notes, as this is now controlled by the LFO. In addition you would
only hear the LFO triggering the envelope for the notes that you hold down. The following Gate Inputs can be se-
lected:
• Amp Envelope
• Filter Envelope
• Mod Envelope

528 SUBTRACTOR SYNTHESIZER

Chapter 29
Thor Polysonic
Synthesizer
 Introduction

Thor is an advanced synthesizer with many unique features.

The design could be described as semi-modular, in that the oscillator and filter sections are open slots that allow the
user to select between various different oscillator and filter types, each with a distinct character. Some of these de-
signs were inspired by selected vintage equipment.
As a result, Thor is capable of producing an astounding array of sounds.
While it offers a lot of scope for serious sound modeling, it still has a basically simple and user-friendly interface.
In the extensive Modulation bus routing section both audio and control signals (CV) co-exist, and more or less any
routing combination can be assigned. Use audio to modulate a CV signal or vice versa - Thor’s modulation capabilities
are virtually limitless.
Thor also features an advanced step sequencer which can be used for creating melody lines or purely as a modula-
tion source.
There are also audio inputs on the back panel. By connecting the output of another device to these inputs, you can
use Thor’s filters, envelopes etc. to process the sound, or you can use the external audio source to modulate a Thor
parameter.
! Please, note that this device is not available in Reason Lite Rack Plugin.

About basic synthesizer terminology

This chapter assumes familiarity with common synth terminology like oscillators, waveforms, filters and envelopes. If
you are new to Reason (or these terms), you may want to read the “Subtractor Synthesizer” chapter first, where these
elements and how they interact are described from a more basic point of view.

Loading and Saving Patches

Loading and saving patches is done in the same way as with any other Reason device - see “Loading patches” and
“Saving patches”.

530 THOR POLYSONIC SYNTHESIZER

 Thor elements
In the picture below an unfolded Thor device is shown.

Thor’s user interface consists of the following elements (from the top down):
• The Controller panel, which is always shown if Thor is unfolded.
See “The Controller panel”.
• The main Programmer panel contains all the synth parameters.
The Programmer can be shown/hidden by clicking the “Show Programmer” button on the Controller panel. See
“Using the Programmer”.
• The Modulation bus routing section.
See “Modulation bus routing section”.
• The Step Sequencer section, where you can program up to 16 steps to produce short melody lines/grooves or
use it as a modulation source.
See “Step Sequencer”.

531 THOR POLYSONIC SYNTHESIZER

 The Controller panel

The Controller panel contains standard Master Volume and Pitch and Mod controls, Keyboard Mode/Note Triggering
sections and four virtual (freely assignable) controls. The panel also has a patch display and standard Select/
Browse/Save patch buttons (these are always shown even if Thor is folded).

The Keyboard Mode section

In this section you make basic keyboard related settings for a patch. It has the following options:

| Function | Description
Polyphony This setting determines the number of voices that you can play simultaneously when Polyphonic mode
is selected. The maximum number of voices is 32.
Release This governs the number of voices that are allowed to naturally decay/ring out (in the release phase of
Polyphony the envelope) when new notes are triggered and Polyphonic mode is selected. E.g. if you set this to “0”,
any new note(s) will cut off the release of any previously triggered notes.
Mono Legato Mono Legato mode is monophonic regardless of the Polyphony setting. It works as follows:
D Hold down a key and then press another key without releasing the previous.
Notice that the pitch changes, but the envelopes do not start over. That is, there
will be no new “attack”.
Mono Retrig Mono Retrig is also monophonic and this mode means that when you press a key the envelopes are al-
ways retriggered.
Polyphonic This is the standard polyphonic play mode - you can play the number of voices set with the Polyphony
parameter.
Portamento On/Off/Auto The knob is used for controlling portamento - a parameter that makes the pitch glide between the
notes you play, rather than changing the pitch instantly as soon as you hit a key on your keyboard. By
turning this knob you set how long it should take for the pitch to glide from one note to the next as you
play them. There are three basic portamento modes:
• In Auto mode, there will only be any portamento when playing more than one
note. If any of the Mono modes is selected, portamento will only affect the
legato notes.
• When set to On, portamento is applied to all notes.
• Off means no portamento.

Note Triggering section

Using the buttons in this section you can select in what way Thor will respond:
• Via note input only.
• Via the Step Sequencer only (see “Step Sequencer”).
• Or both.
The section also has a standard Note On indicator.

532 THOR POLYSONIC SYNTHESIZER

 About the assignable controls
• The rotary knobs and buttons in the Controller panel are assignable controls that can be assigned to multiple
parameters and functions in Thor.
• You assign parameters to the knobs and buttons in the Modulation Routing panel (these are located on the
“Modifiers” sub-menu - see “Modulation bus routing section”).
• Movements of the assignable controls can be recorded as automation.
• Each control can be assigned to any number of parameters.
• Clicking on the label for a Rotary or Button lets you type in an appropriate name for it.
About the button key note function
To the right of the two assignable buttons there are corresponding spin controls and displays. These can be used to
assign a key for turning the button on momentarily, as long as the key is held down.
D Use the spin controls (or click in the display and move the mouse up or down) to assign a key for the button
status.
The assigned key will now turn the function(s) assigned to the button on for as long the key is held down.
D Note that the key note function can only switch from off to on, not the other way around, so make sure the but-
ton is deactivated if you wish to use this function.
An assigned key will not trigger a note, only the button status. Also note that the button will not light up when you
press the assigned key.

The Pitch Bend and Modulation wheels

• The Pitch and Mod wheels on the Controller panel will mirror the corresponding actions on your master key-
board.
• The Range parameter (like for all instrument devices) sets the range of the Pitch Bend action.
• Pitch Bend is pre-wired to the pitch parameter of the three oscillators, but you can of course use it to control
any parameter you like. If you don’t want Pitch Bend to affect oscillator pitch, simply set the Range parameter
to “0”.

Master volume
This is the main volume control for outputs 1 & 2.

533 THOR POLYSONIC SYNTHESIZER

 Using the Programmer
The Programmer contains the main synth parameters.
D To show the Programmer panel, click the “Show Programmer” button on the Controller panel.
The Programmer appears below the Controller panel.

D The Programmer panel is divided into two sections; the Voice section to the left and the Global section to the
right. The Global section has a separate brown panel to differentiate it from the Voice section.
The Voice section contains the basic synth parameters and the parameters are “per-voice”, i.e. all envelope and
LFO cycles are triggered individually for each voice. The Global section to the right contains global parameters that
affect all voices.
D There are three open Oscillator slots, a Mixer, two open Filter slots, a Shaper, three Envelope generators, an
LFO and an Amplifier in the Voice section.
The open Oscillator and Filter slots allow you to select between different types of oscillators and filters.
D The Global section contains a second LFO, a Global Envelope, a third open Filter slot and Chorus and Delay ef-
fects.

534 THOR POLYSONIC SYNTHESIZER

 Basic connections - a tutorial
There are certain pre-defined connections available between the Oscillator 1-3 slots and the Mixer, Filter 1/Shaper,
Filter 2 and Amp sections. On the panel itself, lines with arrows are shown to indicate the standard signal paths.
q Note that you can also connect sections using the Modulation bus section (see “Modulation bus routing sec-
tion”). You are not in any way limited to the pre-defined routings, but they do provide a quick and convenient
way to connect the basic synth “building blocks” together.
In the following tutorial we will create a standard setup using two oscillators and two filters to demonstrate Thor ba-
sics and the (standard) signal path:
1. Select “Reset Device” from the context menu or from the Edit menu.
The “Init patch” is a basic setup with an Analog oscillator in Oscillator slot 1 and a Ladder LP filter in Filter slot 1
loaded. A connection between Oscillator 1, Filter 1 and the Amp section is already activated, so you get a sound
when you play.
Below the Oscillator 1 slot in the upper left corner are two more slots, currently empty. These are the Oscillator 2 and
3 slots, respectively. The three Oscillator slots are basically identical in that they can each be loaded with one of 6 os-
cillator types.
2. Click the arrow pop-up in the upper left corner of the Oscillator 2 slot, and select a second oscillator from the
pop-up that appears.
The following oscillator types are available; Analog, Wavetable, Phase Modulation, FM Pair, Multi and Noise. For a
description of the various oscillator types, see “The Oscillator section”.

Selecting oscillator type.

535 THOR POLYSONIC SYNTHESIZER

 With a basic connection setup, the Oscillator outputs are internally connected to the “Mix” section. To pass the output
signal onwards in the signal chain, you first have to activate a connection. This is done using the two vertical rows of
routing buttons labelled 1, 2 and 3 to the right of the Oscillator section.

• The upper row of routing buttons determine which of the Oscillators 1 to 3 are routed to Filter 1, and the lower
row which of the Oscillators 1 to 3 are routed to Filter 2.
All three oscillators can be simultaneously routed to both filters, serially or in parallel (or any combination of these
variations). This is explained later in this tutorial.
By activating one or more of these buttons means that the oscillator (1 to 3) is routed to the corresponding Filter.
Currently, Oscillator 1 is connected to Filter 1 slot (which is pre-loaded with a Ladder LP filter).
This is indicated by the “1” routing button being lit. The Filter 2 slot is currently not active, which is indicated by a
blank panel.
3. Click the “2” button to the left of the Filter 1 section so that it lights up to activate a connection for Oscillator 2.
Now if you play a few notes you should hear both Oscillator 1 and Oscillator 2, via the Filter 1 section.

• The Filter 1 output passes via the Shaper (currently not activated), on to the Amp section, and finally to the
Main Outputs.
Actually, the Amp section output is routed via the Global section before being sent to the Main Outputs, but as cur-
rently nothing is activated in the Global section the signal passes through unprocessed.

536 THOR POLYSONIC SYNTHESIZER

 4. Next, click the arrow pop-up in the upper left corner of the Filter 2 slot.
A pop-up menu with the four available Filter types appears. For a description of the filter types, see “Filter slots”.

5. Select a type of filter, e.g. a Comb filter for the Filter 2 slot.
Now that the Filter 2 slot in the Voice section is active, you can connect the oscillators to it by using the lower row
of routing buttons.
6. Click the routing buttons “1” and “2” to the left of the Filter 2 slot so that the buttons are lit.
Now the two oscillators are connected to Filter 2.
7. Make sure the arrow routing button that points to the Amp section just above the Filter 2 section is activated.
Now if you play a few notes, both oscillators are routed via both filter sections in parallel. You could of course se-
lect to pass only one of the oscillators via one filter and both oscillators via the other - any combination is possible.

You can also connect the Filter 1 and 2 sections serially, meaning that the output of Filter 1 is passed through Filter
2 before reaching the Amp section. This is done as follows:
8. Switch off the routing buttons “1” and “2” to the left of the Filter 2 slot.
If you leave them on the oscillators will pass through Filter 2 twice; both via Filter 1 and directly. This is also per-
fectly “allowable”, but to make things clearer in this tutorial we will use a standard serial filter setup.
9. Click the Arrow “left” button below the Shaper.
Now the filters are connected serially, with the output of Filter 1 (via the for now inactive Shaper) being connected
to the Filter 2 input. Both oscillators are processed by both filters connected in series.

That concludes this tutorial on how the pre-wired connections in the Voice section can be used, but note that you can
also use the Modulation bus to make connections - see “Modulation bus routing section”.

537 THOR POLYSONIC SYNTHESIZER

 Other pre-defined routing assignments
There are other sections in Thor which are pre-defined and can be used without having to make any prior assign-
ments:
D The Amp Envelope and the Filter Envelope control the volume level and frequency of the Filters (1 & 2), respec-
tively.
The amount of filter envelope control is controllable by using the “Env” parameter in each Filter section.
D The effects (Delay/Chorus) in the Global section are part of the signal chain and can simply be switched on and
used.

The Oscillator section

Oscillators generate the basic raw sound (pitch and waveform) that can in turn be processed by the other parame-
ters. The Oscillator section contains three open slots which can each be loaded with one of six oscillator types. The
three Oscillator slots are numbered 1-3, with the top slot housing Oscillator 1, the middle slot Oscillator 2 and the
bottom slot Oscillator 3.
D The Arrow button in the top left corner of each slot opens a pop-up menu where an oscillator type can be se-
lected for the corresponding slot.

There are six Oscillator types available:

• Analog
• Wavetable

538 THOR POLYSONIC SYNTHESIZER

 • Phase Modulation
• FM Pair
• Multi Oscillator
• Noise
You can also select Off mode (no oscillator).

Common parameters
The specific parameters of the various oscillator types are described separately, but there are also common parame-
ters that apply to all oscillator types. These are:
D Octave (OCT) knob - this changes the pitch of the oscillator in octave steps.
The range is ten octaves.
D The Semi knob changes the pitch of the oscillator in semi-tone steps.
The range is 12 semitone steps (1 octave).
D The Tune knob fine tunes the pitch of the oscillator in cent steps.
The range is +/- 50 cents (down or up half a semitone).
D Keyboard Track (KBD) - this knob sets how much the oscillator pitch tracks incoming note data.
Turned fully clockwise the pitch tracks the keyboard normally, i.e. a semitone per key.
D All oscillators also have waveform selectors and a modifier parameter. How the waveform selection works,
and what parameter is the modifier varies according to the selected oscillator type.

D Important to note is that if you have made a modulation routing to an oscillator parameter e.g. the modifier,
and then change the oscillator type, the modulation will be transferred to the corresponding parameter in the
new oscillator.
The same goes for all common parameters (tuning and tracking). If you switch oscillator type, all common param-
eters are left unchanged.
D Oscillators can be synced - see “About Oscillator Sync”.

D Any oscillator type loaded into the Oscillator 1 slot can also be amplitude modulated by Oscillator 2 - see
“About Amplitude Modulation (AM)”.

Analog oscillator

This is a classic analog oscillator with 4 standard waveforms. The waveform selector button is in the lower left corner
of the oscillator panel, but you can also click directly on the waveform symbols to switch waveform. The four available
waveforms are from the top down (as displayed on the panel): Sawtooth, Pulse, Triangle and Sine.
• The Mod parameter (PW) controls pulse width and only affects the pulse waveform.
By modulating the PW parameter the width of the pulse wave changes, allowing for PWM (Pulse Width Modula-
tion) which is a standard feature in most vintage analog synths.
q For a perfect square wave, set pulse width (PW) to 64.

539 THOR POLYSONIC SYNTHESIZER

 Wavetable oscillator

Wavetable oscillators has been the basis of several vintage synths (PPG, Korg Wavestation and many others).
• With the Wavetable oscillator, you select between 32 wavetables, where each wavetable contains several (up
to 64) different waveforms. By using an envelope or a LFO you can sweep through a wavetable to produce tim-
bre variations.
The parameters are as follows:
D Position is the modifier (Mod) parameter and controls the position within the selected wavetable, i.e. which
waveform is active at a given time.
By modulating the Position you can sweep through the waveforms in the selected wavetable. You can of course
also use a single static waveform in a wavetable if you so wish, by not applying any modulation to this parameter.
D The X-Fade button determines whether the change between waveforms in a wavetable should be abrupt (X-
Fade off), or smooth (X-Fade on).
If set to on, the waveform transitions are cross-faded.
D There are 32 wavetables that can be selected using the up/down buttons or by clicking in the Wavetable dis-
play.
Some of the wavetables have waveforms that sequentially follow the harmonic series, i.e. each following waveform
adds a harmonic. Others have waveform series that produce a sound similar to oscillator sync when swept, and other
wavetables are simply mixed waveforms. The last 11 wavetables are based on wavetables used in the original PPG
2.3 synthesizer.

Phase Modulation oscillator

The Phase Modulation oscillator is inspired by the Casio CZ series of synthesizers. Phase modulation is based on
modulating the phase of digital waveforms to emulate common filter characteristics.
D You have a First and Second waveform which can be combined. Instead of mixing the two waveforms they are
played in series, one after the other.
This adds a fundamental one octave below the pitch of the original sound.
D The PD parameter (Mod) changes the shape of the wave, much like a filter does.
The following waveforms (sequentially from the first) are available as the First waveform:
• Sawtooth
• Square

540 THOR POLYSONIC SYNTHESIZER

 • Pulse
• Pulse and Sine
• Sine and flat (half sine)
• Saw x Sine
• Sine x Sine
• Sine x Pulse
The last three waveforms could be described “resonant”, as these originally were meant to simulate filter resonance.
They didn’t really do this very accurately, but nevertheless constituted an important part of the sound.
The Second waveform has the same available waveforms except the last three, and it can also be bypassed alto-
gether. You can combine waveforms freely, except it is not possible to combine two “resonant” waveforms.

FM Pair oscillator

As the name implies, this oscillator generates FM, where one oscillator (Carrier) is frequency modulated by a second
oscillator (Modulator). Although very simple to use (unlike most hardware FM synths), this oscillator can produce a
very wide range of FM sounds.
D The Carrier and Modulator selector buttons set the frequency ratio between these two oscillators (the range is
1-32).
The frequency ratio is what determines the basic frequency content, and thus, the timbre of the sound.
D The FM knob sets the amount of frequency modulation.
This is also the Modifier parameter. If FM amount is set to zero, there is no FM and the output will be a pure sine
wave.
• If you set FM Amount to zero and step through the values of the Carrier oscillator, you can hear that the pitch
is changed according to the harmonic series.
• Stepping through the Mod oscillator values will change the pitch in the same way, although FM Amount has to
be set to a value other than zero to be able to hear it.
Thus, 2:2 is the same wave shape as 1:1 but one octave higher in pitch, 3:3 is the same wave shape as 2:2 but a fifth
higher in pitch and so on.

541 THOR POLYSONIC SYNTHESIZER

 Multi oscillator

This versatile oscillator can simultaneously generate multiple detuned waveforms (of a set type) per voice. It is great
for producing complex timbres e.g. to simulate cymbal or bell sounds, but can also generate a wide range of harmonic
sounds.
D The following basic waveforms are available: Sawtooth, Square, Soft Sawtooth, Soft Square, Pulse.
You switch waveforms using the button in the lower left corner, or by clicking directly on the waveform symbol.
D The Amount (AMT) parameter governs the amount of detune.
Turn clockwise for more detune. This is also the modifier (Mod) parameter. Using low Amount settings can pro-
duce subtle detune variations that makes the sound shift and move endlessly, like an advanced chorus effect,
whereas higher Amount settings can produce wild, detuned timbres.
D The Detune Mode parameter sets the basic operational mode of the detuning.
If Amount is set to 0, only the “Octave” and “Fifth” Detune modes actually change the sound, as these modes start
off with dual waveforms tuned one octave and a fifth apart, respectively. The “Fifth Up” and “Oct UpDn” modes de-
tune waveforms as the names imply between zero to full Amount settings. “Linear” will change the amount of de-
tune according to where on the keyboard you play; in lower keyboard ranges the amount of detune is stronger
than in higher keyboard ranges and vice versa. The other modes (Interval and Random) basically add multiple
waveforms and detune them in various ways that will produce different results.

Noise oscillator

The Noise oscillator can not only produce white and colored noise, but can also be used either as a pitched oscillator
or as a modulation source.
It has the following basic parameters:
D There is a single Noise parameter (apart from the standard tuning and kbd track knobs).
This is the Noise modifier parameter, that controls different parameters depending on the selected Oscillator
mode, see below.

542 THOR POLYSONIC SYNTHESIZER

 D The Waveform selector button in the bottom left corner is used to set the Oscillator mode.
The following modes are available:’

| Mode | Description
Band In this mode, the Oscillator knob controls bandwidth. Turned fully clockwise, the oscillator produces pure noise.
Turning the knob counter-clockwise gradually narrows the bandwidth until a pitch is produced. The pitch will
track the keyboard normally if the keyboard (KBD) knob is set fully clockwise.
S/H S/H stands for “sample and hold”, which is a type of random generator. The Oscillator knob controls the rate of
the sample and hold. With high Oscillator knob settings, it produces colored noise with a slightly “phased” sound
quality. With lower rate settings you can use the oscillator as a modulation source like a LFO with random values.
For example, if you modulate the pitch of another oscillator using S/H with a low Rate setting as the source, you
will get stepped random modulation of the pitch.
Static As the name implies, this can generate the sound of static interference if you use low Oscillator settings. The
Oscillator parameter controls Density, i.e. the amount of static. High Density settings generates noise.
Color This produces colored noise, which is basically noise where certain frequency areas are filtered, i.e. cutting or
boosting certain frequency areas in the noise. The Oscillator knob controls Color. With a maximum Color setting
you get white noise, and lower settings produces noise emphasizing lower frequencies.
White This produces pure white noise, where all frequencies have equal energy. There is no associated Oscillator pa-
rameter for White noise.

About Oscillator Sync

Oscillator sync is when one oscillator will restart the period of another oscillator, so that they will have the same base
frequency. If you change or modulate the frequency of the synced oscillator you get the characteristic sound associ-
ated with oscillator sync.

Syncing oscillator

Synced oscillator

In Thor, oscillator 1 is always the syncing oscillator, i.e. oscillator 1 controls the base pitch of oscillators 2 and 3, which
are the synced oscillators.

543 THOR POLYSONIC SYNTHESIZER

 D Switch Oscillator Sync on or off by activating the Sync buttons to the left of Oscillator slots 2 and 3.

D The Sync “BW” sliders to the left of Oscillator slots 2 and 3 allows you to adjust the sync bandwidth.
This allows you to change the character of the oscillator sync. The parameter basically sets how abrupt the reset
is - high bandwidth settings produces a more pronounced sync effect and vice versa. The picture above illustrates
high bandwidth reset - if lower bandwidth settings are used the synced osc curve will be more rounded at the reset
points.

About Amplitude Modulation (AM)

AM (Amplitude Modulation) is often referred to as ring modulation. AM works by multiplying two signals together.
D In Thor, Oscillator 2 amplitude modulates Oscillator 1.
The Ring Modulated output will then contain added frequencies which are generated by the sum of, and the differ-
ence between the two signals. This can be used for creating complex, enharmonic sounds.
D The amount of AM is set using the slider to the left of the Oscillator 1 slot.

544 THOR POLYSONIC SYNTHESIZER

 Mix section

The Mix section allows you to adjust the levels and the relative balance of the three oscillators.
D The two sliders controls the output levels of oscillators 1-2 and oscillator 3, respectively.
D The Balance knob sets the balance between oscillator 1 and 2.
The Balance parameter is also a modulation destination, allowing you to modulate the balance of the two oscilla-
tors with e.g. an LFO. Note that the oscillators have to be connected to the filter(s) via the numbered routing but-
tons for the Mix section settings to have any effect.

Filter slots

Thor has three open Filter slots, two in the Voice section (which act per-voice) and one in the Global section which is
global for all voices (see “Global Filter slot”).
D You select (or change) filter type for a slot by clicking the arrow button in the top left corner of a slot.
On the pop-up you can select between 4 filter types and bypass mode. Available filter types are Ladder LP, State
Variable, Comb and Formant, each described separately below.
The following general rules apply:
D Filters are pre-wired to the Filter Envelope (see “Filter Envelope”).

545 THOR POLYSONIC SYNTHESIZER

 • Filters 1 & 2 can be used serially (i.e. the output of Filter 1 goes (via the Shaper) to the input of Filter 2, or in
parallel (meaning that one signal goes to Filter 1 and another to Filter 2).
The pre-defined routings of the three oscillators into the Filter 1-2 sections is described in the “Basic connections -
a tutorial” passage.

Common parameters
As with the open oscillator slots, there are certain parameters which are common for all filter types.
These are as follows:
D All the filter types have large knobs for the filter frequency (FREQ) parameter and the filter resonance (RES)
parameter.
This works slightly differently for the Formant filter - see “Formant filter”.
D The “KBD” parameter sets how the filter frequency tracks incoming note pitch data.
Some filter types (Ladder/State Variable/Comb) can “self oscillate” and be used as extra oscillator sources.
D The “ENV” parameter sets how much the filter frequency responds to the Filter Envelope.
D The “VEL” parameter sets how much incoming note velocity affects the Filter Envelope Amount.
In other words, for this parameter to have any effect it requires that the “ENV” parameter is set to a value other
than zero.
D The “INV” button inverts how the filter frequency responds to Envelope settings.
D The “Drive” parameter allows you to adjust the input gain to the filter.
By driving the filter harder you can add further character to the sound.
D Any parameter settings, as well as any modulation assigned to parameters, will be kept even if you change the
filter type.

Ladder LP Filter

The Ladder LP filter is a low-pass filter inspired by the famous voltage controlled filter patented by Dr. Robert Moog
in 1965. The name originates from the ladder-like shape of the original transistor/capacitor circuit diagram.
The original filter also had certain non-linear characteristics which contributed to the warm, musical sound it is re-
nowned for. These characteristics are faithfully reproduced in the Ladder LP filter.
There is also a built-in shaper in the feedback (self-oscillation) loop. If self-oscillation is activated (see below), the
shaper will distort the sound to produce these non-linear characteristics. To adjust the intensity of this distortion you
use the Drive parameter.
D There are 4 different Filter slopes available; 24, 18, 12 and 6 dB/oct.
24dB slope comes in two different types:
• Type I - The shaper (controlled with the Drive parameter) is placed at the filter output but before the feedback
loop.
• Type II - The shaper (controlled with the Drive parameter) is placed at the filter input after the feedback loop.

546 THOR POLYSONIC SYNTHESIZER

 Note that “Self Osc” (see below) must be activated for the shaper to operate.
D This filter can self-oscillate and will produce a playable note pitch with high Resonance settings if this is acti-
vated.
Self-oscillation can be switched on or off by using the “SELF OSC” button. The “KBD” knob governs how the fre-
quency tracks the keyboard, turned fully clockwise will produce 12 semitones/octave tracking.

State Variable Filter

This is a multi-mode filter which offers 12 dB/octave slope Lowpass (LP), Bandpass (BP), Highpass (HP), plus
Notch and Peak filter modes which are sweepable between HP/LP states, similar to the vintage Oberheim SEM fil-
ter.
The filter modes are as follows:
D LP 12 (12 dB lowpass)
Lowpass filters let low frequencies through and cut off high frequencies. This filter type has a 12dB/Octave slope.
D BP 12 (12 dB bandpass)
Bandpass filters cut both high and low frequencies, leaving the frequency band in between unaffected. Each slope
in this filter type is 12 dB/Octave.
D HP 12 (12 dB highpass)
Highpass filters let high frequencies pass and cut off low frequencies. This filter type has a 12dB/Octave slope.
D The “Notch” and “Peak” filter modes employ a combination of two outputs from the same filter combining LP
and HP set to the same the filter frequency.
The “LP/HP” knob associated to these two filter modes can modulate the state of the filter from low-pass to high-
pass. If the knob is in the mid-position, you get a Peak or Notch filter slope (depending on the mode). The HP/LP
parameter can be assigned as a modulation destination.
D This filter can self-oscillate and will produce a pitch with high Resonance settings if this is activated.
Self-oscillation can be switched on or off by using the “SELF OSC” button. The “KBD” knob governs how the fre-
quency tracks the keyboard, turned fully clockwise will produce 12 semitones/octave tracking.

547 THOR POLYSONIC SYNTHESIZER

 Comb filter

The Comb filter can add subtle pitch variations and phasing-like effects to sounds.
D Comb filters are basically very short delays with adjustable feedback (controlled with the Resonance knob).
A comb filter causes resonating peaks at certain frequencies. Comb filters are used in various signal processing
devices like flangers, and produces a characteristic swooshing sound when the frequency is swept.
D The difference between the “Comb +” and “Comb –” modes is the position of the peaks in the spectrum.
The main audible difference is that negative Comb mode causes a bass cut.
D The Resonance parameter in both cases controls the shape and size of the peaks.
This filter will produce a pitch with high Resonance settings combined with low frequency settings.

Formant filter

The Formant filter type can produce vowel sounds. There are no Frequency or Resonance parameters, instead you
have a horizontal “X” parameter slider and a vertical “Y” parameter slider that operate together to produce the various
filter formant characteristics.
D You can alter the settings of both the “X” and “Y”parameters simultaneously by moving the “dot” inside the
gray rectangle on the filter panel.
Horizontal movement changes the “X” parameter, and vertical movement the “Y” parameter.

D The ENV-VEL-KBD knobs affect the “X” parameter.

The parameter can be CV controlled.
D The “Gender” parameter changes the basic timbre of the vowel generation between male (low Gender set-
tings) and female (high Gender settings) voice characteristics.
Gender can also be CV controlled.

548 THOR POLYSONIC SYNTHESIZER

 Shaper
Waveshaping is a synthesis method for transforming sounds by altering the waveform shape, thereby introducing var-
ious types of distortion. The Shaper can radically transform the sound or just add a little warmth, depending on the
mode and other settings.
D The Shaper input is taken from the Filter 1 output.

D The Shaper is activated with the button in the top left corner of the section.

D The Drive parameter sets the amount of waveshaping.

Tip: By raising the Filter 1 Drive parameter you can add even more grit and distortion to the Shaper output.
D The Shaper has 9 modes, selectable with the spin controls or by clicking in the Mode display, all which distort
the waveform in various different ways.
These modes are; Soft and Hard clip, Saturate, Sine, Bipulse, Unipulse, Peak, Rectify and Wrap. Exactly how the
various modes affect the sound depends on many factors, and there is a slightly random element to the resulting
distortion. We recommend simply trying the different modes to hear what happens - many interesting types of dis-
tortion of the original signal are guaranteed!

Amp section

The Amp (amplifier) section has two inputs (from Filter 1 & 2) and one output that is routed to the Global section (and
on to the Master Level and the Main Outputs).
D The Gain knob controls the level and the Velocity knob controls the Gain modulation, i.e. how much velocity af-
fects the level - positive values means that you get higher level the faster you strike a key.

D The Pan knob controls the relative stereo position of the individual voices.
By applying modulation to this parameter, you can make individual voices appear in different stereo positions when
you play.

549 THOR POLYSONIC SYNTHESIZER

 LFO 1

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO
modulate the pitch of an oscillator to produce vibrato, but there are countless other applications for LFOs.
D LFO 1 will apply modulation polyphonically.
I.e. if LFO 1 modulation of a parameter is assigned, an individual LFO cycle will be triggered for each note you play.
D You select a LFO waveform by using the spin controls beside the waveform display, or by clicking in the dis-
play and moving the mouse up or down.
The following parameters are available for LFO 1:

| Parameter | Description
Rate This sets the frequency or rate of the LFO.
Waveform This sets the LFO waveform. Apart from standard waveforms (sine, square etc.) there are various
different random, non-linear and stepped waveforms. The shape of the waveforms are shown in
the display, and these shapes basically reflect how a signal is affected.
Delay This introduces a delay before the LFO modulation onset after a note is played. Turn clockwise for
longer delay.
KBD Follow This determines if (or how much) the Rate parameter is affected by note pitch. If you turn the knob
clockwise, the modulation rate will increase the higher up on the keyboard you play.
Key Sync As explained previously, LFO 1 is polyphonic and will produce a separate LFO cycle for each note
played. If Key Sync is off, the cycles are free running, meaning that when you play a note the mod-
ulation may start anywhere in the LFO waveform cycle. If Key Sync is on, the LFO cycles are reset
for each note played.
Tempo Sync If this is on, the Rate will be synced to the sequencer tempo.

550 THOR POLYSONIC SYNTHESIZER

 Envelope sections

There are three Envelope generators in the Voice section. These are the Amp envelope, the Filter envelope and the
Mod envelope. Each voice played has a separate envelope. There is also an additional Global Envelope which is de-
scribed separately - see “Global Envelope”.
D The Filter envelope is pre-wired to control the frequency of Filter 1 and 2.
Note that envelope control of filter frequency can be switched off in each Filter section (the Env parameter can be
set to 0), so the Filter Envelope can be used to control other parameters as well.
D The Amp Envelope is pre-wired to control the amplitude (volume).
Similarly, the Amp envelope can also be used to control other parameters, but in the Voice section you cannot
switch off or bypass the Amp Envelope - if no voice is active (i.e. if there is no gate trigger input to the Amp enve-
lope) there will be no output from oscillators or any external audio source routed to the Voice section.
D The Mod Envelope can be freely assigned to control parameters.
This is done in the Modulation section.

Filter Envelope
The Filter Envelope is a standard ADSR envelope as used in the Subtractor.
D By setting up a filter envelope you control the how the filter frequency or some other parameter should change
over time with the four parameters, Attack, Decay, Sustain and Release.
Please refer to the Subtractor chapter for a description of these parameters.
D The “Gate Trig” button can be used to switch off the envelope triggering from notes (which is the normal
mode) and allow the envelope to be triggered by some other parameter.
“Gate Trig” should normally be activated.
D The time ranges of each step are as follows:
Attack: 0 ms - 10,3 s / Decay and Release: 3 ms - 29,6 s. Sustain is not set as a time but as a level (from Off to
0dB).

551 THOR POLYSONIC SYNTHESIZER

 Amp Envelope
The Amp Envelope is also a standard ADSR envelope.
D By setting up a Amp envelope you control the how the amplitude or some other parameter should change over
time with the four parameters, Attack, Decay, Sustain and Release.
Please refer to the Subtractor chapter for a description of these parameters.
D The “Gate Trig” button can be used to switch off the envelope triggering from note input (which is the normal
mode) and allow the envelope to be triggered by some other parameter.
“Gate Trig” should normally be activated.
D The ranges of each step are the same as for the Filter envelope.

Mod Envelope
This is a general purpose ADR (Attack, Decay, Release) envelope with a pre-delay stage before the Attack phase.
The Delay to Decay phase can also be looped. Apart from standard Attack, Decay and Release stages the Mod Env
has the following parameters:

| Parameter | Description
Delay This can set a delay before the onset of the envelope.
Loop If this is activated, the envelope phase from Delay to Decay will continuously loop.
Tempo Sync If this is on, each stage will have a length that corresponds to beat increments of the current
sequencer tempo. E.g. you can have a 1/4 delay before a 1/16 attack phase followed by a
1/8 decay. Each stage can be set a range from 1/32 to 4/1 (4 bars).
If this is off, the envelope times are free running and can be set in seconds (same time
ranges as for the Filter Envelope).
Gate Trigger The “Gate Trig” button can be used to switch off the envelope triggering from notes (which
is the normal mode) and allow the envelope to be triggered by some other parameter. “Gate
Trig” should normally be activated.

Global section

The Global section contains parameters that affect all voices. It contains two effects, an open filter slot, the Global
Envelope and LFO 2.

552 THOR POLYSONIC SYNTHESIZER

 Effects section
There are two global mono in/stereo out effects, a Delay and a Chorus. These effects affect all voices coming from
the Amp section equally if activated. The effects are placed after the Global Filter in the signal chain.
D There are controls for standard Delay/Time and Feedback parameters.
Chorus vs. Delay differ only in the delay time range - Chorus is for chorus effects, i.e. short delays, whereas Delay
produces echo effects.
D Delay Time can be Tempo Synced.
This is set with the Tempo Sync button - if on the delay time is set in beat resolutions synced to the main se-
quencer tempo.
D The Delay and Chorus effects can also be pitch modulated by a built in LFO (the “Mod” parameters).
“Rate” controls LFO speed and “Amount” the Stereo width.
D Dry/Wet governs the balance between the unprocessed (dry) signal and the effect (wet) signal.

Global Filter slot

This is the Filter 3 slot which can be loaded with one of the filter types. Filter 3 is basically set up as the other filter
slots. The difference is that all voices are mixed together before entering the filter. The “ENV” parameter governs
modulation by the Global Envelope. If you play one note the filter envelope will trigger. Adding new notes while a note
is still held down (legato) will not trigger the filter envelope.
See “Filter slots” for a description of the filter types.

Global Envelope
The Global Envelope 4 is an advanced envelope that is free to use for whatever purpose, but remember it is “single
trigger” so it will not retrigger legato notes as explained above. It is an ADSR envelope with a pre-delay stage and a
hold stage before the decay phase. You can make it Loop and Sync the time settings to the song tempo.
Apart from standard ADSR parameters, the Global Envelope has the following parameters:

| Parameter | Description
Delay This can set a delay before the onset of the envelope.
Loop If this is activated, the envelope phase from Delay to Decay will continuously loop.
Hold This allows you to set a “hold” phase before the Decay.
Tempo Sync If this is on, each stage will have a length that corresponds to beat increments of the current
sequencer tempo. E.g. you can have a 1/4 delay before a 1/16 attack phase followed by a
1/8 decay. Each stage can be set a range from 1/32 to 4/1 (4 bars).
If this is off, the envelope times are free running and can be set in seconds (same time
ranges as for the Filter Envelope).
Gate Trigger The “Gate Trig” button can be used to switch off the envelope triggering from notes and al-
low the envelope to be triggered by some other parameter. This button is normally activated.

553 THOR POLYSONIC SYNTHESIZER

 LFO 2
D The LFO 2 is a standard LFO but is not polyphonic like LFO 1. It is not assigned to any parameter in an “Init”
patch so you have to use the Modulation Routing section to use it.
• Also the LFO 2 “Delay” and “Key Sync” parameters are single trigger, i.e. the LFO will not retrigger these pa-
rameters for legato notes.
D You select a LFO waveform by using the spin controls beside the waveform display, or by clicking in the dis-
play and moving the mouse up or down.
The following parameters are available for LFO 2:

| Parameter | Description
Rate This sets the frequency or rate of the LFO.
Waveform This sets the LFO waveform. Apart from standard waveforms (sine, square etc.) there are various
different random, non-linear and stepped waveforms. The basic shape of the waveforms are
shown in the display, and illustrate how a signal is affected.
Delay This introduces a delay before the LFO modulation onset after a note is played. Turn clockwise for
longer delay.
Key Sync If Key Sync is off, the LFO cycle is free running, meaning that when you play a note the modula-
tion may start anywhere in the LFO waveform cycle. If Key Sync is on, the LFO cycle is reset for
each note played.
Tempo Sync If this is on, the Rate will be synced to the sequencer tempo in beat increments (4/1 to 1/32).

Modulation bus routing section

A modulation bus is used to connect a modulation source to a modulation destination. Both audio signals and control
(CV) parameters are available. This creates a flexible routing system that complements the pre-wired routing in the
Voice panel.

Basic operation - simple tutorial

To illustrate the basic operation of the modulation bus section, let’s set up a simple source to destination modulation
assignment:
! If you currently have unsaved settings you wish too keep, don’t forget to save them first.
1. Select “Reset Device” from the Edit menu.
The Init patch is a simple 1 oscillator/1 filter setup, which produces sound when you play, and will serve the purpose
of this tutorial.
• The left half of the modulation section contains 5 columns, Source, Amount, Dest, Amount and Scale.
Below the column headers there are 7 rows. Each row is a modulation bus where you can have a Source to Des-
tination modulation assigned.
2. Click in the top row of the leftmost Source column.
A pop-up menu appears listing all available Source modulation parameters.

554 THOR POLYSONIC SYNTHESIZER

 The upper half of the menu contains Voice section source parameters, and the lower half contains various global play
and performance-oriented source parameters and the Global Envelope, as well as the Step Sequencer, CV and Audio
inputs.

3. Select “LFO 1” from the pop-up.

This means that LFO 1 is the modulation Source, and this can now be assigned to modulate a Destination parameter.
4. Pull down the “Dest” column pop-up in the top row.
A pop-up menu appears listing all available modulation Destinations. The upper half of the menu contains Voice sec-
tion destinations, and the lower half contains Global section destinations, as well as the Step Sequencer, CV and Au-
dio outputs.
5. Select “Osc 1” from the menu and then “Pitch” from the submenu.

This means that Osc 1 pitch is now assigned to be modulated by LFO 1. Next step is to set the amount of modulation
to be applied.
6. Click in the top row Amount column to the right of the Source column, and move the mouse pointer up and
down to set an Amount value.
Both positive and negative Amount values can be set (+/- 100%).
D If you now play a few notes you can hear the oscillator pitch being modulated by the LFO to produce vibrato.
But the vibrato will be constant, which you probably don’t want. This is solved by assigning a Scale parameter,
which allows you to assign another parameter to control the modulation Amount.

555 THOR POLYSONIC SYNTHESIZER

 7. Pull down the “Scale” column pop-up in the top row.
A pop-up menu appears listing all available Scale parameters. The upper half of the menu contains Voice section pa-
rameters, and the lower half contains various play and performance-oriented parameters and the Global Envelope, as
well as the Step Sequencer, CV and Audio inputs.
A typical controller for vibrato is the Mod wheel.
8. Select “Performance” from the menu and then “Mod wheel” from the submenu.

This means that Osc 1 pitch is now assigned to be modulated by LFO 1, and the amount of modulation is controlled
by the Mod wheel. How much the Scale parameter controls the Amount is set using the “Amount” column for the top
row (to the left of the Scale column).
9. Click in the top row Amount column and move the mouse pointer up and down to set an Amount value.
Both positive and negative Scale Amount values can be set (+/- 100%). To fully control the LFO modulation so
that there is no vibrato when the Mod wheel is set to zero, set the Amount to 100%.

10.The modulation routing is now complete!

You now have full control over the vibrato modulation by using the Mod wheel.
• How much modulation will be applied when the Scale parameter is set to maximum is governed by the Source
to Destination Amount parameter.
• How much the Scale parameter controls the modulation is set with the Scale Amount parameter.
• To clear any assigned modulation routing you can use the “CLR” button to the right of the corresponding bus.

556 THOR POLYSONIC SYNTHESIZER

 About the three modulation routing types
As described in the tutorial, the principal operators of the Modulation bus routing system are as follows:
• You have Modulation Source, Modulation Destination and Modulation Amount parameters.
• Optionally, you have a Scale parameter controlling the Modulation Amount, and a Scale Amount that governs
how much the Scale parameter controls the Modulation Amount.
There are three different types of modulation routing busses available in Thor:
D You have seven “Source –> Destination –> Scale” routing busses.
These are the seven rows in the left half of the Modulation section, as covered in the tutorial.

D There are four “Source –> Destination 1 –> Destination 2 –> Scale” busses.
These are the four top rows in the right half of the Modulation section. This works after the same principle but the
Source parameter can affect two different Destination parameters (with variable Amount settings) and a Scale pa-
rameter that affects the relative modulation Amount for both Destinations.

D Lastly, there are two “Source –> Destination –> Scale 1 –> Scale 2” busses.
This means that a modulation Amount can use two Scale parameters.
An example: You have the Mod Envelope as Source and Oscillator Pitch as the Destination (Amount set whatever
you like). As the first Scale parameter we use the Mod Wheel (Amount set to 100 so that no modulation is applied
when the Mod wheel is at zero), and LFO 1 as the second Scale parameter (Amount set to whatever you like).
When you move the Mod wheel, the pitch modulation amount will be modulated by both the Mod Envelope and
LFO 1 simultaneously.

557 THOR POLYSONIC SYNTHESIZER

 Modulation Sources - Voice section
The following parameters can be used as Voice section modulation Sources:

| Parameter | Description
Voice Key Voice Key lets you assign modulation according to notes. There are 4 modes selectable from the
sub-menus:
• Note - this is keyboard tracking. If a positive Amount value is used and the
destination is filter frequency, the filter frequency will track the keyboard,
i.e. increase with higher notes.
• Note2 - this works similarly to Note but within a repeated octave range.
E.g. if Note2 modulates Amp Pan the pan position will move from left to
right within an octave range then start over. If you play chords normally
over the keyboard the effect will be that notes are randomly spread across
the stereo field.
• Velocity - this applies modulation according to velocity (how hard or soft
you strike the keys).
• Gate - this is Gate on/off. E.g. if applied to oscillator pitch you will get one
pitch value (set by Amount) when a key is pressed, and another value (the
unmodulated pitch) when the key is released.
Osc 1/2/3 This allows you to route the audio output from the oscillators to a destination.
Filter 1/2 This is the audio output of the filters. All filter parameters affect the destination.
Shaper This is the audio output of the Shaper module. Note that anything connected to the Shaper, e.g.
Filter 1, affects the Shaper output, and thus the resulting modulation.
Amp This is the audio output of the Amp Gain section.
LFO 1 This allows you to modulate parameters with LFO 1.
Filter Envelope This allows you to modulate parameters with the Filter Envelope.
Amp Envelope This allows you to modulate parameters with the Amp Envelope.
Mod Envelope This allows you to modulate parameters with the Mod Envelope.

558 THOR POLYSONIC SYNTHESIZER

 Modulation Sources - Global
The following parameters can be used as Global section modulation Sources:

| Parameter | Description
Global Envelope This allows you to modulate parameters using the Global Envelope.
Voice Mixer This allows you to modulate parameters using the Left and Right Mixer inputs.
Last Key This will apply modulation according to the last note played (monophonic), either via MIDI, or
from the Step Sequencer. For example, you can use Last Key to make a filter’s frequency
track notes played by the Step Sequencer.
MIDI Key This applies modulation according to notes globally, not per-voice so in other words it is
monophonic. E.g. if you use MIDI Note as Source and a self-oscillating filter’s frequency as
the destination, the filter will track but you will only be able to play one voice at a time. MIDI
Note is handy for transposing Step patterns in real time.
There are 3 modes selectable from the sub-menus:
• Note - this is keyboard tracking. If a positive Amount value is used and
the destination is filter frequency, the filter frequency will track the
keyboard, i.e. increase with higher notes.
• Velocity - this applies modulation according to velocity (how hard or
soft you strike the keys).
• Gate - this is Gate on/off. E.g. if applied to oscillator pitch you will get
one pitch value (set by Amount) when a key is pressed, and another
value (the unmodulated pitch) when the key is released.
LFO 2 This allows you to modulate parameters with LFO 2.
Performance parameters On this sub-menu you can assign the one of the standard Performance controllers to modu-
late/scale parameters; Mod Wheel/Pitch Bend/Breath/AfterTouch/Expression.
Modifiers This is where you assign parameters and functions to be controlled with the virtual 2 Rotary
and 2 Button controls on the Controller panel.
Sustain Pedal This allows you to assign the Sustain Pedal as a modulation source.
Polyphony This allows you to apply modulation according to how many notes you play. E.g. you could
have a short envelope attack when you play single notes, and a long attack when you play
chords.
Step Sequencer This allows you to apply modulation according to the settings for each step in the Step Se-
quencer.
On the sub-menu you can chose to apply modulation according to Gate/Note/Curve 1 and
2/Gate Length/Step Duration settings for each step.
In addition you have Start and End Trig, which sends a gate trigger at the start and end of the
Step sequence, respectively.
CV Inputs 1-4 These are CV inputs on the back panel which facilitates the use of external modulation
sources, (e.g. the Matrix) in Thor. If connected you can freely assign the external CV to any
modulation destination in Thor.
Audio Inputs 1-4 These are Audio inputs on the back panel which allows you to connect external audio signals
and process these using Thor parameters, or use them as modulation sources. See “About
using the Audio inputs”.

559 THOR POLYSONIC SYNTHESIZER

 Modulation Destinations - Voice section
The following parameters can be used as Voice section modulation Destinations:

| Parameter | Description
Osc 1 There are four modulation destinations available on the Osc 1 sub-menu:
• Pitch - this will affect oscillator pitch (frequency).
• FM - this will frequency modulate the oscillator.
The difference between Pitch and FM is that if a high frequency audio signal (i.e. an oscillator
or an external audio signal) is the source, FM will not alter the basic pitch of the source, only
the timbre. If Pitch is used both the pitch and the timbre will be affected.
• There is also a modifier parameter, which differs depending on what
oscillator type is selected. See “The Oscillator section” for details.
• Osc 2 AM Amount - this will control AM modulation amount from Osc
2. See “About Amplitude Modulation (AM)”.
Osc 2/ Osc 3 Oscillator slots 2 and 3 have the same Destination parameters as Osc 1, except that there is
no AM.
Filter 1/ Filter 2 The following destinations are available on the Filter 1 and 2 sub-menus:
• Audio In - this allows you to connect an audio source (e.g. an oscilla-
tor or an external audio signal) to the filter input.
• Frequency - this controls the filter frequency.
• Frequency (FM) - this will apply filter frequency modulation.
The difference between Frequency and FM is that if a high frequency audio signal (i.e. an os-
cillator or an external audio signal) is the source, FM will not alter the basic frequency of the
source, only the timbre. If Frequency is used both the pitch and the timbre will be affected.
• Resonance - this controls filter resonance.
• Drive - this controls the filter’s Drive parameter.
• Gender - this controls the Gender parameter (Formant filter only).
• LPHPMix - this controls the LP/HP parameter (State Variable filter
only).
Shaper Drive This will control the Shaper Drive parameter.
Amp The Amp section has three destinations on the sub-menu:
• Input - this allows you to connect a source (e.g. an oscillator or an ex-
ternal audio signal) to the Amp input.
• Gain - this controls the Amp Gain.
• Pan - this controls the Pan for each voice. Modulating this parameter
with for example LFO 1 means that the Pan position will modulate dif-
ferently for each voice you play.
Mix The Mixer has three destinations on the sub-menu:
• Osc 1+2 Level - this controls the level of both oscillator 1 and 2.
• Osc 1:2 Balance - you can modulate the level balance between oscilla-
tor 1 and 2, e.g. to sweep from one oscillator to the other.
• Osc 3 Level - this controls the level of oscillator 3.

560 THOR POLYSONIC SYNTHESIZER

 | Parameter | Description
Filter Envelope The Filter Envelope mod destinations are as follows:
• Gate - this is the gate input of the envelope. A gate signal applied to
this input will trigger the envelope.
• Attack - this controls the Attack of the envelope.
• Decay - this controls the Decay of the envelope.
• Release - this controls the Release parameter.
Amp Envelope This has the same destination parameters as the Filter Envelope.
Mod Envelope This has the same destination parameters as the Filter Envelope.
LFO 1 Rate This allows you to control the LFO 1 Rate parameter.

Modulation Destinations - Global

The following Global modulation destinations are available:

| Parameter | Description
Portamento This allows you to control the Portamento time parameter.
LFO 2 Rate This allows you to control the LFO 2 Rate parameter.
Global Envelope The Global Envelope mod destinations are as follows:
• Gate - this is the gate input of the envelope. A gate signal applied to
this input will trigger the envelope.
• Attack - this controls the attack time of the envelope.
• Decay - this controls the decay time of the envelope.
• Release - this controls the release time of the envelope.
Filter 3 The following destinations are available on the Filter 3 sub-menu:
• Left/Right In - this allows you to connect a source to the filter input.
• Frequency - this controls the filter frequency.
• Frequency (FM) - this will apply filter frequency modulation.
• Resonance - this controls filter resonance.
• Drive - this controls the filter’s Drive parameter.
• Gender - this controls the Gender parameter (Formant filter only).
• LPHPMix - this controls the LP/HP parameter (State Variable filter
only).
Chorus The Chorus effect has the following destinations:
• DryWet balance
• Delay (time)
• ModRate
• ModAmount
• Feedback

561 THOR POLYSONIC SYNTHESIZER

 | Parameter | Description
Delay The Delay effect has the following destinations:
• DryWet balance
• Time
• ModRate
• ModAmount
• Feedback
Step Sequencer This allows you to control various parameters belonging to the Step Sequencer.
• Trig - this enables control over the Step Sequencer Run on/off status.
• Rate - this enables control over the Step Sequencer Rate.
• Transpose - this enables control over the Step Sequencer base pitch.
E.g. if you apply MIDI Note as a source to this parameter you can
transpose the sequence by playing notes.
• Velocity - this enables control over the Step Sequencer Velocity re-
sponse.
• Gate Length - this enables control over the Step Sequencer Gate
Length response.
CV Output 1-4 This will allow you to send signals to the CV outputs on the back of the device. Note that you
can send CV signals to audio outputs and vice versa.
Audio Output 1-4 This will allow you to send signals to the audio outputs on the back of the device. Note that
you can send CV signals to audio outputs and vice versa.

Scale parameters
The available scale parameters are the same as the Source parameters.

About using the Audio inputs

The 4 Audio inputs on the back panel can be used to connect external audio sources and process them with Thor’s
parameters.
Note that when routing audio to the Voice section, the following things apply:
• There are only mono inputs in the Voice section.
• You need to send a gate trigger for the audio signal to be heard. This can be done in three ways; by playing
notes, via notes played by the Step sequencer or from CV gate signals.
D Routing audio to Global destinations does not require a gate trigger and stereo inputs are provided.
D The external audio sources can also be used purely for modulation, e.g. you can modulate an oscillators pitch
with an audio signal.
This way you can use the audio input source to modulate any available destinations.

562 THOR POLYSONIC SYNTHESIZER

 Step Sequencer

Thor’s Step Sequencer is a further development of the step sequencers which were often present in vintage analog
modular systems. It can be used for programming arpeggios or short melody sequences. Alternatively, it can be used
purely as a modulation source.
You can have up to 16 steps, and each step can be programmed with various values such as Note pitch, Velocity,
Step Duration etc.

Basic operation
The main parameters and functions are as follows:
D The row of 16 buttons are used to program each step’s on or off status.
A lit button means that the step is active, and a dark button means that the step will be a rest (silent).
D Each step button has a knob above it, which is used to set values for the corresponding step.
D The Edit knob determines what value you set with the step knobs.
The available Edit values are Note (pitch), Velocity, Gate length, Step duration and Curve 1 and 2.
D The Run button starts/stops the step sequencer.
What exactly happens when you press Run depends on the Run mode - see below.

Setting the Run mode

The Run mode is set with the lever beside the Run button. The set mode governs how the step sequencer is played
back when you press Run. The options are as follows:
D Repeat mode - this will repeat the sequence continuously.
Click the Run button again or use the Transport to stop.
D 1 Shot mode - this will play the sequence once then stop.
D Step mode - the Run button steps the sequencer forward one step at a time.
D Off - the step sequencer is inactive.

563 THOR POLYSONIC SYNTHESIZER

 Setting the direction

The Direction parameter is used to set the direction of the step sequence. The following options are available:
D Forward - plays the sequence from the first step to the last.
D Reverse - plays the sequence from the last step to the first.
D Pendulum 1 - plays the sequence from the first step to the last, then from the last step to the first.
I.e. the last and first step is played twice when the sequencer reverses direction.
D Pendulum 2 - plays the sequence from the first step to the last, then from the second last step to the first, i.e.
without repeating the last/first step when reversing direction.
D Random - plays the steps in a random order.

Programming step note pitch

To program step note pitch, you proceed as follows:
1. Make sure that the Step Seq Trigger button is activated in the Controller panel.

2. Set the Run mode to “Repeat”.

You don’t have to use Repeat mode but it makes it easier to follow the following steps.
3. Start the step sequencer by pressing the Run button.
You should now hear a sequence of repeated notes, each with the same pitch (C3). The current step is indicated
by a yellow LED above the step buttons.
4. Make sure that the Edit knob is set to Note.

564 THOR POLYSONIC SYNTHESIZER

 5. Turn one of the step knobs above one of the steps.
A tooltip shows you what current note pitch the knob is set to, and when the sequencer repeats you should be able
to hear the change in pitch for that step. Turn clockwise to raise the pitch in semitone increments. Turn counter-
clockwise to lower the pitch.

D You can set the knob’s note range by using the Octave lever to the left of the step buttons.
Available note ranges are 2 Octaves (i.e. one octave up and down from the middle knob position (C3), 4 Octaves
(i.e. two octaves up and down from the middle position (C3), or Full (-C2 to G8).
q Note that the octave range can be set independently for each step. Each step memorizes the current octave
range when the pitch is set for that step, and will keep this octave range until you change the pitch for the step
with a different octave range setting.
D You can either program steps “on the fly” (with the Step sequencer running) or step by step (Step mode).
In Step mode, you press Run to forward the step number one position so you can set step parameters for one step
at a time.
By using this general method you can continue to enter note pitch for other steps.

Inserting rests
To make step sequences more rhythmically interesting, you can program rests for steps.
D This is simply done by pressing one or several step buttons so they go dark.
Dark steps will be rests.
D Note that the Step Duration value still “counts” for rests.

Setting the number of steps

D You can set how many steps a sequence should have before starting over using the Steps knob at the far right
on the panel.
Up to 16 steps can be used. The lit LEDs above each step button show the number of steps currently used. You
can also change number of steps by clicking on a LED directly - the sequencer will then stop/start over at the se-
lected step.

565 THOR POLYSONIC SYNTHESIZER

 Setting Rate
The Rate knob determines the rate of the step sequence.
• You can either use “free running” rates (i.e. not synced to main sequencer tempo) or synced tempo.
This is set with the Sync button on/off status. If Sync is active you can set the tempo in various beat resolutions.

Setting other values for steps

For each step you can also program other parameters with the step value knobs apart from note pitch. You use the
Edit knob to set one the following:
D Velocity - if this is selected as the Edit mode you can set a velocity value for each step.
Default value is 100, range is 0-127.
D Gate Length - if this is selected as the Edit mode you can set a Gate Length value for each step.
Default is 75%. Gate Length determines the length of the note played for that step.
D Step Duration - if this is selected as the Edit mode you can set a Step duration value for each step.
This parameter determines the total length of the step, which is a factor related to the sequencer rate. Range is 1/
4 to 4. E.g. if Rate is 1/16, “1” means a 1/16-note will be played, a “4” means a 1/4-note will be played, and so on.
D The Curves 1 and 2 allow you to set values for each step that can be sent to control parameters of your choice.
This is done in the Modulation bus routing section, where these two independent Curves are selectable as Source
controllers.
q You can compare these curves to the Curve CV output of the Matrix - they simply represent a series of values
which can be applied to anything.

Step Pattern functions

You will find some specific Step pattern functions on the Edit menu (and on the device context menu). These are as
follows:

| Function | Description
Randomize Sequencer Pattern The Randomize Pattern function creates random patterns. The function only randomizes the se-
lected Edit value (e.g. if set to Note, only the note pitch values are randomized, not velocity, gate
length etc.).
Shift Pattern L/R The Shift Pattern functions move the pattern one step to the left or right. All parameters (rests,
note pitch, velocity etc.) are shifted one step.

566 THOR POLYSONIC SYNTHESIZER

 Connections
The following Audio and CV connectors can be found at the back of Thor:

Sequencer Control Inputs

The Sequencer Control CV and Gate inputs allow you to play Thor from another CV/Gate device (e.g. a Matrix or the
RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers note on/off
along with velocity.

Modulation Inputs
D The Rotary control voltage (CV) inputs (with associated voltage trim pots), can modulate the two virtual Rotary
controls.
Thus, any parameter(s) assigned to a Rotary control can be modulated by CV.
D The Filter 1x allows for CV control of the Filter 1 frequency.
If the Formant filter is used this is the “X” parameter - see “Formant filter”.
D The four CV Inputs can receive CV from external sources that will be available as Sources in the Modulation
bus.

Modulation Outputs
Here you can find CV outputs from the Global Envelope and LFO 2, as well as the 4 user assignable CV outputs.

Audio Inputs
The Audio inputs can be used to connect audio outputs from other Reason devices. When connected, you can route
the Audio inputs as a Modulation source to for example one of the filters and process the external signal. See “About
using the Audio inputs”.

Audio Outputs
Thor has 4 outputs:
• 1 Left (Mono)/2 Right - these are the main stereo outputs.
• 2 additional outputs (3 and 4), which can be assigned in the Modulation section.

567 THOR POLYSONIC SYNTHESIZER

568 THOR POLYSONIC SYNTHESIZER
Chapter 30
Alligator
Triple Filtered Gate
 Introduction

The Alligator is a three-channel gate effect with a built-in pattern player. It can chop up audio in a wide variety of
ways and process it with three parallel filters, distortions, a phaser and a delay. The Alligator can be used for process-
ing sustaining sounds like strings and pads, adding rhythms and accents. It can also be used on loops and other
rhythmic material, changing the feel and sound. Applied to a whole mix, the Alligator can be a powerful remix tool, to-
tally reshaping the material.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

About the Patch format

Alligator patches have the file extension ".gator". The Factory Sound Bank contains a number of Alligator effect
patches for use as is or as starting points for further tweaking. Patches are loaded and saved using the standard pro-
cedures.
q Don't forget that you can also save Alligator settings as part of a Combinator patch. Combining an instrument
device with an Alligator is a quick way to create gated, rhythmic pads.

Overview and signal flow

The Alligator may seem overwhelming at first - it's got quite a few knobs and buttons on its front panel. However,
once you've understood the basic signal flow it's actually pretty straightforward. Read through the description below
and get familiar with the basics - it will help you a lot when working with the Alligator.
Here is a simplified diagram of how the Alligator works:

Gate 1
High Pass Filter

Gate 2 Mixer
Audio Input Band Pass Filter

Gate 3
Low Pass Filter

570 ALLIGATOR TRIPLE FILTERED GATE

 You normally connect the Alligator as an insert effect, so that all of the audio signal passes through the effect device.
The incoming signal is split into three, parallel channels. For each channel, there is a separate gate - when that gate
is open the signal passes through and when it's closed, the channel is silent. The gates can be opened in four ways:
• By the built-in patterns.
There are 64 patterns, each with three "tracks" independently controlling the three gates.
• By clicking the Manual Gate buttons on the front panel.
• By triggering the gates with the MIDI notes F#1, G#1 and A#1.
This way you can play the Alligator live, with velocity control over the gate levels, see “Playing the Alligator live”.
• By connecting CV cables to the Gate inputs on the back of the Alligator and sending Gate signals, e.g. from a
Matrix or Redrum.
When a gate is open, the signal passes through a filter. The three channels have different types of filters: High Pass,
Band Pass and Low Pass, respectively. This means the channels will have different sound characteristics.
Finally, the three channels are mixed together again and sent to the main output.
That's the signal flow in its most basic form. Looking at the front panel, you can see the signal split and the three
channels with their gates and filters:

However, as you can see, there are quite a few other settings as well. Let's take a closer look at one of the channels
(the band pass filter, in this example):

LFO
To High Pass From High Pass

Band Pass FX Pan and

Audio Input (Drive, Phaser,
Amp Env Filter Delay Send)
Volume

To Low Pass From Low Pass

Gate
(pattern, CV,
MIDI or Trig
button) Filter Env

In this, more detailed diagram, we see that the gate isn't a simple on/off switch - there is actually an amplitude enve-
lope controlling the volume of the channel. When the gate is opened, the envelope is triggered and the sound is let
through according to the envelope settings. You can use the amp envelope to soften the attack, to make the notes
shorter and more snappy, etc. The gate also triggers a filter envelope, so that each note can get an articulated filter
contour. The filter can also be modulated by a global LFO.
Next in the channel are FX settings: a distortion unit, a swirling phaser and a send to a built-in delay unit. Since these
settings are independent for the three channels, they can give you a lot of variations.
Finally, there are Pan and Volume controls. Even a function as basic as stereo panning can make for really interest-
ing, spatial effects - especially since you can pan the three channels, the dry signal and the delay independently!

571 ALLIGATOR TRIPLE FILTERED GATE

 Parameters
Common effect device parameters
Like all effect devices, Alligator features a Bypass/On/Off switch and an input level meter. These are described in
“Common effect device features”.

Pattern section

Pattern On
When this is on, the built-in pattern player will run in sync with the song tempo, controlling the three gates. Turn it off
if you want to control the gates manually or with MIDI/CV.

Shuffle
Shuffle on the Alligator works in the same way as shuffle on the Redrum and Matrix devices. It will delay every sec-
ond 1/16th note in the playing pattern according to the Shuffle amount setting in the I/O device, creating a shuffle
or swing feel.
Note that Shuffle will work best when Resolution is set to 1/16.

Pattern selector
This is where you select which one of the 64 built-in patterns should play back, controlling the gates. There is a guide
to the patterns in “The built-in patterns”.

Resolution
When this is set to 1/16 (default) the built-in patterns will be based on 1/16th notes. Changing the Resolution set-
ting allows you to scale the patterns, making them play back faster or slower in relation to the song tempo.

Shift
This will offset the pattern relative to the song playback, moving it “sideways”. The range is ±16 steps, with the step
length determined by the Resolution parameter. For example, if you set Shift to -1 with Resolution at 1/16, the pat-
tern will be moved one sixteenth note to the left. This means the pattern will play one sixteenth note “early” (the start
of the pattern will occur a sixteenth note before the downbeat in the song).

572 ALLIGATOR TRIPLE FILTERED GATE

 Gate and Amp Envelope

Manual Gate Trig buttons

Clicking one of the Manual Trig buttons will open the corresponding gate. It will remain open for as long as you keep
the mouse button pressed. However, the sound may fade out slowly or quickly depending on the amplitude envelope
settings.
While the gate is held open manually, it won’t be affected by the built-in pattern. This means you can use the buttons
to override the pattern, holding a channel open (if Amp Env Decay is long) or muting it (if Amp Env Decay is short).

Gate indicators
These light up when the gates are open.

Amp Env Attack

When a gate is opened, the Amplitude Envelope is triggered. This controls the input level to the corresponding filter.
Amp Env Attack sets how long it takes for the level to reach its maximum after the gate opens. Normally, this is kept
at a low value for quick, snappy attacks. Raising the Attack parameter will make the notes fade in, blurring the pat-
terns.

Amp Env Decay

Directly after the attack phase, the input level will fade down to zero again. The time this takes is set with the Amp
Env Decay parameter. Setting the Decay knob to its maximum value will set the decay time to infinity, which will result
in a maximum “sustain” level. Lowering the Decay setting will make the pattern notes shorter.

Amp Env Release

This determines how quickly the sound fades out after the gate is closed. If you raise this setting, the sound will never
fade out completely between gates, and the pattern will become blurred and more pad-like.

573 ALLIGATOR TRIPLE FILTERED GATE

 Filters and Modulation

The three channels have identical settings, even though their filters are of different types. Below, all descriptions ap-
ply to all three channels, if not explicitly stated.

Filter On button
When this is on, the channel’s signal passes through the filter. Turning this off bypasses the filter. Note though that
the Gate, Amp Envelope, effects and other settings are still active.

LFO Amount
Determines how the filter frequency should be affected by the global LFO (see below). This is a bipolar control, allow-
ing for positive or negative modulation of the filter frequency.

Frequency
q For a general introduction to different filter types, see “The Filter Section” in the Subtractor chapter.
• For the high pass filter, this is the cutoff frequency.
Frequencies below this will be removed from the signal. Turning this parameter up will gradually remove more and
more of the signal, leaving only the highest frequencies.
• For the band pass filter, this is the center frequency.
Lower and higher frequencies will be removed from the signal.
• For the low pass filter, this is the cutoff frequency.
Frequencies above this will be removed from the signal. Turning this parameter down will gradually remove more
and more of the signal, leaving only the lowest bass frequencies.

Resonance
The filter resonance emphasizes the frequencies around the set filter frequency. Turning this up will make the filter
sound more pronounced and ringing.

Envelope Amount
This determines how the filter frequency is affected by the Filter Envelope (see below). This is a bipolar control, al-
lowing for positive or negative modulation of the filter frequency.

574 ALLIGATOR TRIPLE FILTERED GATE

 LFO Waveform

The global LFO offers nine different waveforms, ranging from sine, triangle and square to random and various
stepped forms.

LFO Frequency

Sets the rate of the LFO, used for continuous modulation of the filters. If LFO Sync is activated, the LFO Frequency
is expressed as a note value relative to the song tempo; if not, the LFO Frequency is free.

LFO Sync

Turn this on to synchronize the LFO to the song tempo.

Filter Env Attack

Like the amplitude envelope, the filter envelope is triggered by the gates. There are in fact three individual envelopes,
one for each filter, but they share the same controls. For the filter envelopes to have any effect on the sound, you
need to set the Env Amount parameters to negative or positive values for one or more filter channels.
The Filter Env Attack determines how quickly the filter envelope rises to its maximum value when the gate is opened.

Filter Env Decay

Directly after the attack phase, the filter envelope signal will fall to zero again. The time this takes is set with the Filter
Env Decay parameter.

Filter Env Release

This determines how quickly the filter envelope signal falls to zero after the gate is closed. To fully hear the effect of
this parameter, you need to raise the Amp Env Release parameter - otherwise the level will drop to zero directly when
the gate closes and you won’t hear any filter changes.

575 ALLIGATOR TRIPLE FILTERED GATE

 Effects

The three channels have identical effect parameters. Distortion and phaser effects are separate for the three chan-
nels (although the phasers have common controls). The delay is a global effect, working much like a send effect in a
mixer.

Drive Amount

Sets the amount of distortion for the channel.

Phaser Amount

Sets the amount of phaser effect for the channel.

Delay Amount

This works like an effect send, determining how much of the signal should be sent to the built-in delay effect. The
send is post-volume: If you lower the volume for a channel, the signal sent to the delay will be lowered as well.

Delay Time

This is a standard delay unit with a maximum delay time of 2/4 (when synced to the song tempo) or 1 second.

576 ALLIGATOR TRIPLE FILTERED GATE

 Delay Sync

Turn this on to set the delay time in musical values relative to the song tempo.

Delay Feedback

This determines the number of delay repeats.

Delay Pan

Sets the stereo panning of the delay repeats.

Phaser Rate

The rate of the phaser sweep.

Phaser Feedback

This is similar to the resonance control on a filter. Raise the feedback to get a more pronounced, “singing” phaser ef-
fect.

577 ALLIGATOR TRIPLE FILTERED GATE

 Mix controls

These parameters determine the signal mix being sent to the main outputs on the back. There are also individual out-
puts for the three gate/filter channels. If you connect these outputs, the corresponding channel signals will be re-
moved from the main mix, leaving only the delay return signal and the dry signal.

Channel Pan

Sets the stereo pan/balance of the channel.

Channel Volume

The volume of the channel.

Dry Ducking

The Ducking parameter will apply the Amp Envelope to the dry signal - but inverted. This means that whenever the
Amp Envelope is “high”, the dry signal will be lowered in volume or “ducked”. The result is a sort of mirror to the sound
from the three gated channels.
! Note that this is only audible if the Dry Volume has been raised.

Dry Pan

Sets the stereo pan/balance of the dry, unprocessed signal.

578 ALLIGATOR TRIPLE FILTERED GATE

 Dry Volume

Sets the volume of the dry, unprocessed signal. Mixing in a bit of the dry sound is useful for subtler processing, e.g.
when you just want to animate a pad rather than chop it up.

Master Volume

This is the master volume of the mixed signals. The signals from the separate channel outputs on the back won’t be
affected by this.

Audio connections

Main Inputs and Outputs

The Alligator is normally connected as a stereo in-stereo out effect. Should you connect a mono input signal, the out-
put will still be in stereo due to the pan controls.

Separate Outputs
These output the signals from the individual gate/filter channels. Connecting one of these outputs will remove the
corresponding channel signal from the main output. The separate output signals are taken after the Channel Volume
controls but are unaffected by the Master Volume.

579 ALLIGATOR TRIPLE FILTERED GATE

 CV connections

Gate inputs
These are used for controlling the gates from other devices, using CV. When a gate input receives a CV value of 7 or
higher, the gate will be opened. Higher values result in higher input level for the gate channel (i.e. the gates are ve-
locity sensitive).
• If you select an Alligator device and create a Matrix Pattern Sequencer, its gate output will be auto-routed to
the first available Gate input. Also, the Matrix Curve CV output will be auto-routed to the corresponding CV
Freq input on the Alligator.
You can create up to three Matrix Pattern Sequencers with the Alligator selected and the Matrix devices will be
auto-routed to separate Gate and CV Freq inputs on the Alligator.
! Note: If you want the gates to be controlled by CV only, you need to turn off the Pattern player on the front
panel. Otherwise, the gate sources will be combined.

CV Modulation inputs
These jacks allow you to modulate the filter frequencies of the three filters, as well as the global LFO rate.

Gate Outputs
The three Gate outputs simply send out the current Gate values, regardless of whether these are controlled by the
built-in pattern player, the buttons on the front panel, MIDI or CV. You can use these to trigger other sounds and ef-
fects in time with the gates.

LFO CV Out
This is the output of the built-in LFO, for modulating parameters in other devices.

580 ALLIGATOR TRIPLE FILTERED GATE

 The built-in patterns
Pattern No.

This is an overview of the 64 built-in patterns. The black dots signify open gates with gate 1 (high pass filter) at the
top. Most patterns are two bars long, but some are shorter. All patterns will repeat continuously when the Pattern
function is on.

581 ALLIGATOR TRIPLE FILTERED GATE

 Methods and Tips
Playing the Alligator live
The three gates in the Alligator can be triggered by the MIDI notes F#1, G#1 and A#1, with the lowest note con-
trolling the low pass filter channel and so on. This can be very useful when playing live:
1. Create and connect an Alligator as an insert effect for the audio material you want to process.
This could be a recorded pad, a loop or even the full mix.
2. Make sure the Alligator is selected and select “Create Track for Alligator” from the Edit menu.
This creates a track with a note lane for the Alligator.
D If you like, turn off the Pattern section on the Alligator.
3. Start playback and use the keys F#1, G#1 and A#1 to play the gates.
Remember that the gates are velocity sensitive and that the Amp Envelope settings affect the gated sound.

Playing the gates from Matrix patterns

While the 64 built-in patterns are rather versatile, sometimes you might want to create your very own patterns. The
easiest way to achieve this is to connect Matrix Pattern Sequencers:
1. Select the Alligator device.
2. Create a Matrix from the Create menu or Device palette.
It is automatically connected to one of the Gate inputs on the backside. The Curve CV output on the Matrix is au-
tomatically connected to the corresponding CV Freq input on the Alligator.
3. Repeat these two steps twice, to create two more Matrix devices.
Note that you need to select the Alligator for the Matrix to be auto-routed to the Gate and CV Freq inputs. Now
you have three Matrix devices, one for each gate/channel.
4. Turn off the Pattern player in the Alligator.
5. Use the Gate section in the Matrix display to program custom gate patterns.
The Curve can be used for controlling the Filter Frequency of the corresponding frequency band in the Alligator.
q Another interesting trick is to load a rhythmic loop into a Dr Octo Rex device and connect its Slice Gate Output
to a Gate input on an Alligator. This makes the slices in the REX loop trigger the gate.

Controlling other sounds and effects

The gate outputs on the Alligator back panel allow you to control other stuff with the built-in Alligator patterns. Here’s
an example:
1. Create an instrument with a sustaining sound, such as a pad.
2. Turn on loop mode in the main sequencer and record some chords.
3. Create an Alligator and select a suitable pattern.
You might want to tweak the sound as well, change the filter settings, adjust fx and pan, etc.
With some tweaking, the filtered pattern can sound a bit like a drum beat or percussion loop. Then it can be nice to
add some punch and bottom to the Low Pass channel, making it feel more like a kick drum or bass line:
4. Create a Kong Drum Designer.
5. Load a punchy bass drum sound into Pad 1.
6. Flip the rack around and route a CV cable from Gate 3 Out on the Alligator to the Gate input for Kong’s pad 1.
Now the bass drum sound is triggered along with Gate 3, played by the Alligator’s pattern player.
7. Adjust the Kong level to taste.
8. If you like, add other sounds for Gate 1 and 2 in the same way.

582 ALLIGATOR TRIPLE FILTERED GATE

Chapter 31
Audiomatic
Retro Transformer
 Introduction

The Audiomatic Retro Transformer Rack Extension effect device was designed mainly with focus on spicing up dull
mixes. The Audiomatic Retro Transformer is inspired by the Hipstamatic picture editing app - but designed for audio.
By selecting one of the 16 presets you can instantly change the character of your sound - almost like applying a
“magic skin”.
At moderate levels the Audiomatic Retro Transformer can breathe life into your individual tracks, as well as into your
final mixes, by adding a subtle “shimmer” to your sound. Used to its extremes, it can provide hard-edge and
aggressive sounds as well. You can easily control the mix between dry and processed signal with the Dry/Wet knob
on the front panel.
In Reason the Audiomatic Retro Transformer Rack Extension device can be found on the Effects palette.
! Please, note that this device is not available in Reason Lite Rack Plugin.

Using Audiomatic Retro Transformer

The Audiomatic Retro Transformer is designed to be used mainly as an insert effect, on individual tracks as well as in
the master section. Another very nice application is to use the Audiomatic Retro Transformer as an insert effect in
parallel mixer channels. Then, you can sculpt the effect sound even further using e.g EQ and compressor, for
example.
The front panel layout is very simple and straight-forward: the device features 16 Preset buttons, a Transform knob,
an input Gain knob and a Dry/Wet knob.

Gain

D Set the input level Gain.

Range: - INF to + 12 dB.
! Use the Gain control cautiously when using Audiomatic Retro Transformer on the whole mix. It is easy to get
unwanted distortion in some of the Presets, e.g. Tape, when the input signal is loud.
q Don’t be afraid to push the gain up - especially in presets that feature built-in compression! High input Gain
settings can also render very interesting distorted sonic results. You can always compensate the level by
lowering the Volume, see “Volume”.

584 AUDIOMATIC RETRO TRANSFORMER

 The Presets

The 16 Preset buttons are arranged in a four by four matrix. The upper row of presets contains subtle effects, the
second row a little less subtle and so on all the way down to the fourth row, which contains the most far out effects.
The display shows a picture which reflects the selected effect Preset.
The following Presets are available:

Tape
This simulates the character of an analog tape recorder.

Hi-Fi
This simulates the classic Loudness compensation function, which was very popular in numerous consumer class hi-
fi systems in the 70s and 80s.

Bright
This preset adds brightness to the sound, and removes bass.

Bottom
This preset tightens up and enhances the low frequencies in the sound.

Spread
This preset is a spatial effect, which spreads the stereo width of the sound and also changes the frequency
characteristics.

Radio
This simulates a small transistor radio.

VHS
This simulates sound recorded with a VHS camera.

Vinyl
This simulates the background sound and noise from a somewhat scratchy vinyl record.

585 AUDIOMATIC RETRO TRANSFORMER

 MP3
This simulates the sound of a poorly encoded/decoded MP3 file.

Psyche
A psychedelic sound experience. “Turn on, tune in, drop out”, sort of.

Cracked
The effect of a broken speaker with lots of distortion.

Gadget
A hollow “robotic” type of effect.

Circuit
A “circuit-bending” type of effect with bit-crushing on top.

Wash
A washing machine on sound check at an open air venue?

PVC
A pretty far out sci-fi preset.

Eerie
A very nice and scary waterphone effect.

Transform

D Transform the effect with the Transform knob.

The knob usually controls a combination of different (hidden) parameters in each of the Presets. This makes it
behave more like a “morph” control rather than an “amount” control.

Dry/Wet

D Set the balance between the dry input signal and the transformed (effect) signal with the Dry/Wet knob.
q If you want only subtle effects, turn the Dry/Wet knob more towards the Dry position. For more prominent
effects, turn the Dry/Wet knob towards the Wet position.

586 AUDIOMATIC RETRO TRANSFORMER

 Volume

D Set the output volume with the Volume knob.

Range: - INF to + 12 dB.
q Use the Volume knob in combination with the “Gain” knob, to compensate for high input Gain settings.

Connections

CV Modulation In
Transform
A bipolar CV signal patched here modulates the Transform parameter (see “Transform”). A positive signal increases
the Transform parameter amount and a negative signal decreases it.
q Patch an LFO signal here and use with the PVC Preset, for a nice sweeping sound!
! Note that the default Transform parameter range can not be exceeded.

Dry-Wet
A bipolar CV signal patched here modulates the Dry/Wet front panel control (see “Dry/Wet”). Zero modulation
means that the current Dry/Wet knob setting is valid.

Input L&R
D Patch the audio signals you want to process here.
If your signal is in mono, connect only to the L (left) input.

Output L&R
These are the stereo audio outputs.
• The signal routing (stereo/mono/dual mono) is Preset dependent.

587 AUDIOMATIC RETRO TRANSFORMER

588 AUDIOMATIC RETRO TRANSFORMER
Chapter 32
BV512 Vocoder
 Introduction

The BV512 is an advanced vocoder device with a variable number of filter bands. It also has a unique 1024-point
FFT vocoding mode (equivalent of 512-band vocoding) for very precise and high quality vocoded speech. By con-
necting the BV512 to two instrument devices, you can produce anything from vocoded speech, singing or drums to
weird special effects.
Even if you have worked with a vocoder before, please read the following section. Knowing the basic terms and pro-
cesses will make it much easier to get started with the BV512!
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

How does a vocoder work?

Carrier and modulator
A vocoder accepts two different input signals, a “carrier” and a “modulator”. It analyzes the modulator signal, applies
its frequency characteristics to the carrier signal and outputs the resulting “modulated” carrier signal.
In the most typical case, the carrier signal is a string or pad sound and the modulator signal is speech or vocals - the
result will be a talking or singing synth sound. The modulator could also be drums or percussion (for rhythmically
modulated sounds and effects) or any sound with changing frequency content.

Filter bands
Technically, a vocoder works in the following way: The modulator signal is divided into a number of frequency bands
by means of bandpass filters (called the “modulator filters” or “analyzing filters”). The signal in each of these bands is
sent to a separate envelope follower (which continuously analyzes the level of the signal). The carrier signal is sent
through the same number of bandpass filters (the “carrier filters”), with the same frequency ranges as the filters for
the modulator signal. The gain of each bandpass filter is controlled by the level from the corresponding envelope fol-
lower, and the filtered signals are combined and sent to the vocoder’s output.
In this way, the carrier is filtered to have roughly the same frequency characteristics as the modulator. If the modula-
tor signal has a lot of energy in one of the frequency bands, the gain of the corresponding filter band for the carrier
signal will be high as well, emphasizing those frequencies in the output signal. If there is no signal at all within a fre-
quency band in the modulator signal, the corresponding band in the output signal will be silent (as the gain will be
zero for that filter).
There are several factors determining the quality of the vocoder sound, but the most important is the number of filter
bands. The larger the number of filter bands, the closer will the output signal follow the modulator’s frequency char-
acteristics. The BV512 offers 4, 8, 16 or 32-band vocoding.
q Even if a high number of bands will make the sound more precise and intelligible, this isn’t always what’s de-
sired! Vocoding with a lower number of bands can give results that sound different, fit better in a musical con-
text, etc.

FFT vocoding
The BV512 has an additional FFT mode, in which the vocoding process isn’t based on bandpass filters as described
above. Instead, FFT (Fast Fourier Transform) analysis and processing is used. This equals 512 “conventional” fre-
quency bands and results in a very precise and detailed vocoder sound. Note:

590 BV512 VOCODER

 • The FFT mode is best suited for vocoding speech or vocals, giving crystal clear and highly intelligible results.
It is not so well suited for vocoding drums and percussion, since the FFT process is inherently “slower” than the
regular filtering and doesn’t respond as quickly to transients, and also there will be a slight delay added to the sig-
nal (in the region of 20ms). A workaround solution to this would be to move the modulator signal slightly ahead to
compensate for the delay.
• Where the conventional filter bands are distributed logarithmically (i.e. the same number of filter bands per oc-
tave), the 512 bands in the FFT mode are distributed linearly.
This means a lot of the bands will be in the high frequency range - this is one of the reasons for the clear sound
but it is also something to keep in mind when making settings for the vocoder in FFT mode.

Setting up for vocoding

! Note that the vocoder set-up might differ, depending on which DAW you are using. The important thing in the
vocoder set-up is to be able to route MIDI to an instrument to play the carrier sound - and at the same time also
be able to route the modulator audio (e.g. your voice) to the vocoder. This might require both an instrument
track and an audio track in your DAW. Consult the manual for your DAW for details.
Here are some guidelines for creating a typical vocoder setup. We assume here that you have a MIDI keyboard con-
nected. For details on the parameters, see “BV512 parameters”.
1. Create the instrument device you want to use for the carrier signal.
This could typically be a synth or a sampler. In this example we choose a Subtractor synthesizer.
2. Set up the carrier device (Subtractor) for a sustaining, bright sound.
It’s important to have high frequencies (a lot of harmonics) in the carrier. On the Subtractor, a simple but effective
carrier sound would be based on a sawtooth wave, with the filter fairly open. For more about choosing carrier
sounds, see “Choosing a carrier sound”.
3. Select the carrier device (Subtractor) and create a BV512 Vocoder.
If you flip the rack around you will see that the carrier input of the Vocoder is automatically connected to the Sub-
tractor output:

4. Route the microphone/audio signal to the Modulator Input on the BV512.

Depending on which DAW you are using this routing could be a little different. If possible, connect the appropriate
Audio In on the I/O device to the Modulator input on the back of the BV512.
5. Make sure the “Dry/Wet” knob on the BV512 Vocoder is turned to “Wet” (fully clockwise).
6. Play some notes or chords on your MIDI keyboard and sing through the microphone.
The result should be the classic vocoded vocal sound.

591 BV512 VOCODER

 7. Try the different filter band options and note the difference in sound.

8. You can also adjust the vocoder sound by clicking and dragging the bars in the lower display.
Each bar corresponds to a frequency band, with low frequencies to the left and high frequencies to the right. You
adjust the level of a band by dragging its bar up or down. Clicking and dragging across the bars allow you to
change the levels of several bars, much like drawing an eq curve.

The upper display shows the spectrum of the modulator signal, for display only.
D To reset a band to ±0 dB, press [Ctrl](Win) or [Cmd](Mac) and click on it.
You can also reset all bands to zero by bringing up the context menu for the Vocoder device and selecting “Reset
Band Levels”.
9. If the vocoder sound is “muddy” or indistinct, try raising the “HF Emph” knob on the Vocoder.
This parameter (High Frequency Emphasis) boosts the high frequencies in the carrier signal.
10.Try out the other parameters if you like.
See “BV512 parameters” for details.

Using the BV512 as an equalizer

The BV512 has a unique equalizer mode, in which the device works purely as an insert effect (the modulator input
isn’t used). This allows you to use the processing filters of the vocoder as a kind of graphic equalizer.

Setting up
1. Select the device that you want to process through the BV512.
2. Create a BV512 device.
It is automatically connected to the instrument output(s), using the Carrier Input jacks.
3. Set the switch to the left of the displays to “Equalizer”.

592 BV512 VOCODER

 In use

In equalizer mode, you cut or boost frequencies by clicking and dragging in the lower display - just as with a regular
graphic equalizer. The usage and results differ depending on which mode is selected:
• 4 - 32 band mode
As in vocoder mode, the number of bars in the display conforms to the number of bands selected (4, 8, 16 or 32).
With a higher number of bands you get a more detailed control over the frequency response. However:
D In these modes, the equalizer will “color” the sound even if all bands are set to ±0 dB!
This is due to phase interaction and overlap between the bandpass filters.
Therefore you probably want to use the 4 - 32 band mode for coloring and mutating sounds - not for subtle, “clean”
equalizing.
• FFT (512) mode
In FFT (512) mode you still get 32 bars in the display, but the each bar may control several frequency bands (re-
member that there are 512 bands in FFT mode). Since the frequency bands are distributed linearly in FFT mode,
bars to the left in the display control few frequency bands while bars to the right control many frequency bands.
D In FFT (512) mode, setting all bands to ±0 dB is the same as bypassing the equalizer - the sound will not be af-
fected.
This makes FFT mode suitable for “clean” equalizing, where you want to boost or cut some frequencies without
changing the basic sound character.
D However, FFT mode equalizing is not suited for very drastic frequency cuts or boosts, as this may give audio
artefacts due to the workings of FFT processing.
Still: as always, there are no hard and fast rules. Let your ears judge!
D Keep in mind that FFT mode also introduces a slight delay to the signal.

BV512 parameters
On the front panel of the BV512 Vocoder, you will find the following parameters and displays:

| Parameter | Description
Bypass/On/Off switch In Bypass mode, the carrier signal passes through the device unaffected and the modulator signal is disregarded.
In On mode, the device outputs the vocoded or equalized signal. Off mode cuts the output, silencing the device.
Level meters Show the signal level of the carrier and modulator signals, respectively.
Band switch Selects the number of filter bands (4, 8, 16 or 32) or FFT (512) mode.
Equalizer/Vocoder switch Determines whether the BV512 should work as a vocoder or an equalizer. In Equalizer mode, the Modulator input
is disregarded.
Modulation level display The upper display shows the spectrum of the modulator signal.

593 BV512 VOCODER

 | Parameter | Description
Frequency band level adjust The lower display allows you to adjust the level of each filter frequency band, by clicking and dragging the corre-
sponding bar. In vocoder mode this affects the vocoded sound. In equalizer mode, this is where you cut or boost
frequencies. To reset a band to ±0 dB, press [Command] (Mac) or [Ctrl] (Win) and click on its bar in the display. To
reset all bands, select “Reset Band Levels” from the device context menu.
Note: when FFT (512) mode is selected, each of the 32 bars in the display corresponds to several frequency
bands, with bars to the right in the display controlling progressively more bands (due to the FFT bands being lin-
early distributed over the frequency range).
Hold button Clicking this button “freezes” the current filter settings. While the button is lit, the modulator signal doesn’t affect
the sound - the carrier signal is filtered with the settings as they were the moment you activated Hold. Click the
button again to turn off Hold. Hold is also automatically reset (turned off) when you stop sequencer playback - just
like the pitch bend and modulation wheels on synth devices.
This function can be controlled via CV or MIDI, for sample and hold-like effects.
The Hold button is not available in Equalizer mode.
Attack This is a global attack time control, affecting all envelope followers (see “Filter bands”). Normally you probably want
this set to zero, to make the vocoder react as quick as possible. Raising the Attack time can be useful for “smear-
ing” sounds, creating pads, etc.
Not available in Equalizer mode.
Decay Similarly, this controls the decay time for all envelope followers, i.e. how quick the filter band levels drop. Adjust this
according to taste and context.
Not available in Equalizer mode.
Shift Shifts the carrier filters up or down in frequency, drastically changing the character of the vocoded (or equalized)
sound. This parameter can be controlled via CV, for phaser-like sweeps and special effects.
HF Emph Boosts the high frequencies in the carrier signal. This is sometimes desired to get a clearer vocoded sound. The
(High Frequency Emphasis) reason is that a carrier signal should theoretically contain roughly equal energies in all frequency ranges for best
results - in a typical synth sound the high frequencies are often weaker than the low frequencies. Raising the HF
Emph control will rectify this.
Not available in Equalizer mode.
Dry/Wet Determines the balance between modulator sound (dry) and vocoded sound (wet). To get the pure vocoder sound,
set this to wet (turned fully right).
Not available in Equalizer mode.

594 BV512 VOCODER

 Connections

The back panel of the BV512 offers the following connections:

Individual band levels

These are CV outputs and inputs.
• The upper row outputs CV signals generated by the envelope followers for each frequency band.
• The lower row are CV level inputs to the individual bandpass filters through which the signal is processed (the
“vocoder filters”).
Connecting a CV signal to one of the inputs breaks the internal signal path from the corresponding envelope fol-
lower (in other words, that frequency band is now controlled by the CV signal you’ve connected - not by the corre-
sponding frequency band in the modulator signal).
• If 16 band mode is selected, each output/input pair corresponds to a separate frequency band.
In 8 band or 4 band mode, only the 8 first or 4 first output/input pairs are used. In 32 band mode, each output is
a mix of two adjacent frequency bands and each input controls two bands. Finally, in FFT (512) mode each out-
put/input pair corresponds to several frequency bands.
There are several interesting uses for the Individual band levels connectors: you can cross-patch frequency bands so
that e.g. low frequencies in the modulator signal controls high frequency bands in the vocoder, you can extract CV
signals for controlling synth parameters in other devices, you can base the vocoding on CV signals from other devices
rather than on a modulator signal, etc. See “Using the individual band level connections” for details.

Other CV connections
| Connection | Description
Shift (CV in) This allows you to control the Shift parameter from an external CV source. A sensitivity knob determines how much
the Shift setting is affected by the CV signal.
Hold (Gate in) When a gate signal is sent to this input, the Hold function is activated (see “Hold button”). Hold remains on until the
gate signal “goes low” (falls to zero). By connecting e.g. a Matrix to this input, you can create “stepped” vocoder
sounds, sample and hold-like effects, etc.

Audio connections
| Connection | Description
Carrier input This is where you connect the instrument device that provides the carrier signal (or the device to be processed in
Equalizer mode) - typically a synth or sampler device. The vocoder can handle mono or stereo carrier signals.
Modulator input This is where you connect the instrument device that provides the modulator signal, in mono. This connection is not
used in Equalizer mode.
Output In Vocoder mode, the outputs carry a mix between the vocoded signal and the modulator signal (as set with the
Dry/Wet control on the front panel). In Equalizer mode the output is the carrier signal, processed through the equal-
izer filter.
Note that the output will be in mono if the Carrier input is in mono, and vice versa - the BV512 does not process
mono into stereo.

595 BV512 VOCODER

 Tips and tricks
Choosing a carrier sound
As always, which carrier sound to choose is a matter of taste and musical context. However, here are a few guidelines
to help you get a good result:
• The carrier sound should preferably have a lot of harmonic content (brightness) - dark or muffled sounds will
not “give the vocoder much to work with”.
• Often, you want the carrier sound to sustain at an even level (i.e. it shouldn’t “die out” when you hold a chord).
Similarly, you most often want a reasonably fast attack (although not with a distinct, sharp click or edge).
• You may want a sound that is rather static over time, without drastic envelope control of filter cutoff for exam-
ple.
• If you want to play vocoded chords, the carrier sound must of course be polyphonic.
Here are some hands-on suggestions for carrier sounds:
D A simple Subtractor pad based on a sawtooth waveform.
You could simply start with the initial patch (as set up when you create a new Subtractor device). Open the filter,
turn off envelope modulation of the cutoff frequency and raise the Amp Envelope Sustain.
If you want a classic, rich chorus-like sound, use two detuned oscillators - or better still, add a UN-16 Unison de-
vice as an insert effect between the Subtractor and the vocoder!

A simple but effective carrier sound setup.

D A similar fat carrier sound can be obtained using a Malström device with a patch based on the “Sawtooth*16”
graintable.
With the Malström you can get a stereo carrier signal with no extra devices: simply select the “Sawtooth*16”
graintable for both oscillators, detune the oscillators slightly with the Cent controls and raise the Spread parameter
to the desired stereo width. No filter routings are necessary.
D For a more distinct and precise sound, try using a narrow pulse waveform.
You get this by selecting e.g. a sawtooth wave on the Subtractor, setting the Phase Mode selector to “–” and turn-
ing the Phase knob to the left until you get the desired sound. This type of carrier sound lends itself well to mono-
phonic vocoder lines in the lower registers.
D Use noise as a carrier.
Try using pure noise (possibly filtered down a bit) for robotic voices, whispering and special effects. It’s also very
useful to add a bit of noise to a sawtooth or pulse sound - this makes vocoded speech clearer and more intelligible.
D Use sampled strings or choir sounds.
A rich drawbar organ sample can also be a cool carrier sound.

596 BV512 VOCODER

 D For unusual vocoder sounds, try using the Malström as carrier device, e.g. with a glassy, digital pad sound se-
lected.
Try turning up the Attack and Decay controls on the BV512, for smeared, rhythmic or pseudo-random modulation
of a pad.

Choosing a modulator sound

The modulator sound should typically have varying level and harmonic content. As we’ve already mentioned, the most
typical modulator sounds are vocals or speech and drums or percussion.
D The quickest way to get an “internal” modulator sound is to use a rhythmic loop in the Dr. Octo Rex device.
This way you don’t have to program a rhythm pattern. On the other hand, using a Redrum as modulator allows you
to create exactly the rhythm you want and fine-tune the sounds and the groove.

Using the modulator as carrier

You can get cool special effects by using the same device both as carrier and as modulator. For example, try process-
ing a Redrum device in the following way:
1. Create a Redrum device and set up the desired patch and pattern.
2. Create a Spider Audio Merger & Splitter device.
3. Create a BV512 Vocoder.
4. Flip the rack around and connect the devices in the following way:

The output of the Redrum goes into the splitter section of the Spider, and is split into two signals. One signal goes into the carrier input
of the vocoder, the other goes into the modulator input.

597 BV512 VOCODER

 This is essentially the required connections, but for best results it’s a good idea to add some distortion and/or com-
pression to the carrier signal - this increases the amount of high frequencies in the carrier signal:
5. Press [Shift] and create a Scream 4 distortion device.
6. Connect the distortion device as an insert effect between the Spider and the carrier input of the vocoder.
Now, the carrier signal will be processed in the distortion device, but not the modulator signal.
7. Play back the pattern and experiment with the settings on the vocoder and distortion device.
D This technique can also be used to process vocals and speech.
D Try adjusting the Shift parameter for new effects and sounds.
Remember that you can route CV to the Shift parameter on the back of the BV512 - use e.g. a Matrix or an LFO
output on a synth device!

Controlling the Hold function

As described in “Hold button”, pressing the Hold button on the front panel “freezes” the current filter spectrum until
you deactivate it again. This can be used for creating sample & hold-like effects, stuttering or garbled vocoder
sounds:
D Connect e.g. the Gate output on a Matrix device to the Hold input on the back of the BV512.
By playing back a gate pattern on the Matrix, the Hold function will repeatedly be turned on and off according to
the programmed rhythm in the pattern. Hold will be active for the length of each gate signal.
D Automate the Hold function with the main sequencer, either by recording it or by drawing in its controller lane.
• If you route MIDI to the BV512 you can control the Hold function in two ways by default: By pressing a damper
pedal connected to your MIDI controller or by playing the note C4.
In both cases, the Hold function will be momentary - Hold is on until you release the pedal or key.

Using the individual band level connections

As described in “Individual band levels”, the individual band level connectors on the back are CV output and input
jacks. The upper row sends out the CV signals from the envelope followers for the different frequency bands, while
the lower jacks are CV inputs for controlling the individual bandpass filters (breaking the internal connection from the
envelope followers). There are several interesting things you can do with these connections:

Crosspatching frequency bands

By connecting outputs to inputs in alternative configurations, you can drastically change the result of the vocoding.
For example, you could have low frequencies in the modulator signal give high frequencies in the vocoded sound and
vice versa. Note:

598 BV512 VOCODER

 • In 4 band and 8 band mode, only the 4/8 first output/input pairs are used.
• In 32 band mode and FFT (512) mode, each connection corresponds to two or several frequency bands.
This means that connecting an output to the input with the same number is not the same as using the internal sig-
nal path (no CV cable connected). You can hear this quite clearly in FFT (512) mode: connect all outputs to the
corresponding inputs and gradually remove the CV cables while listening to the vocoder sound - the sound will
progressively get more detailed.

Extracting CV from the vocoder

You can connect an individual band level output to any CV input on any device. This means you can use the vocoder
as an envelope follower, having elements in the modulator sound control a parameter in another device, e.g. an effect.
Note:
• The Attack and Decay settings on the BV512 panel affect the envelope followers, and thus the rise and fall
times of the CV signals from the individual band level outputs.
• If you are using the vocoder in a mode with many bands, but want a broader frequency range to generate the
CV signal, you can merge several band outputs into one CV signal - use a Spider CV Merger & Splitter device.

Controlling vocoder bands from an external source

Connecting a CV source to an individual band input breaks the internal connection from the corresponding envelope
follower. This way you can “manually” control the vocoder filters. Some applications:
D Connect the CV outputs for one or more envelopes in the carrier device to individual band inputs.
When you play the carrier instrument, one or more of the bandpass filters in the vocoder will automatically open,
adding an extra attack to the sound. Useful if you really want to “play” the carrier, rather than just hold a chord.

599 BV512 VOCODER

 D Connect the gate outputs on a Redrum to individual band level inputs.
With this connection (and no device connected to the Modulator input), the Redrum will serve as a pattern se-
quencer, opening and closing different filter bands. To adjust the gate times, set the drum sounds to Gate mode
and use the Length parameter. The result is totally different from using the audio signal of the Redrum as modu-
lator.

The vocoder bands are now solely controlled by the gate signals from the drum channels - the modulator input isn’t used.
Note that you can use a Spider CV Merger & Splitter device to split a gate signal, sending it to several bands. Also, note that the
velocity of the programmed drum notes govern the level of the corresponding filter bands.

“Playing” the vocoder from a MIDI keyboard

! To route MIDI notes to the BV512 in Reason Rack Plugin, you need to put it in a Combinator and turn on "Re-
ceive Notes" in the programmer (see “About the Key (receive MIDI Notes) column” in the Combinator chapter).
If you have routed MIDI to the BV512, playing notes from C1 and up will control individual filter bands. For example,
in 16 band mode, C1 controls band 1, C#1 band 2 and so on up to D#2 (which controls band 16).
• The level of the bands is proportional to key velocity (how hard you play).
• A band will be “open” until you release the corresponding key.
• Bands to which you have connected a CV signal (using the individual band level inputs on the back panel) will
not respond to MIDI keys.
Note that with this function, you “play the modulator bands”. However, you still need both a carrier and a modulator
signal to get any sound. Typically, you would first record the notes or chords for the carrier device in the sequencer,
then create a sequencer track for the vocoder and “play” it from your MIDI keyboard while playing back the recorded
carrier notes and at the same time inputting a signal on the Modulator input.
q An interesting application of this is to patch the vocoder as an insert effect for the whole mix (the output of the
main mixer connected to the carrier input), and “play the vocoder” while inputting a signal on the Modulation
input. Only the frequency bands for which you press keys will be attenuated. Use the FFT (512) mode for best
results.

600 BV512 VOCODER

 Using the BV512 as a reverb
This is a very special trick which can be quite cool. Proceed as follows:
1. Create a Mixer 14:2 device from the Utilities Palette.
2. Create a Redrum device.
The “vocoder-reverb” is best suited for drums, even though nothing stops you from using it on other sounds.
3. Create a Subtractor and a vocoder.
The Subtractor will automatically be routed to the carrier input. We don’t need a dedicated modulator device in this
setup.
4. Flip the rack around and connect Aux send 1 on the Mixer to the modulator input on the vocoder.
5. While you’re there, re-route the vocoder output to Aux return 1.
This way, our vocoder-reverb will be connected as a regular send effect.

6. Set the vocoder to FFT (512) mode, turn the Decay knob to between 6 and 7 and turn the Dry/Wet control to
“wet” (fully right).
7. On the Subtractor, set up a noise sound as follows:
D Turn the Oscillator Mix knob fully to the right.
D Turn on the Noise section (but make sure Osc 2 is off).
D In the Noise section, turn Color to around twelve o’clock.
D Open the filter fully and make sure resonance is set to 0.
D Make sure Filter Envelope Amt is 0 (and turn off velocity modulation).

601 BV512 VOCODER

 D Raise the Sustain to full in the Amp Envelope section.

Now we want the Subtractor to play a continuous noise. You could just route MIDI to it, play a note and keep it
pressed, but that will probably wear you out in the long run. Better to use a Matrix:
8. Create a Matrix and route it to the Subtractor.
We really only need the Gate connection - the note number isn’t important with the noise patch.
9. Set up a one step pattern with a tied gate (press [Shift] and draw the gate) and start playback on the Matrix.
Now the vocoder gets a continuous noise signal as carrier.
10.Create a suitable drum pattern on the Redrum and start pattern playback.
11.Gradually turn up send 1 for the Redrum channel in the mixer.
This now serves as a balance control between the dry drum sound and the reverb, generated by the voccoded
noise! Set it to a suitable reverb level.
12.Use the Decay control on the vocoder to adjust the reverb decay time.
13.Use the Noise Color control on the Subtractor to make the reverb darker or brighter.
You could use the filter cutoff for this as well.
That’s it - a pretty good reverb sound with a lot of control. Although the settings above give the most natural sound,
you can vary the sound and create special-FX reverb in the following ways for example:
D Switch the vocoder to a lower band mode.
D Lower the cutoff and add some resonance in the Subtractor filter.
D Modulate the Subtractor filter with a fast LFO.
D Set the Subtractor filter to HighPass mode to remove the bottom end from the reverb.
D Turn off the Matrix controlling the Subtractor and “play” the noise patch yourself (or from the sequencer). This
way you can create gated reverb effects, etc.

602 BV512 VOCODER

 Creating a stereo reverb
What you’ve got above is a mono reverb. Here’s how to make it stereo:
1. Select the Subtractor and create a Spider Audio Merger & Splitter device.
2. Create a DDL-1 delay.
3. Connect the devices in the following way:
The Subtractor output should be routed to a Splitter input on the Spider. One split output should be routed to one
of the carrier inputs on the vocoder, the other split output should be routed to the delay. The delay output (mono)
should be routed to the other carrier input on the vocoder.

The vocoder will now get a “fake-stereo” noise carrier signal.

4. Make sure the output from the vocoder is connected in stereo to the Aux return on the Mixer.
5. Finally turn down the Feedback on the delay, set it to all “wet” and set the decay time to a second or so.
When you now start playback on the Redrum, the reverb will be in stereo!

603 BV512 VOCODER

604 BV512 VOCODER
Chapter 33
Channel Dynamics
Compressor & Gate
 Introduction

Channel Dynamics Compressor & Gate is the rack version of the Dynamics section in Reason’s Main Mixer. The rack
version is mainly intended for use in the VST3 plugin version of Reason, since it lacks the Main Mixer. However, you
can of course use it wherever you like - also in the stand-alone Reason program. The Channel Dynamics Compressor
& Gate device has identical specifications compared to the channel strip version, with the addition of Input Gain and
Dry/Wet Mix controls.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel reference
Global controls
Input Gain

D Adjust the input gain to the Channel Dynamics device.

Range: +/- 18.00 dB
! The Input Gain knob controls the input signal, regardless if the Compressor and/or Gate is on or not.

Mix

D Set the mix between dry and processed signal with the Mix knob.
By setting a mix of <100% you can use the Channel Dynamics device for “parallel processing”, i.e. the dry and ef-
fect signals are being output together.

606 CHANNEL DYNAMICS COMPRESSOR & GATE

 The Compressor section

Compressors reduce dynamic range by evening out the difference between loud and quiet signals. This makes signal
levels easier to balance, and can add punch and sustain to the sound.
The compressor/limiter in Channel Dynamics is a flexible processor which has soft-knee (a gradual, smooth onset of
compression) characteristics but can be switched to peak limiting, where levels above the set threshold are instantly
reduced. The compressor also features automatic make-up gain. The parameters are as follows:

On
D Click the On button to activate the Compressor section.

Peak
D Activate to change the signal detection from RMS to Peak, which results in an instant attack time.
Peak mode is suitable for compression of sounds with fast attacks like drums.

Fast
D Click this to make the compressor react to signals above the set Threshold level in a fixed time of 3 ms for
20 dB gain reduction.

Ratio
D Set the amount of gain reduction applied to the signal above the set Threshold (see below).
Range: 1:1 (no reduction) to Infinite:1

Threshold
D Set the level at which onset of compression should occur.
Signals below the Threshold setting are unaffected, but when the level exceeds the threshold, compression kicks
in. Automatic make-up gain (based on the Ratio and Threshold settings) is applied to compensate for level reduc-
tion caused by compression, to maintain a steady output level.
Range: -52 dB to 0 dB

Release
D Set the time it should take before the compressor lets the sound through unaffected - after the signal level
drops below the set threshold.
Set this to short values for “pumping” compressor effects, or to longer values for a smoother change of the dynam-
ics.
Range: 100 ms to 1000 ms

Compressor Gain Reduction Meter

The LED meter shows the gain reduction applied by the compressor.

607 CHANNEL DYNAMICS COMPRESSOR & GATE

 The Gate/Expander section

Gating or expansion will attenuate signals below a set threshold; the opposite of compression. It can be used to re-
duce or eliminate unwanted background noise that may be present when there is no signal to mask it. Gating is also
commonly used to reduce microphone “bleeding”, e.g. when recording a close-mic’ed drum kit you can use gating to
silence the tom microphones when the toms aren’t being played to tighten up the sound, and for special effects like
“keying” (see below).
Higher expansion ratios (10:1 and above) are referred to as noise gating, where the channel is completely silenced
if the level drops below the set threshold.
The Gate/expander has the following parameters:

On
D Click the On button to activate the Gate/Expander section.

Exp
In Expander mode, the Range (gain reduction) is not as severe as in Gate mode, but more gradual around the Thresh-
old. Also, the gain reduction is not attenuated by a certain dB but is scaled. The major difference compared to the
Gate function is that instead of muting the signal once it drops below the Threshold (gate), the Expander still lets
some signal through but at a lower level, resulting in a signal of lower volume. You could think of an Expander as an
“inverted” compressor, i.e., it expands the dynamic range of the input signal.
D Click the Exp button to change the operating mode from Gate to Expansion.
The Range knob (see below) then controls the expansion amount.

Hold
D Set the time the gate should stay fully open after the signal falls below the Threshold.
Hold interacts with the Release parameter such that Release only starts acting after the set Hold time.
Range: 0 ms to 4000 ms

Fast
The normal attack time for the Gate is normally1.5 ms per 40 dB.
D Click the Fast button to lower the attack time to 100µs (microseconds) per 40 dB.
This can be is useful for percussive material were the waveform rises steeply in a very short time.

Range
D Set the amount of gain reduction applied to signals below the set Threshold.
The Range can be set from 0 dB (no reduction) to -40 dB. If the Exp button is on (see above), the Range knob
controls the expansion amount.

608 CHANNEL DYNAMICS COMPRESSOR & GATE

 Threshold
D Set the level at which the gate opens or closes.
Signals below the Threshold setting are gated, but when the level exceeds the threshold, the gate opens.
Range: -52 dB to 0 dB
! Note that the threshold for closing the gate is slightly lower than the threshold for opening the gate. This is to
avoid undesirable gate triggering with signal levels close to the set threshold.

Release
D Set the time it should take for the gate to go from open to fully closed.
Fast release times will fade the signal abruptly once the level falls below the threshold, and longer release times
will slowly fade out the signal.
Range: 100 ms to 1000 ms

Gate Gain Reduction Meter

The LED meter shows the gain reduction applied by the gating/expansion.

External Sidechain

You can use external signals to trigger the Compressor and/or the Gate/Expander. This is done by connecting an
external signal to the Sidechain Inputs on the back of Channel Dynamics.

The Sidechain inputs.

When a cable is connected to the Sidechain Input(s) the “Connected” LED is lit on the front panel:

D Click the Sidechain button to “key” (trigger) Channel Dynamics from the external signal instead of the channel
signal.
For example, you could use a drum loop to trigger the gate for a channel playing a synth pad to create rhythmic
chord effects.

609 CHANNEL DYNAMICS COMPRESSOR & GATE

 Connections

CV Outputs
These two CV modulation outputs can be used for controlling other devices that feature CV modulation inputs. The
modulation parameters are:
• Compressor Gain Reduction (see “Compressor Gain Reduction Meter”)
• Gate Gain
This CV signal is high when the gate is open and goes low when the gate closes.

Sidechain Input Left & Right

D Patch an external audio signal to be used as sidechain signal here.
See “External Sidechain” for more information.

Input Left & Right

D Patch the audio signals you want to process here.
If your input signal is in mono, connect only to the L (left) input.

Output Left & Right

These are the audio outputs.

610 CHANNEL DYNAMICS COMPRESSOR & GATE

Chapter 34
Channel EQ Equalizer
 Introduction

Channel EQ Equalizer is the rack version of the EQ section in Reason’s Main Mixer. The rack version is mainly
intended for use in the VST3 plugin version of Reason, since it lacks the Main Mixer. However, you can of course use
it wherever you like - also in the stand-alone Reason program. The Channel EQ device has identical specifications
compared to the channel strip version.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel reference
Global controls

Gain
D Adjust the Channel EQ gain with this knob.
This is useful for level-compensating when doing drastic boosts or cuts.
Range: +/- 18.00 dB
The signal level is shown in the meter.

The Filter section

The Filter section contains a 12 dB/octave low pass filter and an 18 dB/octave high pass filter.
The parameters are as follows:

612 CHANNEL EQ EQUALIZER

 HPF On
D Click the On button to activate the high pass filter.

HPF Freq
The High Pass filter (HPF) removes low frequencies from the signal, resulting in a thinner sound. The HP filter slope
has a 18 dB/Octave roll-off.
D Set the desired cutoff frequency for the high pass filter.
Range: 20 Hz - 4 kHz

LPF On
D Click the On button to activate the low pass filter.

LPF Freq
The Low Pass filter (LPF) removes high frequencies from the signal, making the sound less bright. The LP filter has
12 dB/Octave roll-off curve.
D Set the desired cutoff frequency for the low pass filter.
Range: 100 Hz - 20 kHz.

The Equalizer section

The Equalizer section features a four-band EQ with parametric midrange controls and high and low frequency shelv-
ing bands. The EQ can be switched between two operating modes, each with slightly different curve characteristics.
The Equalizer has the following parameters:

LF Gain
The LF section provides low frequency shelving equalization. All frequencies below the set LF Frequency will be cut
or boosted by the set LF Gain amount.
D Set the LF Gain/Attenuation amount.
Range: +/- 20 dB

LF Frequency
D Set the LF Frequency.
Range: 40 Hz - 600 Hz

613 CHANNEL EQ EQUALIZER

 LF Bell
D Click the LF Bell button to switch the LF EQ to peaking characteristics.
This means it works like a regular parametric EQ band, cutting or boosting the signal around the set LF Frequency.
Bell mode has a fixed bandwidth or "Q" value.

LMF Gain
The low medium frequency EQ is fully parametric.
D Set the LMF Gain/Attenuation amount.
Range: +/- 20 dB

LMF Frequency
D Set the LMF (center) Frequency.
Range: 200 Hz - 2 kHz

LMF Q
The “Q” parameter adjusts the bandwidth around the set center LMF Frequency. The higher the Q value, the narrower
the affected frequency range - except in “E” mode (see “E Mode”).
D Set the Q value for the LMF EQ.
Range: 0.70 - 2.50

E Mode
D Click to switch to E Mode.
When the E Mode button is activated the EQ will have slightly different curve characteristics. In normal mode (E
button deactivated), the Gain setting will also affect the bandwidth (Q value) for the LMF and HMF EQs. The
higher the Gain, the narrower the bandwidth and vice versa.
With E Mode activated, the bandwidth is constant at all Gain settings.

HMF Gain
The high medium frequency EQ is fully parametric.
D Set the LMF Gain/Attenuation amount.
Range: +/- 20 dB

HMF Frequency
D Set the HMF (center) Frequency.
Range: 600 Hz - 7 kHz

HMF Q
The “Q” parameter adjusts the bandwidth around the set center HMF Frequency. The higher the Q value, the nar-
rower the affected frequency range - except in “E” mode (see “E Mode”).
D Set the Q value for the LMF EQ.
Range: 0.70 - 2.50

614 CHANNEL EQ EQUALIZER

 HF Gain
The HF section provides high frequency shelving equalization. Frequencies above the set corner Frequency will be
cut or boosted by the set HF Gain amount.
D Set the HF Gain/Attenuation amount.
Range: +/- 20 dB

HF Frequency
D Set the HF Frequency.
Range: 1.5 kHz - 22 kHz

HF Bell
D Click the HF Bell button to switch the HF EQ to peaking characteristics.
This means it works like a regular parametric EQ band, cutting or boosting the signal around the set HF Fre-
quency. Bell mode has a fixed bandwidth or "Q" value.

615 CHANNEL EQ EQUALIZER

 Connections

CV Inputs
These six CV modulation inputs, with associated trim pots, can be used for controlling the following parameters from
CV modulation sources:
• HPF Frequency (see “HPF Freq”)
• LPF Frequency (see “LPF Freq”)
• HMF Gain (see “HMF Gain”)
• HMF Frequency (see “HMF Frequency”)
• LMF Gain (see “LMF Gain”)
• LMF Frequency (see “LMF Frequency”)

Input Left & Right

D Patch the audio signals you want to process here.
If your input signal is in mono, connect only to the L (left) input.

Output Left & Right

These are the audio outputs.

616 CHANNEL EQ EQUALIZER

Chapter 35
Master Bus Compressor
 Introduction

The Master Bus Compressor is the rack version of the Master Bus Compressor in Reason’s Main Mixer. The rack
version is mainly intended for use in the VST3 plugin version of Reason, since it lacks the Main Mixer. However, you
can of course use it wherever you like - also in the stand-alone Reason program.
The Master Compressor is perfect for providing the final “fairy dust” to your mix. It can add punch and cohesion, and
generally make the mix sound bigger and more powerful. It’s perfect for use on a drum channel or in a mixer bus. The
compressor is very straightforward in operation and features make-up gain as well as program-adaptive Release.
The Master Bus Compressor device has identical specifications compared to the Master Section version, with the ad-
dition of Input Gain and Dry/Wet Mix controls.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel reference
Global controls

Mix
D Set the mix between dry and processed signal with the Mix knob.
By setting a mix <100% you can use the Master Bus Compressor for parallel processing, i.e. the dry and effect
signals are being output together.

618 MASTER BUS COMPRESSOR

 Compression controls

Compressors reduce the dynamic range by evening out the difference between loud and quiet signals. This makes
signal levels easier to balance, and can add punch and sustain to the sound, as well as “glue” together a final mix.
The parameters are as follows:

Input Gain
D Adjust the input gain to the Master Bus Compressor.
Range: +/- 18.00 dB

Threshold
D Set the level at which onset of compression should occur.
The lower the Threshold, the more compression is applied.
Range: -30 dB to 0 dB

Ratio
Ratio specifies the amount of gain reduction applied to signal levels above the set Threshold. A 2:1 compression ratio
effectively means that a signal level 2dB above threshold will have a signal gain of 1dB.
D Set the amount of gain reduction applied to the signal above the set Threshold.
Range: 2:1, 4:1 and 10:1

Attack
D Set the time it should take before the compressor should react to signals above the set Threshold.
Range: 0.1 ms, 0.3 ms, 1 ms, 3 ms, 10 ms and 30 ms.

Release
D Set the time it should take before the compressor lets the sound through unaffected - after the signal level
drops below the set Threshold.
If set to “Auto”, the Release time will be program adaptive, so the Release time is automatically increased following
long peaks and decreased following short peaks.
Range: 0.1 s, 0.3 s, 0.6 s, 1.2 s and “Auto”.

Make Up
Make-Up gain compensates for level reduction caused by compression and helps maintain a steady output level.
D Adjust the output gain from the Master Bus Compressor device.
Range: - 5 dB to + 15.00 dB

Compressor Gain Reduction Meter

The meter shows the gain reduction amount in dB applied by the compressor.

619 MASTER BUS COMPRESSOR

 External Sidechain

You can use external signals to trigger the Master Bus Compressor. This is done by connecting an external signal to
the Sidechain Inputs on the back of the Master Bus Compressor.

The Sidechain inputs.

When a cable is connected to the Sidechain Input(s) the “Connected” LED is lit on the front panel:

D Click the Sidechain button to “key” (trigger) Master Bus Compressor from the external signal instead of the
channel signal.
For example, you could use a kick drum pattern to trigger the sidechain function, to achieve a rhythmic pumping
effect.

620 MASTER BUS COMPRESSOR

 Connections

Comp Gain Reduction

This CV output sends out the gain reduction as a CV signal (see “Compressor Gain Reduction Meter”).

Sidechain Input Left & Right

D Patch an external audio signal to be used as sidechain signal here.
See “External Sidechain” for more information.

Input Left & Right

D Patch the audio signals you want to process here.
If your input signal is in mono, connect only to the L (left) input.

Output Left & Right

These are the audio outputs.

621 MASTER BUS COMPRESSOR

622 MASTER BUS COMPRESSOR
Chapter 36
The MClass Effects
 The MClass effects
The MClass effects package consists of four effect devices, which are available as separate devices on the Effects
palette in the Browser. There are also a number of Mastering Combi patches available in the Effects folder in the
Factory Sound Bank. These mastering patches consist of all four MClass device types, patched together and orga-
nized as Combinator patches. The Mastering Combis are perfect to use as Master Insert FX to process the final mix.
The individual MClass effects are as follows:
• The MClass Equalizer - this has low and high shelving bands, two fully parametric bands and a low cut “anti-
rumble” switch. See “The MClass Equalizer”.
• The MClass Stereo Imager - this can be used to control the stereo width for the high and low frequency ranges
separately. See “The MClass Stereo Imager”.
• The MClass Compressor - this single band compressor features sidechain input and program-adaptive release.
See “The MClass Compressor”.
• The MClass Maximizer is a special limiter tailored for loudness maximizing without risk of clipping. See “The
MClass Maximizer”.

Common effect device parameters

While the specific parameters for the MClass effect devices are described below, some features and procedures are
common to all effect devices. Please, refer to “Common effect device features” for information about the Input meter,
the Bypass/On/Off switch and Signal Flow Graphs on the effect devices.

624 THE MCLASS EFFECTS

 The MClass Equalizer

The MClass Equalizer consists of two independent, fully parametric bands plus high and low shelving bands and a lo
cut switch.
This is most often used as an insert effect, in mono or stereo.

Activating the separate EQ bands

The separate bands are organized as vertical strips on the panel, in the following order (from left to right): Lo Cut/Lo
Shelf/Parametric 1/Parametric 2/Hi Shelf.
You activate the separate bands by clicking the button at the top of each strip (none of the bands are activated by de-
fault).

Lo Cut
The Lo Cut switch will simply cut frequencies below 30 Hz (by 12 dB/Octave). This is useful for removing low fre-
quency “rumble”.
q When you are using the MClass Equalizer with a compressor or Maximizer, activating the Lo Cut switch pre-
vents subsonic sound from “topping” the compressor/limiter, and allows them to operate as efficiently as pos-
sible.

Parametric 1-2 parameters

A parametric equalizer will boost or cut frequencies around the selected frequency. The following parameters are
available for both the parametric bands:

| Parameter | Description
Frequency This determines the center frequency of the EQ, i.e. at which frequency the level should be de-
creased or increased. The range is 39 Hz to 20 kHz.
Gain Specifies how much the level of the selected frequency range should be boosted or cut. The gain
range is ±18 dB.
Q This governs the width of the affected area around the set center frequency. The higher the value,
the narrower the affected frequency range.

625 THE MCLASS EFFECTS

 Lo/Hi Shelf parameters
A shelving equalizer will boost or cut frequencies below or above the selected frequency.
The following parameters are available for the Lo/Hi Shelf bands:

| Parameter | Description
Frequency Frequencies below (Lo Shelf) or above (Hi Shelf) the selected frequency will be boosted or cut.
• The Lo Shelf range is 30 Hz to 600 Hz.
• The Hi Shelf range is 3 kHz to 12 kHz.
Gain Specifies how much the level should be boosted or cut. The gain range is ±18 dB.
Q This governs the slope of the shelving curve. The higher the value, the steeper the curve slope. High
Q settings will also produce a “bump” in the opposite cut/boost direction at the set frequency.

About the graphic display

The graphic display to the left in the device panel shows the frequency response curve as set by the EQ parameters.
This gives a visual feedback and helps you tailor the EQ settings.

The MClass Stereo Imager

The MClass Stereo Imager splits the signal into two frequency bands; “Hi” and “Lo” and allows you to widen or narrow
the stereo image of each band. A typical application of the Stereo Imager is to widen the higher frequencies and nar-
row the lower frequencies. This will make the bass end “tight” whilst “opening up” the higher frequencies.
This is most often used as an insert effect in stereo.
! The MClass Stereo Imager does not create stereo from mono input! For the device to work properly it must
connected with stereo in/out, and the input signal must contain a stereo audio signal.

Parameters
The following parameters are available:

| Parameter | Description
X-Over This determines the crossover frequency between the Hi and Lo band. Range is 100 Hz - 6 kHz. Frequen-
Frequency cies below this will be affected by the Lo Width setting; frequencies above will be affected by the Hi Width
setting.
Lo Width This adjusts the stereo width for the Lo band. Turn anti-clockwise to narrow the stereo width (i.e. to make
it more “mono”), and clockwise to widen the stereo image. Center position means no change from original
signal. The “Active” LED indicates whether Low Width is activated or not. Note that for the Lo band, it is
more common to narrow the stereo image, as the low frequency content in a mix is usually mixed center
and can become less defined if widened.
Hi Width This adjusts the stereo width for the Hi band. Turn anti-clockwise to narrow the stereo width (i.e. to make
it more “mono”), and clockwise to widen the stereo image. Center position means no change from original
signal. The “Active” LED indicates whether Hi Width is activated or not.
Solo switch This allows you to listen to the Lo and Hi bands separately, for reference purposes. “Normal” is the stan-
dard operating mode.

626 THE MCLASS EFFECTS

 Connections

Apart from standard L/R inputs and outputs, there are also “Separate” L/R outputs on the back panel. The Separate
outputs can either carry the Lo or Hi band output, which is set by the switch beside the outputs. These outputs can
be used to apply processing separately to either the Lo or Hi band.
q If you set the Solo switch to “Lo” and the Separate output switch to “Hi”, the device will operate as a basic
crossover filter, delivering the Lo band signal from the main output and the Hi band signal from the Separate
out.

The MClass Compressor

This is a single-band compressor capable of everything from subtle compression to aggressive pumping effects. Like
all dynamics processors it is best used as an insert effect.
The features include “soft-knee” compression for more musical and unobtrusive compression, program-adaptive re-
lease time and a sidechain input for de-essing and other dynamics processing. Additionally, you have a CV output, al-
lowing you to have the amount of gain reduction control other Reason parameters.

Parameters:
| Parameter | Description
Input Gain The Input Gain controls the ”drive” of the compression. This determines how much compression the signal will have in
conjunction with the Threshold. Range: ±12 dB.
Threshold This sets the level at which onset of compression occurs. When the input level is below the Threshold setting the sig-
nal is unaffected. When the input level exceeds the threshold, compression kicks in.
In practice, this means that the lower the Threshold setting (and the higher the Input Gain), the more compression will
be applied. Range: -36 dB to 0 dB
Soft Knee Normally signals above the threshold will be compressed immediately at whatever ratio is set. This can be very notice-
able, especially when using high compression ratios. When Soft Knee is activated, the onset of compression will be
more gradual, producing a less drastic result.
Ratio This lets you specify the amount of gain reduction applied to the signals above the set threshold. The Ratio can be set
from 1:1 (no reduction) to  (Infinite).
Gain meter This shows the amount of gain reduction (in dB).
Solo Sidechain This allows you to monitor the signal connected to the sidechain input (see below).
Attack This governs how quickly the compressor will apply its effect when signals rise above the set threshold. If you raise this
value, the response will be slower, allowing more of the signal to pass through the compressor unaffected. Typically,
this is used for preserving the attacks of the sounds. Range: 1ms to 100ms.
Release When the signal level drops below the set threshold, this determines how long it takes before the compressor lets the
sound through unaffected. Set this to short values for intense, “pumping” compressor effects, or to longer values for a
smoother change of the dynamics. Range: 50ms to 600ms.
Adapt When this is used, set Release to the time you want for short peaks - when longer peaks occur, the Release time is au-
Release tomatically increased.
Output Gain This controls the output gain and can be used to compensate for the gain reduction caused by compression. Range:
±12 dB.

627 THE MCLASS EFFECTS

 About the Sidechain inputs
D When a signal is connected to the Sidechain inputs, it is this signal that will trigger the compression.
This signal is not passed to the compressor output, and will thus not be heard (unless Sidechain Solo is activated).
D The signal connected to the normal L/R inputs is the signal that will be processed by the compressor (as
usual).
Here follows two examples of how you can use sidechain processing:

Example 1 - using the Sidechain inputs to create ducking effects

“Ducking” is when the level of one signal is reduced by the presence of another signal. A typical application is to au-
tomatically lower the level of a musical bed when a voice-over starts, and to automatically bring the level up when the
voice-over stops. To set this up, we can name the musical bed in the example “Device A”, and the voice-over “Device
B”. Proceed as follows:
1. Connect Device A and Device B to separate channels on a mixer device.
In this example the signal produced by Device A should be continuous, and the signal produced by Device B
should be intermittent, i.e. it should contain both silent passages and signal passages.
2. Select Device A and then select an MClass compressor from the Create menu.
The effect will be auto-routed as an insert effect to Device A.
3. Connect a Send on the mixer device to the Sidechain inputs on the MClass compressor.
As soon as anything is connected to the Sidechain inputs, the “Active” indicator lights up on the device panel. The
compressor will no longer react to the signal produced by Device A.

4. Turn up the corresponding AUX Send level for Device B on the mixer.
This means that the Device B signal now feeds both the mixer's input, and the sidechain input on the compressor,
which in turn triggers the gain reduction.
5. If you now start playback of both devices, the level of Device A will be lowered whenever Device B sounds, and
be raised again when Device B stops.
The amount of gain reduction, how quickly it lowers the level, and the time it take for the level to return to normal
again is determined by the corresponding Gain/Threshold/Ratio and Attack/Release parameters.

Example 2 - using the Sidechain inputs to create frequency sensitive compression

By inserting an equalized signal to the sidechain inputs you can make the compression more or less sensitive to a
certain frequency range. A typical application of this is “de-essing” - where harsh “S”-sounds in vocal material is re-
duced or eliminated.
Frequency sensitive compression is set up as follows:
1. Hold down [Shift] and create an instrument device.
Pressing [Shift] means no auto-routing connections to/from the device are made.
2. Hold down [Shift] and create a MClass Equalizer.
3. Hold down [Shift] and create a MClass Compressor.
4. Create a Spider Audio Merger and Splitter device.
5. Connect the outputs of the instrument device to the A and B inputs on the Spider.

628 THE MCLASS EFFECTS

 6. Route one pair of the split outputs of the Spider to the MClass Equalizer inputs.
7. Route the Equalizer outputs to the Sidechain inputs on the MClass Compressor.
8. Route another pair of the split outputs of the Spider to the MClass Compressor.
Now, the compressors normal audio inputs are fed the unprocessed signal, and the sidechain inputs are fed the
equalized signal.

9. Next, route the outputs of the Compressor to inputs on a mixer device.

10.Activate the Solo Sidechain button on the compressor’s front panel.
You will now only hear the equalized signal.
11.Now use the parameters on the MClass Equalizer to boost the frequencies you wish should trigger the gain re-
duction and cut the frequencies you wish to avoid triggering the gain reduction.
You can use rather extreme eq settings - the signal will not be heard when Solo Sidechain is deactivated anyway.
E.g. for de-essing you should separate and boost the offending “S” frequencies as much as possible.
12.Deactivate Solo Sidechain when you have finished tweaking the equalizer.
Now, the compressor will be more sensitive to the frequency area you tuned in with the equalizer, and thus react
more to these frequencies. Note, however, that the whole signal will still be compressed - not just the boosted fre-
quencies - so in case of de-essing you should usually use fast Attack and Release settings so that the gain reduc-
tion does not affect the rest of the program too much.

CV Outs
On the back of the MClass Compressor you can find a “Gain Reduction” CV out connector. This can be used to mod-
ulate other parameters with the amount of gain reduction applied by the compressor. This means that the compres-
sor works as an envelope follower. You could for example have the audio signal level control pan in a mixer or a synth
parameter.

629 THE MCLASS EFFECTS

 The MClass Maximizer

This is a loudness maximizer, a special type of limiter which can significantly raise the perceived loudness of a mix
without risk of hard clipping distortion. Features include a 4 ms look ahead function for “brick wall” limiting and a Soft
Clip function.
The MClass Maximizer should be used as an insert effect, and is designed to be placed at the end of the signal chain
between the mixed final output and the I/O device.
Parameters
| Parameter | Description
Input Gain The Input Gain sets the basic volume of a mix. If this is set very high, you should use Look
Ahead mode or the Soft Clip function to eliminate the risk of hard clipping distortion. Range:
±12 dB.
Limiter On/Off This turns the Limiter section on or off.
Look Ahead On/Off If activated, this will introduce a very short delay (4 ms) to the signal. This delay is used to de-
tect peaks in the signal before they actually occur. If high peaks are detected the limiter is
“ready for them” and gain reduction is applied to transparently control the peaks.
Attack This governs how quickly the Limiter will apply its effect. If set to Fast with Look Ahead acti-
(Fast/Mid/Slow) vated (and the Output Gain is set to 0 dB) you will get “brick wall” limiting - no signal peaks
over 0 dB will pass.
Release This determines how long it takes before the Limiter lets the sound through unaffected. If
(Fast/Slow/Auto) Auto is activated, the Release time will automatically adapt to the program material.
Output Gain This controls the output gain and should normally be set to 0 dB.
Soft Clip On/Off If this is activated, it also acts a 0 dB brick wall limiter but in a slightly different way. The signal
will be “soft-clipped” which adds a pleasant, warm sounding distortion to the signal. It can be
used simply to get this effect, or as a safeguard against hard clipping distortion if Look Ahead
with Mid or Slow attack settings are used (or if Look Ahead is deactivated).
Soft Clip Amount This controls the amount of soft-clipping distortion. Note that if Soft Clip is on but the
Amount is set to zero, the distortion will be like hard clipping, and thus less pleasing to the
ear.
Output level meter This is a more detailed meter than found on the mixer. You can switch the meter characteris-
(Peak/VU) tics between Peak (faster response to peaks) and VU mode (average levels).

630 THE MCLASS EFFECTS

Chapter 37
Neptune Pitch Adjuster
and Voice Synth
 Introduction

The Neptune Pitch Adjuster and Voice Synth device is a combined monophonic vocal pitch corrector, pitch shifter
and polyphonic voice synth. Neptune was designed with focus on high-quality vocal processing but can also be used
on other material. However, due to the signal characteristics of other types of audio (complex inharmonic instrument
sounds, polyphonic material etc.) the result of the pitch adjustments might not be what you would expect. Don’t hes-
itate to experiment, though!
Used as a pitch corrector, Neptune can automatically correct flat monophonic input signals and output corrected
notes in real-time. The pitch correction can be adjusted from totally transparent to hard, robot-like. The correction can
be controlled from predefined scales, from input MIDI notes or from a combination of both!
When used as a pitch shifter, Neptune performs overall pitch-shifting of incoming monophonic audio in real-time and
transposes the output to a defined value within a ±1 octave range.
The Voice Synth section in Neptune allows you to add additional voices - harmonies - to a monophonic vocal input
signal and to control the voices via MIDI - from sequencer notes or by playing on your MIDI master keyboard.
Neptune also features controls for adjusting formants and for adding pitch bend and vibrato to the processed signal.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Typical use cases

Neptune can be used in a variety of ways and situations. Here are some examples:
• Pitch correction of a vocal track
Neptune is used as an Insert Effect on the vocal audio track, with the Pitch Adjust’s Scale set to desired scale. Set
the Correction Speed parameter to a moderate value for transparent pitch correction - or set the Correction Speed
parameter to max value and the Preserve Expression parameter to minimum for a hard “robotic” character.
• Octave dub
Neptune is used as a Send Effect. The Pitch Adjust section is turned off and the Transpose section is activated.
The Semi parameter is set to +12 to transpose up 1 octave. The Send FX control on the mixer is used for balanc-
ing the level of the octave-shifted voice. The Formant Shift control is used for making the transposed voice sound
more natural (or unnatural, if that's desired).
• Create backing harmonies on a vocal track
Neptune is used as an Insert Effect on the vocal audio track and MIDI Input is routed to the Voice Synth via the
Neptune sequencer track. Play some notes on your MIDI master keyboard and use the Voice Synth breakout jacks
connected to a separate Mix Channel for further processing.
• Pitch shift drums (non-pitched input)
The Pitch Adjust section is turned off and the Transpose section is activated. The Transpose parameters are set to
desired values.

632 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Overview and basic concepts
Sections overview

The Neptune panel is divided into three main sections that are connected in series:
• The Pitch Adjust section.
Here is where you control the pitch correction settings, including scale, correction speed etc. Here is also where
you find the big display where you can view Input Pitch, Target Note and Catch Zone etc. See “Using pitch correc-
tion”. for more details.
• The Transpose section.
Here is where you control the pitch shifting parameters. See “Using pitch shifting (Transpose)” for details.
• The Formant section.
Here is where you can control and adjust formants. See “Using Formant control” for details.
These three main sections can be used separately, or in combinations, by clicking the ON/Off button of the respec-
tive section:

The first section in the signal chain is the Pitch Adjust section. To get a realistic result out of the pitch correction, you
will probably want to use it in combination with the Formant section - especially if you control the target pitch via MIDI
and the target pitch differs a lot from the input pitch.
The Transpose section can be used stand-alone when you only want to perform stiff pitch shifting of the input audio.
If you want to pitch-shift atonal audio, you should use the Transpose section stand-alone, or in combination with the
Formant section, for best result. You can also use the Transpose section together with the Pitch Adjust section for
transposing pitch corrected audio by a defined amount.
The Formant section can be used stand-alone if you only want to change the character of the input audio - without
affecting the pitch. Used on vocals or speech, the Shift knob lets you control the “gender” of the voice.
Besides these three main sections, there are also additional parameters for defining audio input characteristics (see
“Input signal type”), MIDI control (see “MIDI Input”) and output level control (see “The Output Mixer section”).

633 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 The display

The big display in the center of the Neptune panel shows the following information:
• The Input Pitch of the incoming audio signal.
The detected Input Pitch is displayed as a yellow vertical line above the keyboard.
• The Target Note to which Neptune corrects the output signal.
The Target Note is displayed as a green vertical line above selected notes on the keyboard.
• An orange horizontal line between the Input Pitch and the Target Note.
The orange line shows the distance and direction from Input Pitch to Target Note.
• The Catch Zones, i.e. the pitch “window” which determines to what Target Note to correct the Input Pitch.
The Catch Zones are displayed as red horizontal boxes above each selected key on the virtual keyboard. As soon
as a detected Input Pitch lies within a Catch Zone, the Catch Zone box above the corresponding Target Note
switches to green. See “Setting Catch Zone Size”.
• The virtual keyboard, where you can select your own notes or custom scale to correct the audio to.
See “Setting Root Key and Scale”.

Setting up for pitch processing

The most flexible method of performing pitch adjustments is to apply it to already recorded audio tracks in the se-
quencer. Doing so will give you total freedom to edit and change the pitch adjustment settings afterwards without
needing to re-record any audio.
To set up Neptune for pitch adjustments of recorded audio tracks, proceed as follows:
1. Create a Neptune device and use the Reason Rack Plugin instance as an insert effect on the audio track.
2. If you want to control the pitch adjustment via your MIDI keyboard, route MIDI to the Neptune device from your
DAW.
Now, you are all set for pitch adjustments of the audio track. For information about pitch correction, see “Using pitch
correction”, for information about pitch-shifting, see “Using pitch shifting (Transpose)”, and for info about creating ad-
ditional vocal harmonies, see “Using the Voice Synth”.

Using pitch correction

Pitch correction is great if you, for example, have a vocal track in your song that contains flat notes. Instead of having
to overdub the flat parts on the audio track, you can automatically adjust the pitches of the flat notes using Neptune.
You can also sing through the Neptune in real-time and get a beautifully pitched voice!
There are two basic ways of using pitch correction in Neptune:
• Automatic pitch correction using Scales, see “Using automatic pitch correction”.
• Manual pitch correction using MIDI note input, see “Using manual pitch correction”.
You can also combine the two methods if you like. First of all, though, you will need to make some basic settings:

634 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Basic settings for pitch correction
Proceed as follows to set up the panel parameters for pitch correction:
1. Set up the Neptune device as desired according to the descriptions in “Setting up for pitch processing”.
2. Make sure the “Pitch Adjust” button is activated and that the “Transpose” button is off.

• Whether the Formant button should be on or off depends on the situation.

For very small pitch corrections, you might want to leave this off; for more substantial pitch changes you will get
the most natural sound when this is activated.
3. Define the Input signal type by using the “Low Freq”, “Wide Vibrato” and “Live Mode” buttons:

• Activate “Low Freq” for low-frequency material such as a deep voice etc.
The “Low Freq” mode will make Neptune detect low frequency notes in a more precise way. Note that the latency
will become longer due to the fact that low frequencies have longer cycle times.
• If the input voice contains a lot of vibrato, it might be a good suggestion to activate the “Wide Vibrato” button.
If your input audio has a heavy vibrato, this can cause Neptune to detect the wrong pitches. The result can be wob-
bling notes, unwanted swoops and glides etc. Activating the Wide Vibrato button will make the pitch detection ig-
nore any vibrato in the input audio, eliminating the problems. Note however that the vibrato can still be retained in
the processed sound, by raising the Preserve Expression parameter (see “About the Preserve Expression param-
eter”).
• If you are going to use pitch correction by singing through Neptune in real-time, try activating the “Live Mode”
button.
This will reduce the latency of the pitch correction to a minimum, which might be preferable if you want to monitor
the corrected signal as you sing. Note, however, that the audio quality may be lower in Live Mode.

635 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 4. Make sure the "MIDI" button is set to "To Pitch Adjust".

This will enable the pitch correction to be controlled also via MIDI (see “Using manual pitch correction”).

Using automatic pitch correction

With the Root and Scale functions you can define what notes you want the incoming audio to be corrected to. The in-
put audio will then be automatically corrected to the defined scale, without further user interaction. With the Catch
Zone, Correction Speed and Preserve Expression parameters you can set the character of the adjusted signal.
q It’s also possible to manually override the automatic settings via MIDI at any time, see “Using manual pitch
correction”.

About the Scale Memory

There are four Scale Memory slots in the Pitch Adjust section. The purpose of Scale Memories is to allow for auto-
mation of different Root Key, Scale and Catch Zone settings. A selected Scale Memory slot automatically stores
which Root Key and Scale notes are active (see “Setting Root Key and Scale”), just like a Redrum stores what 16th
note buttons are selected in a pattern. The Scale Memory slots also store Catch Zone settings (see “Setting Catch
Zone Size”). By using automation of the Scale Memory slots from the Neptune sequencer track, you can easily switch
characteristics throughout the song. The Scale Memory settings are automatically saved with the document when
you save your Song.
! Before you perform any edits of the Root Key, Scale and/or Catch Zone parameters, make sure you have se-
lected a Scale Memory slot that you want to overwrite.

Setting Root Key and Scale

! Any Root Key and/or Scale settings changes you make are automatically stored in the currently selected Scale
Memory slot in real-time, see “About the Scale Memory”.
Define the Root Key and Scale parameters as follows:

636 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 1. Set the Root Key with the Root spin controls.
The key can be chosen between C and B in a one octave span, covering each key in the western 12-tone scale.

.
2. Set desired scale with the Scale spin controls.
The Scale parameter can be set to any of the following preset scales, as indicated in the Scale display:

C# D# F# G# A#
C D E F G A B
.
Chromatic

C# D# F# G# A#
C D E F G A B
.
Major with C as Root Key

C# D# F# G# A#
C D E F G A B
.
Natural Minor with C as Root Key

C# D# F# G# A#
C D E F G A B
.
Harmonic Minor with C as Root Key

C# D# F# G# A#
C D E F G A B
.
Dorian with C as Root Key

C# D# F# G# A#
C D E F G A B
.
Mixolydian with C as Root Key
! * When Chromatic is selected, the Root display will switch to show “--” since all notes in the 12-tone scale are
included and the root key is of no importance here.

637 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 D You can also define your own custom scale by clicking on the keys in the display.
Selected notes in your custom scale will be indicated with green LEDs:

A custom scale containing the notes D, G and A.

When you set your own scale both the Root Key and Scale displays will switch to show “--”.
• The set Key and Scale is automatically repeated for every octave throughout the entire note range.
! Note that this doesn't necessarily need to be an actual scale - sometimes you may only need to activate one or
two notes, passing the other notes through, uncorrected.

Setting Catch Zone Size

The Catch Zone Size parameter defines what pitches in each octave should be “caught” and adjusted towards the
closest Target Notes.
! Any Catch Zone Size settings changes you make are automatically stored in the currently selected Scale Mem-
ory slot in real-time, see “About the Scale Memory”.
The Catch Zones are shown as boxes above each selected note in the keyboard display:

The Catch Zones for the notes in a custom scale with notes D, G and A selected.
• The Catch Zone size is set with the Catch Zone Size knob.
The range is ±20 to ±600 cent, set in 20 cent steps. The range is always centered around the selected notes in
the scale and the default value is ±100 cents.
! Pitches outside or in-between Catch Zones are not caught and adjusted, but let through unprocessed.

638 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 • Catch Zones cannot overlap each other. If the Catch Zone Size is increased so that adjacent zones touch each
other, they will stop expanding in that direction and meet “half ways”.
The Catch Zone size in the picture below is set to ±150 cents. As you can see, the Catch Zones for notes G and
A have “collided” and therefore met half ways in the center of the G# note. If the Catch Zone size should be in-
creased further, the G Catch Zone would expand only downwards and the A Catch Zone only upwards.

The Catch Zones for the selected notes D, G and A in a custom scale.
• If the Catch Zones should extend on either side of the 12-note keyboard range, they will “wrap around”.
In the picture below, notes C and G are selected in a custom scale. The Catch Zone Size is set to ±250 cents.
Since the Catch Zone for the C note extends also to the left outside the display, the Catch Zone “wraps around” vi-
sually and continues from the B note down to the A# note. Since the Scale is repeated downwards and upwards
for every octave, the result of this setting is that the Catch Zone for the C note will cover the B and A# notes in ev-
ery octave.

The Catch Zones for the selected notes C and G in a custom scale, with the C note Catch Zone “wrapped around” to cover also notes
B and A#.

Setting Correction Speed

The Correction Speed parameter controls how fast the Input Pitch should be adjusted to the Target Note. The range
is from Slow (knob turned fully counter-clockwise) to Fast (knob turned fully clock-wise).
• For a natural, transparent correction, a setting around the 12 o’clock position is ideal in most situations.
• A very fast correction speed will create almost a “stepped” correction.
This is the setting of choice for creating the infamous “robot voice” effect known from numerous radio hits.
• A slow correction speed will make the pitch correction almost unnoticeable during fast passages in the music.
This is because the correction won’t have time to set in before new incoming pitches are detected.

639 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 About the Preserve Expression parameter

The Preserve Expression parameter controls how much vibrato in the input audio should be let through when you use
a fast Correction Speed setting (see “Setting Correction Speed”).
• With a minimum Preserve Expression value and a fast Correction Speed there will be almost no natural vibrato
left in the pitch corrected voice.
• With max Preserve Expression value and a fast Correction Speed the original vibrato is still preserved.
• With max Preserve Expression value and a slow Correction Speed the original audio is let through almost un-
affected.

About adding Pitch Bend and Vibrato

It’s also possible to add Pitch Bend and/or Vibrato via MIDI while using the automatic pitch correction:
D Select “To Pitch Adjust” in the MIDI Input section.

This allows for incoming MIDI Pitch Bend and Vibrato (Mod Wheel) data to control the output pitch and to add vi-
brato to the output signal.
• Clicking and moving the wheels on the panel will also generate pitch bend and vibrato.

Using manual pitch correction

In Neptune it’s possible to override the automatic (scale) pitch correction settings with monophonic MIDI note data at
any time. This makes it possible to control the desired output pitches either by playing live on a MIDI master keyboard
- or by playing back recorded MIDI notes from the Neptune track in the sequencer!
1. Add a Reason Rack Plugin instance and create a Neptune device.
2. To allow for MIDI note control of pitch correction, select “To Pitch Adjust” in the MIDI Input section:

• When you sing through Neptune and hold down a key on your MIDI master keyboard, the output pitch will cor-
respond to the held MIDI note.
As soon as any MIDI note is received by Neptune, the Root Key, Scale and Catch Zone settings will be temporarily
ignored. However, the Correction Speed (see “Setting Correction Speed”) and Preserve Expression parameters
(see “About the Preserve Expression parameter”) will still be active.

640 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 ! Note that MIDI control of the pitch is monophonic.
• When pitch correction is controlled via MIDI, the corrected audio signal will also respond to Pitch Bend and Vi-
brato modulation.
! If you transpose pitches several semitones compared to the input pitch, there could be strange formant effects
that may or may not be desirable. To make heavily pitched signals sound more natural, you can use the for-
mant correction function described in “Using Formant control”.
! Note that MIDI control of the pitch correction will only momentarily override the Root Key, Scale and Catch
Zone settings. As soon as there is no MIDI Note present, the automatic (scale) pitch correction settings will
take over again.
q It’s also possible to generate polyphonic voices (harmonies) via MIDI using the Voice Synth function, see “Us-
ing the Voice Synth”.

Using pitch shifting (Transpose)

Pitch shifting basically means altering the pitch of the input audio by a set factor - the transpose value - and output
the result. As opposed to pitch correction, pitch shifting can be made on any type of monophonic audio input signal;
it doesn’t even have to have a regular pitch (tone). Pitch shifting can be used to generate a number of cool effects, for
example:
• Voice doubling effects by using slight transposition of only 5-10 cents.
• Adding an additional voice one octave up or down from the original lead vocal.
• Creating dark spooky monster voices by transposing speech down heavily.
• Creating chipmunk effects by transposing speech up heavily.
The list goes on...
Use the Transpose function as follows:
1. Click the Transpose button to activate the Transpose section.

2. Adjust the transposition using the Semi and Cent spin controls in the Transpose section.
The Transpose range is ±12 semitones with a fine tuning range of ±50 cents.
D If you want to transpose pitch corrected signals, make sure the Pitch Adjust section is on and has the desired
parameter settings.
D If you want to transpose non-pitched signals, such as speech, switch off the Pitch Adjust section.

641 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Using Formant control
What are formants?
Formants can be described as a sonic “footprint” of an acoustic space. The practical effect of formants could be com-
pared to a multi-peak filter acting on the frequencies in a sound. An acoustic guitar, for example, has a body shape
which makes is sound the way it does. The same goes for a human vocal tract (throat and mouth cavity); every human
vocal tract has a unique “shape” which gives the voice its character. It is these shapes that produce formants. A big
difference between a vocal tract and a guitar body, though, is that the vocal tract changes shape as you sing different
vowels. This also means that the formants will change.
When you pitch shift a signal up or down, the formant “multi-peak filter” will move up or down with the signal fre-
quency (just like a traditional synthesizer filter would with full keyboard tracking activated). The result will be a signal
that not only is pitched but also changes character. In some situations this might be what you want, but when it
comes to pitch shifting vocals you will probably often want the pitch-shifted signal to sound like it’s sung by the same
person. Therefore, Neptune features a formant control function.
The Formant section in Neptune lets you control the formants so they don’t move along with the pitched signals.
Neptune continuously samples and analyses the input audio and determines both the pitch and the current formants
of the signal in real-time. The formants are then automatically applied to the pitch-adjusted output signal in real-time.
The picture below shows schematic examples of a 500 Hz signal, pitch-shifted down and up one octave respectively,
without and with formant correction. The formants are the peaks of the gray dotted lines in the graphs:

dB Input signal at 500 Hz (blue line) dB Input signal at 500 Hz (blue line)
60 with harmonics (red lines) 60
with harmonics (red lines)
40 40

20 20

0 0
0 1000 2000 3000 4000 f (Hz) 0 1000 2000 3000 4000 5000 6000 7000 8000 f (Hz)

dB Signal pitched-shifted down dB Signal pitched-shifted up

60 -1 octave to 250 Hz without 60 +1 octave to 1 kHz without
formant correction formant correction
40 40

20 20

0 0
0 1000 2000 3000 4000 f (Hz) 0 1000 2000 3000 4000 5000 6000 7000 8000 f (Hz)

dB Signal pitched-shifted down dB Signal pitched-shifted up

60 -1 octave to 250 Hz with 60 +1 octave to 1 KHz with
preserved formants. preserved formants.
40 40

20 20

0 0
0 1000 2000 3000 4000 f (Hz) 0 1000 2000 3000 4000 5000 6000 7000 8000 f (Hz)

Pitch-shifting without and with formant correction applied. The left column shows -1 octave pitch-shifting and the right column +1
octave.

642 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Using the Formant function
When the Formant section is not active, the formants will move along with the pitched signal. On vocals this could
generate unnatural “gender change” effects, especially on heavily pitched voices. If this is not the desired effect, you
should activate the Formant section:
1. Activate the Formant function by clicking the Formant button.

2. Adjust the Formant shift with the Shift knob.

The Shift range is ±1 octave.
• The default 12 o’clock position means that the formants will be “locked” to the input signal and won’t move
with the shifted pitches.
Used on vocals this will give the impression of the same person singing at the adjusted pitches. (Refer to the pic-
ture in the “What are formants?” section for graphical examples of this particular Shift setting.)
• Turned further counter-clockwise, the formants will be “locked” to the input signal and displaced downwards
relative to the input signal.
Used on vocals this will give the impression of a more deep male character voice.
• Turned further clockwise, the formants will be “locked” to the input signal and displaced upwards relative to
the input signal.
Used on vocals this will give the impression of a more soprano/female/child character voice.
! Note that Voice Synth has its own separate “built-in” formant correction and is not affected by the Formant
section settings.

Using the Voice Synth

The Voice Synth function lets you use Neptune as a harmony processor to generate polyphonic voices out of a
monophonic input signal. The application is similar to how you would use a vocoder, such as the BV512 Vocoder, for
example. However, using the Voice Synth in Neptune will produce completely transparent and natural sounding har-
monies with pitches controlled from MIDI Note data.
1. Add a Reason Rack Plugin instance and create a Neptune device.
2. Click the MIDI radio button to select “To Voice Synth” in the “MIDI” section. Also make sure the Voice Synth
volume slider is raised:

• When you sing through Neptune and hold down a note or a chord on your MIDI master keyboard, the output
pitch(es) will correspond to the held MIDI note(s).
• The Voice Synth harmonies will also respond to any Pitch Bend and Vibrato modulation.
! The parameters in the Pitch Adjust, Transpose and Formant sections are ignored by the Voice Synth.
q You can mix Pitched Signal with the Voice Synth signal in the Mixer section. If you only want the Voice Synth
sound, lower the Pitched Signal fader.

643 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Panel parameters
On the front panel of Neptune, you will find the following parameters and displays:

Level Meter and Bypass/On/Off switch

| Parameter | Description
Bypass/On/Off switch In Bypass mode, the input signal passes through unaffected to the main outputs of the device. The separate Voice
Synth outputs are automatically muted. In On mode, the device outputs the processed signal. If the Voice Synth is
used, its output signal are routed to the separate Voice Synth outputs. Off mode mutes the inputs, silencing the de-
vice.
Level meter Shows the input signal level.

Bend and Vibrato wheels

! Choose what the Bend and Vibrato wheels should control by selecting desired destination in the MIDI Input
section - see “MIDI Input”.

• The Pitch Bend wheel is used for bending the pitch of notes, much like bending the strings on a guitar or other
string instrument.
• The Vibrato wheel can be used for applying vibrato to pitch corrected signals or to the Voice Synth harmonies.
Most MIDI keyboards have Pitch Bend and Modulation controls. Use these to control pitch bend and vibrato, or use
the wheel controls on the panel by clicking and moving the mouse.

Bend Range
The Range parameter sets the maximum amount of pitch bend, i.e. how much it is possible to change the pitch by
turning the wheel fully up or down. The maximum range is ±12 semitones (±1 octave). You change the value by
clicking the spin controls above of the display.

Vibrato Rate
Use the Rate knob to set the rate of the vibrato controlled by the Vibrato wheel.

644 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Input signal type

Here, you define the characteristics of the input signals you use.

Low Freq
Activating Low Freq mode will ensure more accurate tracking of low-frequency audio signals, such as bass voices.
Low Freq mode is suitable for input frequencies below the note F1, which is approximately 44 Hz. The detection in
Low Freq mode will work down to approximately 22 Hz which corresponds to note F0.
! Note that the pitch detection latency will be somewhat longer in Low Freq mode due to the longer cycles times
of low frequency signals.

Wide Vibrato
If your input audio has a heavy vibrato, this can cause Neptune to detect the wrong pitches. The result can be wob-
bling notes, unwanted swoops and glides etc. Activating the Wide Vibrato button will make the pitch detection ignore
any vibrato in the input audio, eliminating the problems.
Note however that the vibrato can still be retained in the processed sound, by raising the Preserve Expression param-
eter (see “About the Preserve Expression parameter”).

Live Mode
This is an ultra-fast tracking mode, perfect for when you want to monitor your pitch adjusted signal in real-time. How-
ever, we recommend that you turn this off when it's time to play back and mix your recordings, for the highest audio
quality.

MIDI Input

The MIDI Input section features a radio button which allows you to route incoming MIDI to either of the following des-
tinations:

Pitch Adjust
Routes incoming MIDI Note data to the Pitch Adjust section for manual control of pitch correction, see “Using manual
pitch correction”. Pitch Bend and Vibrato (Mod Wheel) MIDI data will also be routed to the Pitch Adjust section.

645 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Voice Synth
Routes incoming MIDI Note data to the Voice Synth for generating additional harmony voices, see “Using the Voice
Synth”. Pitch Bend and Vibrato (Mod Wheel) MIDI data will also be routed to the Voice Synth.

Pitch Adjust section

Pitch Adjust section On/Off

Click the Pitch Adjust button to activate/deactivate the Pitch Adjust section.

Root, Scale and Keyboard

Set the Root key and Scale with the spin controls. You can also create your own custom scales by selecting individual
notes in the Keyboard display. See “Setting Root Key and Scale” for more details on how to use these functions.

Scale Memory
The purpose of Scale Memories is to allow for automation of different Root Key, Scale and Catch Zone settings. A
selected Scale Memory slot automatically stores which Root Key and Scale notes are active. The Scale Memory slots
also store Catch Zone settings.
The edits you make of the Root, Scale and Catch Zone parameters are automatically stored in the selected Scale
Memory slot. The settings of the Scale Memory are saved together with the rest of the Song data. Refer to “About the
Scale Memory” for details on how to use the Scale Memory slots.

Catch Zone Size

With the Catch Zone Size slider you set the range of the input pitches to be corrected to specific notes. Refer to “Set-
ting Catch Zone Size” for details on how to use this function.

Correction Speed
Set the time is should take to adjust the pitch to the set scale. See “Setting Correction Speed” for more details.

Preserve Expression
Set how much vibrato in the input audio should be let through when you use a fast Correction Speed setting. See
“About the Preserve Expression parameter” for more details.

646 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Transpose section

Click the Transpose button to activate the Transpose section. When active, the output pitch will be transposed ac-
cording the settings of the Semi and Cent parameters. The Transpose function can be used either on pitch corrected
signals (with the Pitch Adjust section active) or on non pitch corrected signals (with the Pitch Adjust section deacti-
vated).

Semi and Cent

Set desired transposition value, in semitones and cents, with the spin controls. The range is ±12 semitones and ±50
cents.
See “Using pitch shifting (Transpose)” for more details.

Formant section

Click the Formant button to activate the formant control function. When active, the formants of the input signal will be
preserved and won’t move with the adjusted output pitches.
! The Formant section settings have effect on the Pitch Adjust section and on the Transpose section, if they are
active. However, the Formant section can also be used stand-alone to only displace the formants of the input
signal.

Shift
Use the Shift knob to displace the formants within an range of ±1 octave.
See “What are formants?” and “Using the Formant function” for more details.

The Output Mixer section

The Pitch Adjust and Voice Synth parameters control the output from the Neptune in the following way:

647 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Pitched Signal
This slider controls the total output volume of the Pitch Adjust, Transpose and Formant sections routed to the Left
and Right outputs (see “Audio Out”).

Voice Synth
This slider controls the output volume of the Voice Synth routed either to the main Left and Right outputs or to the
separate Voice Synth Left and Right outputs if these are connected (see “Voice Synth Out”).
! Note that the Voice Synth function must be activated, otherwise this control will have no effect - see “Using the
Voice Synth”.

Connections
Flipping the Neptune around reveals an array of connection possibilities. Some of these are CV (control signal) re-
lated and some are audio signal related.

Sequencer Control
Note
The Note input allows you to control the pitch of either the Pitch Adjust or Voice Synth (depending on what is cur-
rently selected in the MIDI Input section - see “MIDI Input”). The pitch could be controlled from a Note CV output of a
Matrix or an RPG-8, for example.
! A Gate signal on the Gate input (see below) must also be present for the Note Input to work.

Gate
The Gate input should be used in combination with the Note modulation input (see above). As soon as a gate signal
is present, any note modulation on the Note input will be activated. The Gate could be controlled from a Gate CV out-
put of a Matrix or an RPG-8, for example.

CV In
These control voltage (CV) inputs can be used for modulating various Neptune parameters from other devices. The
inputs control the following parameters:

648 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

 Bend
The Pitch Bend input allows you to control pitch bend of either the Pitch Adjust or Voice Synth (depending on what
is currently selected in the MIDI Input section - see “MIDI Input”). The pitch bend could be controlled from a Pitch
Bend CV output of an RPG-8, for example.

Vibrato
The Mod Wheel input allows you to control the Vibrato amount of either the Pitch Adjust section or Voice Synth (de-
pending on what is currently selected in the MIDI Input section - see “MIDI Input”). The Vibrato could be controlled
from a Mod Wheel CV output of an RPG-8, for example.

Formant
The Formant Shift input allows you to control the Shift parameter in the Formant section from a CV source on an-
other device. The Formant Shift parameter accepts bipolar control signals (-63 to +64).

CV Out
These control voltage (CV) outputs can be used for modulating other device parameters from Neptune:

Pitch
The Pitch output allows you to control the pitch of other devices from the Pitch Adjust section, either directly or via
the Transpose section (depending on if Transpose is active or not). The Pitch CV corresponds to the pitch of the pitch
adjusted and transposed signal. The Pitch output could be connected to the OSC Pitch CV input of another synth de-
vice, for example.

Amplitude
Neptune features an internal envelope follower. The Amplitude output sends out a control signal from this envelope
follower based on the audio input level to Neptune. The Amplitude output could be connected to the Master Volume
or Level CV input of a synth device, for example.

Audio In
Route your audio input signal(s) to the Left (and Right) audio input(s) to the right on the rear panel.
! If you want to use Neptune in mono, connect only to the Left input.

Voice Synth Out

Connect to these outputs if you want to break out the audio from the Voice Synth as separate signals and, for exam-
ple, route to a separate Mix Channel for separate processing.
! When these are connected, the Voice Synth audio will be removed from the main Audio Out outputs.

Audio Out
The Left and Right outputs are the main stereo outputs of Neptune. Here, the audio from the Pitch Adjust, Transpose
and Formant sections are routed. If the Voice Synth Sep Output (see “Voice Synth Out”) are not connected, the Voice
Synth audio output are also routed to the main outputs.
! If you want to use Neptune in mono, connect only to the Left output.

649 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH

650 NEPTUNE PITCH ADJUSTER AND VOICE SYNTH
Chapter 38
Pulveriser
 Introduction

Pulveriser is a very versatile stereo in/out compression+distortion+filter device, capable of mangling any sound liter-
ally beyond recognition, but also capable of producing more subtle musical effects. Pulveriser features a wonderful
compressor, coupled with a nice warm distortion, plus a multi-mode filter. The different sections of Pulveriser can be
modulated by an LFO and by an Envelope Follower to allow for really organic modulation effects. Pulveriser also fea-
tures a Dry/Wet mix control so you can utilize parallel processing in the unit itself, i.e. mix in the processed signal in
parallel with the dry signal - great for parallel compression of drum loops etc.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

About the Patch format

Pulveriser features programmable effect presets. Included are a number of factory Patches which can be used as
they are or provide you with a good starting point for further tweaking.
Patches use the file extension “*.pulver”. Loading and saving Patches is done in the same way as for instrument de-
vices.

Parameters
Pulveriser contains three main effect sections; Squash (compression), Dirt (distortion) and Filter (multi-mode filter
with cutoff and resonance controls). In addition, Pulveriser features two modulation sections - an LFO with selectable
waveforms and an Envelope Follower which can modulate the effect sections. The parameters in each section are as
follows:

Common effect device parameters

While the specific parameters for the Pulveriser effect device are described below, some features and procedures
are common to all effect devices. Please, refer to “Common effect device features” for information about the Bypass/
On/Off switch.
Note, however, that the Input Meter is different on the Pulveriser device. Instead of the standard multi-segment LED
indicator found in other Reason effect devices, Pulveriser features a red lamp which reflects the intensity of the input
signal.

652 PULVERISER
 Signal Routing selector

With the Signal Routing selector to the bottom left on the front panel you determine the signal flow through Pul-
veriser:
• Squash > Dirt > Filter
This setting puts the compressor before the distortion and filter sections in series:

Input signal Output

signal

Dry
signal
Mixed
signal

Wet
signal
In this configuration the compressor and distortion affect the entire unfiltered input signal.
• Filter > Squash > Dirt
This setting puts the filter before the compressor and distortion sections in series:

Input signal Output

signal

Dry
signal
Mixed
signal

Wet
signal
In this configuration the compressor and distortion affect the filtered input signal. This way you can define what
frequencies should enter the compressor and distortion sections.

653 PULVERISER
 The Squash section

The Squash section is where you set the compression of the signal - from subtle compression to heavily “pumping”
effects.

Squash
The Squash knob affects compression ratio, threshold and make up gain in a nice and musical way.

Release
The Release knob controls the compressor release time. If you set this to a fairly low value and have the Squash
amount fairly high, you will get that nice “pumping” compression effect.

The Dirt section

The Dirt section is where you can add distortion to the signal - from gentle to screaming outrage.

Dirt
The Dirt knob controls the level of distortion.

Tone
The Tone knob controls a lowpass filter. Turned fully clockwise the filter is completely open, letting all frequencies
through.

654 PULVERISER
 The Filter section

The Filter section features five different filter types. Each filter type has controls for Frequency and Peak (resonance
amount).

Filter selector
Select one of the following different filter types (or bypass):
• Bypass
This will let the input signal through the Filter section unaffected.
• Lowpass 24
This is a lowpass filter with a slope of 24dB/octave, similar to the LP24 lowpass filter in the Subtractor (see “24
dB Lowpass (LP 24)” in the Subtractor chapter).
• LP12+Notch
This is a lowpass filter with a slope of 12dB/octave, in series with a notch filter. The cutoff frequency of the low-
pass filter and the center frequency of the notch filter are the same. Changing the Frequency control will generate
sort of an “animated” effect.
• Band Pass
This is a bandpass filter similar to the BP12 filter in the Subtractor (see “Bandpass (BP 12)” in the Subtractor
chapter.
• High Pass
This is a highpass filter with a slope of 12dB/octave, similar to the HP12 filter in the Subtractor (see “Highpass
(HP12)” in the Subtractor chapter.
• Comb
This is a comb filter similar to the “Comb +” Filter in Malström (see “Comb + & Comb –” in the Malström chapter).

Frequency
The Frequency knob controls the cutoff frequency or center frequency depending on selected filter type.

Peak
The Peak knob controls the resonance amount.

655 PULVERISER
 The Tremor section

The Tremor section in Pulveriser is an LFO which can modulate the Filter Frequency parameter and/or the main out-
put Volume parameter. The Tremor section features the following parameters:

Rate
Controls the LFO rate. The Rate range in Pulveriser is very wide and can reach way up in the audio frequency range.
The rate can also be synced to the sequencer tempo by clicking the Sync button to the right (see “Sync” below). In
sync mode, the Rate knob controls the sync resolution.
As a special feature, the rate can also be modulated from the envelope follower in the Follower section, see “The Fol-
lower section”.

Waveform selector
Select one of nine LFO waveforms. Apart from standard waveforms (sine, triangle, square etc.) there are random,
non-linear and stepped waveforms. The shape of the waveforms are shown in the display and reflect how the modu-
lated signal is affected.

Sync
Click the Sync button to synchronize the LFO Rate to the main sequencer tempo. In Sync mode the LFO cycle is also
synced to the sequencer, which means that the LFO cycle chases the current sequencer position.
Control the sync resolution with the Rate knob, see “Rate” above.

Spread
The Spread button introduces a stereo effect by modulating the left and right audio channels with two LFO signals
phase shifted 180 degrees in relation to each other. This means that you are able to generate kind of a “roto-
speaker” effect to the processed signal.

Lag
The Lag control acts like a lowpass filter on the LFO signal, making the LFO signal smoother. This is especially no-
ticeable on waveforms with sharp edges or transients like the square, sawtooth and stepped waves. On the sinewave
you will barely notice any effect since it’s already “smooth” by nature.

Modulation Amount knobs

On either side of the Tremor section are two bipolar modulation amount knobs, with zero modulation at the 12 o’clock
position. Since the modulation parameters are bipolar it means that if the knob is in the “-” sector it will invert the LFO
wave shape. This is especially useful in LFO Sync mode where you can define the “direction” of the modulation in re-
lation to the sequencer playhead.
D Turn the left knob to control the modulation of the Frequency parameter of the Filter section.
D Turn the right knob to control the modulation amount of the wet signal of the Volume parameter.
This will introduce a tremolo effect to the Wet signal fed via the Volume control.
! Note that the Dry signal in the Blend section is never affected by the tremolo effect! This means that if the
Blend knob is set to fully Dry, there won’t be any tremolo effect of the volume.

656 PULVERISER
 The Follower section

The Follower section features an envelope follower which analyzes the amplitude of the incoming signal and outputs
a modulation (CV) signal that corresponds to the incoming audio level. The modulation signal can then control the
Frequency parameter in the Filter section and/or the LFO Rate parameter in the Tremor section. The Follower sec-
tion features the following parameters:

Trig
Click/hold the Trig button to manually trig/gate the envelope follower. Clicking/holding the Trig button will make the
envelope follower output a modulation signal according to the settings of the Attack and Release parameters de-
scribed below. If you hold the Trig button for a longer period than the Attack time, the Follower will output maximum
CV signal level. When you then release the Trig button, the CV signal level will drop according to the Release time
and continue to follow the audio input signal level instead.

Threshold
This defines at which input signal level the envelope follower should trig. Set to a low value, the envelope follower will
react as soon as there is any audio signal present on the Pulveriser inputs. Set to a high value, the envelope follower
will react only on loud input signals, or from a manual Trig signal.
The red lamp to the right of the Threshold knob gives a visual indication of the CV signal level.
• On the back of Pulveriser you will find a Follower CV output - this delivers the CV signal from the envelope fol-
lower, allowing you to dynamically control parameters in other devices.

Attack
This controls how fast the envelope follower should react after the input signal has reached above the Threshold
value. Note that the attack time can only be increased compared to the input signal - never shortened.

Release
This controls how fast the envelope follower CV signal should drop to zero after the input signal has decreased below
the Threshold value. Note that the release time can only be increased - never shortened.

Modulation Amount knobs

To the left of the Follower section are two bipolar modulation amount knobs.
D Turn the upper left knob to control the modulation of the Rate parameter of the Tremor section.
If the modulation knob is in the “+” sector the rate will increase according to increased audio level. If the modula-
tion knob is in the “-” sector the rate will decrease according to increased audio level.
If Sync is enabled in the Tremor section, the rate will jump between the different resolutions according to the Fol-
lower modulation amount.
! Modulating the Tremor Rate in Sync mode can produce noise when the Rate parameter switches between res-
olutions. This noise can be eliminated by increasing the Lag amount in the Tremor section, see “Lag”.

657 PULVERISER
 D Turn the lower left knob to control the modulation amount of the Frequency parameter of the Filter section.
If the modulation knob is in the “+” sector the Filter Frequency will raise according to increased audio level. If the
modulation knob is in the “-” sector Filter Frequency will drop according to increased audio level.

Blend

With the Blend knob you control the mix between the dry and wet signal. With the knob set somewhere in between
the Dry and Wet position you will have parallel processing. This can be useful if you, for example, want to process a
drum loop with compression (Squash) and distortion (Dirt) and mix the processed signal with the dry before sending
it to the outputs.

Volume

With the Volume knob you set the total output level of the dry+wet signals.

658 PULVERISER
 Modulation inputs and outputs

CV Modulation inputs
On the back of Pulveriser you will find CV inputs for controlling the following parameters:

Squash
Use this for dynamically changing the amount of compression in the Squash section.

Dirt
Use this for dynamically changing the amount of distortion in the Dirt section.

Filter Frequency
Use this for dynamically changing the Frequency parameter in the Filter section.

Tremor Rate
Use this for dynamically changing the LFO Rate parameter in the Tremor section.
If the LFO is in Sync mode, the rate will jump between the different resolutions according to the CV modulation input
signal amount.

Volume
Use this for dynamically changing the output volume from Pulveriser.

Follower
Use this for controlling the envelope follower signal from an external source. The internal envelope follower signal is
replaced with the CV signal that is inserted here.
! Note that the Attack and Release controls can still be used to shape the CV input signal.

Audio Modulation inputs

! Note that these modulation inputs accept audio rate signals, which is really cool!

659 PULVERISER
 Filter Frequency
Use this for dynamically changing the Frequency parameter from an external audio signal. The result of this modula-
tion is Filter FM.

Volume
Use this for dynamically changing the output volume from an external audio signal. The result of this modulation is
amplitude modulation (AM) of the Pulveriser output signal.

CV Modulation outputs
Follower
On this output the control signal from the Follower is present.

Tremor
Here, the LFO CV signal from the Tremor section is present.

Demolition tips and tricks

Don’t restrict yourself to using Pulveriser as a basic compressor or distortion unit - you may be surprised to find how
often Pulveriser can add power, warmth, color and “movement” to your sounds. Here are some examples:

Beef up your sounds

Use Pulveriser to fatten up your sounds:
1. Select the Low Pass 24 filter type in the Filter section.
2. Set the Frequency parameter to a low value so you can barely hear the lowest frequencies.
3. Set the Threshold, Attack and Release knobs in the Follower section to minimum values.
4. Slowly increase the Follower > Filter Amount knob until you can hear the lower frequencies of the sound.
5. Adjust the Blend knob until you are satisfied with the mix.
6. Adjust the Blend, Squash and Dirt knobs to fine-tune the sound.

Make your pads tremble

Use Pulveriser to create movements in your pad sounds:
1. Select the Comb filter type in the Filter section.
2. Set the Peak knob in the Filter section to a fairly low value.
3. Turn the Tremor > Filter Modulation amount knob past the 12 o’clock position.
4. Set the Tremor Rate knob to around the 10 o’clock position.
5. Set the Threshold, Attack and Release knobs in the Follower section to minimum values.
6. Set the Follower > Tremor Amount knob to around the 2 o’clock position.
7. Set the Tremor > Volume knob to around the 3 o’clock position.
8. Play a pad sound which has long Attack, Decay and Release times and a fairly low Sustain value on its amp en-
velope.
Notice how the Tremor Rate and intensity change and modulate the filter and volume according to the pad volume.

660 PULVERISER
Chapter 39
Quartet
Chorus Ensemble
 Introduction

Quartet Chorus Ensemble is a fabulous sounding chorus device, with four different characteristic chorus/ensemble
algorithms. Each of the four algorithms can have their own unique parameter settings - including the Dry/Wet
parameter - so you could switch between the algorithms and get the exact result you are looking for.
Quartet is designed to be used mainly as an insert effect, for spicing up individual instrument sounds with nice dense
choruses and modulations.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel reference
Global controls
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Routing

D Click the Routing selector to select “Stereo” or Dual Mono” from the pop-up menu.
Stereo: With this selected the L+R input signals are mixed before being sent into the stereo effect. This means
you can connect a mono input signal and get stereo output signals.
Dual Mono: The L+R input channels are processed independently.

Width

The Width control can be used for setting the stereo width - from mono to nice and wide stereo.
! Note that the Width parameter can be set individually for each of the four chorus algorithms.

662 QUARTET CHORUS ENSEMBLE

 Dry/Wet

This controls the mix of the dry and processed signals.

! Note that the Dry/Wet parameter can be set individually for each of the four chorus algorithms.
! Note that all chorus/ensemble effects require some amount of dry signal to produce the desired effect. There-
fore, 100% Wet also includes a certain amount of dry signal.
q If you are using Quartet as a send effect you would probably want to have the Dry/Wet knob set to 100%.

Chorus

The Chorus effect algorithm simulates multiple detuned “copies” of the input signal. The Chorus is basically a delay
line with adjustable feedback. The principle is to split the input signal in two, run one signal dry and the other through
the delay line, and then sum the two signals.
The picture below shows the basic principle of the chorus:

Feedback
+ Output signal
Delay Line
Input signal

: audio signal
: control signal
Delay Mod Mod
Depth Rate

Amplitude

Frequency
Delay (log)

663 QUARTET CHORUS ENSEMBLE

 Delay

Here you set the delay time between the dry and processed signals. In practice, this determines where the notches/
peaks will appear in the frequency spectrum.
Range: 1.00-30.00 ms

Mod Depth

This determines the depth of the LFO modulation, i.e. by how much the delay time should be modulated. If you set this
to 0, the delay time will be static (most effective if you add some feedback).

Mod Rate

This determines the frequency of the LFO modulating the delay time. The higher the value, the faster the sound will
oscillate.
Range: 0.10-5.00 Hz

Feedback

This governs the amount of effect signal fed back to the input, which in turn affects the intensity and character of the
chorus effect. Turning this towards 100% produces a flanger type of effect with a pronounced resonance “tone”,
while keeping it around 50% produces a more gentle chorus effect.

664 QUARTET CHORUS ENSEMBLE

 BBD

The BBD is a bucket brigade delay line which simulates vintage ensemble effects. Historically, the bucket brigade
delay line was built up by a series of (analog) capacitors, that were clocked to consecutively transmit signals, via one
capacitor at a time, thus creating a delayed signal. The BBD algorithm in Quartet features three chorus effects in par-
allel, and therefore provides a much richer and denser effect than the Chorus algorithm.
The picture below shows the basic principle of the BBD algorithm:

Delay Line

Delay Line
+ Output signal

Delay Line
Input signal

: audio signal
: control signal
Delay Mod Mod Noise
Depth Rate Mod

Amplitude

Frequency
Delay (log)

Delay

Here you set the delay time between the dry and processed signals. The delay is preset scaled between the three de-
lay lines. In practice, this determines where the notches/peaks will appear in the frequency spectrum.
Range: 1.00-30.00 ms

665 QUARTET CHORUS ENSEMBLE

 Mod Depth

This determines the depth of the LFO modulation, i.e. by how much the delay time should be modulated. If you set this
to 0, the delay time will be static (unless you are using Noise Mod, see “Noise Mod”).
! If Mod Depth and Noise Mod (see “Noise Mod”) are both set to 0, the Width control (see “Width”) has no ef-
fect.

Mod Rate

This determines the frequency of the LFO modulating the delay time.
Range: 0.20-10.00 Hz

Noise Mod

This amplitude-modulates the signal with lowpass filtered noise, and generates a kind of “sparkling” effect.
! If Noise Mod and Mod Depth (see “Mod Depth”) are both set to 0, the Width control (see “Width”) has no ef-
fect.

FFT

The FFT algorithm simulates a type of chorus/ensemble effect by utilizing noise modulation of the signal partials.
First the signal is analyzed using FFT (Fast Fourier Transform) and converted to a representation in the frequency
domain. Then, the partials are modulated by noise to achieve a very nice and dense ensemble effect.

FFT Size

This sets the accuracy (and speed) of the frequency analysis. “1” is the fastest detection and preserves transients in
the signal - but this also leaves out detection of low frequencies. “4” is the most accurate detection. However, it’s also
slower since it also detects low-frequency material (which takes a little longer to detect).

666 QUARTET CHORUS ENSEMBLE

 Mod Depth

This determines the depth of the noise modulation of the signal’s partials. The parameter controls a combination of
noise amplitude and bandwidth. The result also depends on the Frequency Range parameter (see “Frequency
Range”).
The picture below shows how the Mod Depth parameter affects the partials at full Frequency Range:

Amplitude

Mod D Mod D Mod D Mod D

Frequency
(log)
Frequency Range

Frequency Range

The Frequency Range parameter determines which part of the frequency range should be noise-modulated and
which part should be left unaffected.
D Set the desired Frequency Range by dragging either “handle” sideways.

D To move the Frequency Range while maintaining the currently set bandwidth, drag the area between the “han-
dles” sideways.

667 QUARTET CHORUS ENSEMBLE

 The picture below shows how the Mod Depth parameter affects the partials in the signal at two different Frequency
Range settings:

Amplitude Amplitude

Mod D Mod D Mod D Mod D Mod D Mod D

Frequency Frequency
(log) (log)
Frequency Range Frequency Range

The first example shows the modulation of the partials at full bandwidth. The second example shows the partial mod-
ulation with the lower Frequency set to a higher value. In the second example, only the upper partials are modulated.
The lower partials are left unaffected.

Grain

The Grain algorithm generates an ensemble effect by “extracting” grains from the input signal in real-time and then
cross-fading through the grains in various ways. The method is similar to the “Long Grains” algorithm used in the
Grain Sample Manipulator device in Reason. The picture below shows the principle for the Grain algorithm:
Level

Input signal
Time

5 “extracted” grains

Level
The resulting signal is generated by
appending and crossfading the grains.

Time

An example of a signal generated from 5 grains of the input signal.

668 QUARTET CHORUS ENSEMBLE

 Phase

The Random Phase function randomly alters the phase of the grains to create a “bubbly” kind of effect, caused by
phase cancellation. This is especially noticeable when the Jitter parameter (see “Jitter”) is set to a low value.

Size

This controls the grain length. High values produce a more smooth effect, whereas low values generate more of a
“stuttering” effect.

Mod Depth

This randomly changes the initial pitch of the grains.

Jitter

The Jitter function modulates the grain playback position randomly. The Jitter function can be great for generating
chorus-like effects and to make a sound more “alive”, depending on the other settings.

Density

The Density function is a combination of grain size, playback rate and the amount of grain overlap. High values pro-
duce a really fat and dense chorus/ensemble effect, whereas low values generate a “thinner” effect.

669 QUARTET CHORUS ENSEMBLE

 Connections

CV Input
Mod Depth
This CV input can be used for modulating the Mod Depth parameter in the different algorithms. The input accepts
bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.
! Note that the CV Modulation is global for the Mod Depth control in all algorithms.

Width
This CV input can be used for modulating the Width parameter in the different algorithms. The input accepts bipolar
control signals. The input signal can be attenuated with the corresponding attenuation knob.
! Note that the CV Modulation is global for the Width control in all algorithms.

Dry/Wet
This CV input can be used for modulating the Dry/Wet parameter in the different algorithms. The input accepts
bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.
! Note that the CV Modulation is global for the Dry/Wet control in all algorithms.

Input Left & Right

D Patch the audio signals you want to process here.
If your input signal is in mono, connect only to the L (left) input.
! Note that the result also depends on the current Routing setting (see “Routing”).

Output Left & Right

These are the audio outputs.
! Note that the result also depends on the current Routing setting (see “Routing”).

670 QUARTET CHORUS ENSEMBLE

Chapter 40
Ripley Space Delay
 Introduction

Ripley Space Delay combines delay and reverb with powerful sound shaping and deep modulation for delay effects
with character.
At its heart Ripley Space Delay is a stereo delay with a unique band filter in the feedback loop, a lush reverb, and
several effects (including noise, drive, sample rate & bit reduction, and EQ) for coloring your sound. Combined in
different ways you can get everything from clean and digital taps and warm analog delay to crazy lo-fi chirps and
washed out ambiances. Coupled with extensive modulation options, Ripley Space Delay is a delay design playground.

Panel overview
The Ripley Space Delay front panel contains the following sections:

1 3 4

5 6 7 8

10
2 9

11

The Ripley Space Delay front panel sections.

672 RIPLEY SPACE DELAY

 • 1. Delay section.
• 2. Feedback Filter section.
• 3. Reverb section.
• 4. Patch selector (for browsing, loading and saving patches).
• 5. Noise section.
• 6. Distortion section.
• 7. Bitcrusher section.
• 8. Output section.
• 9. EQ section.
• 10. Ducking section.
• 11. Modulation and Modulation Matrix sections.

Signal flow
The picture below shows the signal flow in Ripley Space Delay:

Amount Decay Size Width Low Gain High Gain Amount

Space Out
SPACE DIST DIGITAL EQ OUTPUT
Output Out
DUCKER

Serial/Parallel
Time Time(2) Multi Offset Dry/Wet Tone Drive Dry/Wet Rate Bit Freq Gain Q Dry/Wet Gain

DELAY Send+Return jacks

In Input Delay Out

EXTERNAL FEEDBACK
FB Wobble Pan Width LOOP (Optional)

Lo Cut Hi Cut

Feedback

Amount
FEEDBACK NOISE
FILTER Character
: routing/patch point
: initial placement
(x8) : audio signal
Freq Shift Offset Filter 1-8 Hi End Type : control signal

Ripley Space Delay signal flowchart.

• The green patch points in the picture above indicate where the routable sections (Noise, Dist, Digital, EQ, and
Ducker) could be routed.

673 RIPLEY SPACE DELAY

 Panel reference
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

The Delay section

The Delay section features parameters relating to delay time, tempo sync, feedback, wobble and stereo behavior.

Sync, Keep Pitch and Dual Delay buttons

D Click the Sync button to enable Tempo Sync, which affects the “Delay Time” parameter(s).
D Click the Keep Pitch button to eliminate pitch changes when the delay times are changed.
Note that the pitch will change if you are changing Multiply values, though (see “Multiply”)
D Click the Dual Delay button to get separate delay times for the Left and Right channels (see below).

Delay Time

D Drag up/down in the display, or turn the Time knob. to change the delay time. Alternatively, repeatedly click
the Tap button to manually tap the desired delay time.
Range: 2-5000 ms. In Sync mode, the range is 1/128 to 8/4.

674 RIPLEY SPACE DELAY

 D In Dual Delay mode, drag up/down in the respective L/R display, or set the time for the respective channels
with the Time L and Time R knobs.
Alternatively, repeatedly click the Tap buttons to manually tap the desired delay time for the Left and Right chan-
nels respectively.
Range: 2-5000 ms. In Sync mode, the range is 1/128 to 8/4.

Multiply

D Set the Multiply selector to 1/2 to reduce the delay Time(s) and Offset by half, or to X2 to double the Time(s)
and Offset values.
! Note that changing the Time Multiply parameter always re-pitches the delayed signal (one octave up or down),
regardless of the Keep Pitch parameter setting. This simulates the characteristic “octave effect” found in old
delay effects.

Offset

D Set the delay time offset between the left and right channels.
Setting a delay offset is perfect for creating stereo delay effects.

Feedback and Freeze

D Turn the Feedback knob to set the number of delay repetitions.

At zero Feedback there will only be a single repeat. Unity gain is achieved at 100%, where the delay repetitions
will be infinite. If you increase the Feedback beyond 100% the gain will increase for every repeat so a distorted
signal is produced. You can limit the level/distortion using the “The Feedback Limiter”.
Range: 0-130%
D Click the Freeze button to freeze the audio currently in the delay buffer.
This will cause infinite repeats of the delay repetitions and also stop new audio from entering the delay buffer -
great for temporary stutter/loop effects, and also for creating special time/pitch effects using the “Multiply” func-
tion.

675 RIPLEY SPACE DELAY

 Wobbler

This emulates a tape speed wobbling effect where the speed of the “tape” (and consequently, the pitch of the signal)
wobbles randomly.
D Set the wobble amount with the knob.
Note that the pitch will always wobble, regardless of if the Keep Pitch function is active or not.

Ping Pong

With Ping-Pong enabled, the stereo position of each delay repeat will alternate between the left and right channels.
The Pan knob determines the stereo width as well as the position of the initial repeat. When the Pan knob is set to
full Left, the first delay repeat will be panned hard Left, the second will be panned hard Right, the third repeat will be
panned hard Left, and so on. When the knob is set to full Right, the order is reversed (R > L > R etc).

Delay Width

D Set the desired stereo width of the Delay output signal with the Delay Width knob.

The Feedback Filter section

The Feedback Filter makes it possible to filter the delay repeats so that each repeat is filtered slightly differently. A
great example of this is the “dub delay” effects you could achieve with tape echo effects back in the day.
The Feedback Filter has two (internal) modes:
• Band mode
The Band mode features eight fixed bandpass filters where you can control the level of each band.
• Hi-Lo mode
The Hi-Lo mode features a single bandpass filter, where you can set the lower and upper cutoff frequencies to
create a filter band.
! It’s also possible to process the feedback signal by connecting external devices to the “External Feedback
Loop Send & Return L&R” jacks on the rear panel. Any added effects device(s) will be connected in series be-
fore the Feedback Filter, i.e. they will not bypass the Feedback Filter.

676 RIPLEY SPACE DELAY

 The Band mode

D Click or drag the level bars of each frequency band to get the desired initial frequency response of the delay re-
peats.
Note that for each repeat the frequency response is altered slightly.
! Note that setting high levels (above the center line) of the bands will increase the volume for each “loop”,
which could cause loud volumes! As a protection, you can use the Feedback Limiter function, see “The Feed-
back Limiter”.
D Turn the Freq Shift knob to shift all the filters up or down in frequency, changing the character of the feedback
signal.
This parameter can also be modulated from the “The Modulation Matrix”, for phaser-like sweeps and special ef-
fects.
D Turn the L/R Offset knob to offset the frequencies of the Left and Right channels, to create a stereo effect.

The Low/High mode

D Drag either edge of the band horizontally to set the cutoff frequencies.
D Drag the entire “frequency band” horizontally to change the band’s position.
D Drag the Low Cut and Hi Cut “boxes” up/down to change the cutoff frequencies.

The Feedback Limiter

The Feedback Limiter is useful if you have set high levels of the bands in Band mode (see “The Band mode”). Acti-
vating Feedback Limiter will reduce the levels of any “overdriven” filter bands. The Feedback Limiter also prevents
loud volumes in external feedback loops (see “External Feedback Loop Send & Return L&R”).

677 RIPLEY SPACE DELAY

 The Reverb section

The Reverb section contains a lush stereo reverb with selectable routing and width control.
D Set the reverb level with the Amount knob.
D Set the reverb time with the Decay knob.
D Set the reverb “room size” with the Size knob.
D Click the Parallel button, or the routing image, to switch between Serial and Parallel processing of the delay
and reverb effects.
D Set the desired stereo width of the reverb signal with the Space Width knob.

The Noise section

The Noise section allows you to add noise to the signal - or to amplitude-modulate the signal by noise - to simulate
the noise of old hardware echo machines, for example. At the bottom of the Noise section you can choose where to
apply the noise in the signal chain.
D Set the noise level with the Amount knob.
D Set the noise frequency content with the Character knob.
D Add high-frequency noise (before running the signal through the Noise Type section) with the Hi End knob.

678 RIPLEY SPACE DELAY

 D Select noise type with the Noise Type selector.
The hand-crafted Noise Types simulates various types of noises, spanning from High Frequency (HF) noise down
to Low Frequency (LF) Noise.
D Click the Stereo button to get the noise effect in stereo.
D Click the triangular routing selector to choose one of the following destinations for applying the noise:
“Input” applies the noise to the audio inputs of Ripley Space Delay.
“Feedback” applies the noise to the Feedback Filter section.
“Delay Out” applies the noise to the Delay Out signal.
“Space Out” applies the noise to the Reverb Out signal.
“Output” applies the noise to the main output signal.
q See the “Signal flow” picture to see the patch points.

The Distortion section

The Distortion section allows you to distort the signal. At the bottom of the Distortion section you can choose where
in the signal chain the distortion should be applied.
D Set the distortion character with the Tone knob.
D Set the distortion mix level with the Dry/Wet knob.
D Set the overdrive amount with the Drive knob.
D Click the triangular routing selector to choose where to insert the distortion to the signal:
“Input” applies the distortion to the audio inputs of Ripley Space Delay.
“Delay Out” applies the distortion to the Delay Out signal.
“Space Out” applies the distortion to the Reverb Out signal.
“Output” applies the distortion to the main output signal.
q See the “Signal flow” picture to see the patch points.

The Digital section

The Digital section allows you to add digital distortion (sample rate and bit reduction) to the signal. At the bottom of
the Digital section you can choose where in the signal chain the distortion should be applied.

679 RIPLEY SPACE DELAY

 D Set the sample rate with the Rate Crush knob.
Setting a lower sample rate will cause aliasing, which adds inharmonic overtones to the signal.
D Set the bit depth with the Bit Crush knob.
The lower the bit depth, the more digitally distorted the signal.
D Click the triangular Output routing selector to choose where to apply the bitcrushing:
“Input” applies the rate/bit crushing to the audio inputs of Ripley Space Delay.
“Delay Out” applies the rate/bit crushing to the Delay Out signal.
“Space Out” applies the rate/bit crushing to the Reverb Out signal.
“Output” applies the rate/bit crushing to the main output signal.
q See the “Signal flow” picture to see the patch points.

The EQ section

The EQ section features low and high shelving bands with fixed frequencies, plus a parametric mid band. At the bot-
tom of the EQ section you can choose where in the signal chain the EQ should be applied.
• Low Gain
Defines how much the level of the fixed frequency low band should be boosted (positive values) or lowered (neg-
ative values).
Range: +/-18 dB
• Freq
This determines the center frequency of the EQ mid band, e.g. at which frequency the level should be decreased
or increased.
Range: 61.0 Hz to 12.69 kHz
• Gain
Specifies how much the level of the mid band (Freq) should be boosted (positive values) or lowered (negative val-
ues).
Range: +/-18 dB
• Q
This governs the width of the affected area around the set mid band (Freq) frequency. The higher the value, the
narrower the affected frequency range.
• High Gain
Defines how much the level of the fixed frequency high band should be boosted (positive values) or lowered (neg-
ative values).
Range: +/-18 dB
D Click the triangular Output routing selector to choose where to apply the EQ to the signal:
“Input” applies the EQ to the audio inputs of Ripley Space Delay.
“Delay Out” applies the EQ to the Delay Out signal.
“Space Out” applies the EQ to the Reverb Out signal.
“Output” applies the EQ to the main output signal.
q See the “Signal flow” picture to see the patch points.

680 RIPLEY SPACE DELAY

 The Output section

The Output section consists of a Dry/Wet knob and an Output Gain knob.
D Set the relation between the unprocessed and the processed signal with the Dry/Wet knob.
If you use Ripley Space Delay as a send effect, set the Dry/Wet knob to 100% Wet.
D Set the output level with the Output Gain knob.
Range: -Inf to +18.1 dB

The Ducker section

Ducking attenuates the level of the Wet (processed) signal until the amplitude of the Dry signal drops, at which time
the Wet signal is faded back in. This is useful for adding the effect to the silence that comes after you have played/
sung your lovely lead line. The effect will be attenuated while you are still playing/singing so you will avoid muddling
up the sound.
D Set the ducking level of the effect signal with the Amount knob.
At 100%, the effect signal will be completely silent as long as there is a signal coming into Ripley Space Delay.
The ducking attenuation level is shown by the orange bar below the Amount knob.
D Click the triangular Output routing selector to choose where to apply the ducking effect:
“Delay Out” applies the ducking to the Delay Out signal.
“Space Out” applies the ducking to the Reverb Out
“Output” applies the ducking to the main output signal.
q See the “Signal flow” picture to see the patch points.

The Modulation and Modulation Matrix sections

The Modulation and Modulation Matrix section consists of two separate LFOs, one Envelope Follower and a Macro &
Matrix section. All four sections can be active at the same time.

681 RIPLEY SPACE DELAY

 LFO 1 & LFO 2

An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. The LFO section features two separate
general purpose LFOs, that can be assigned to control selectable Destination parameter(s) in the Modulation Matrix
section, see “The Modulation Matrix”.
D Click the LFO 1 or LFO 2 buttons to access the parameters for the respective LFO.
LFO 1 and LFO 2 are identical in functionality.
D Select an LFO waveform by clicking the spin controls to the right of the waveform display, or by dragging up or
down in the display.
Besides the standard waveforms (sine, triangle, pulse, etc.) there are random, slope and stepped waveforms. The
shape of the waveforms are shown in the display.
! All LFO signals are bipolar (+/-), except for the Exponential Pulse, which is unipolar (+).
D Set the LFO frequency with the Rate knob.
D Click the Beat Sync button to sync the LFO to the main sequencer Tempo.
The Rate parameter now controls time divisions.
D Turn the Phase knob to adjust the phase of the LFO waveform.
Range: +/-180 degrees
D Assign the desired Destination parameters in the Modulation Matrix, as described in “The Modulation Matrix”.
The LFO signal affects the assigned Destination parameters of the Modulation Matrix according to the orange bi-
polar level indicator. The center of the indicator represents the zero level of the LFO signal:

Follow

The Envelope Follower allows for modulation based on the level of an audio signal. The modulation can be assigned
to control selectable Destination parameter(s) in the Modulation Matrix section, see “The Modulation Matrix”.
D Select modulation source from the Source drop-down.
The sources are: Input signal, Delay Out signal, Space Out signal, the Freeze Button and the Macro Button.
D Set at which input signal level the envelope follower should trig with the Sens knob.
Set to a low value, the envelope follower will react as soon as there is any signal present at the selected Source.
Set to a high value, the envelope follower will react only to loud input signals - or when triggered from the Freeze
button or Macro button signal, if any of these are selected as Source.
Range: +/-12 dB

682 RIPLEY SPACE DELAY

 D Set how fast the envelope follower should react after the input signal has reached above the Sens value with
the Rise knob.
Note that the Rise time can only be increased compared to the input signal - never shortened.
D Set how fast the envelope follower should drop back to zero after the input signal has decreased below the
Sens value with the Fall knob.
Note that the Fall time can only be increased compared to the input signal - never shortened.
D Assign the desired Destination parameters in the Modulation Matrix, as described in “The Modulation Matrix”.
The envelope follower signal affects the assigned Destination parameters of the Modulation Matrix according to
the orange unipolar level indicator. The center of the indicator represents the zero level of the envelope follower
signal:

Macro & Matrix

The Macro & Matrix allows for modulation controlled by the Macro Button and Knob, or by a number of other select-
able Sources. The modulation can be assigned to control selectable Destination parameter(s) in the Modulation Ma-
trix section.
1. Select the desired Source(s) in the Modulation Matrix:

2. Select the desired Destination parameters and modulation Amounts in the Modulation Matrix:

D Click the Button to activate, or click the LED above the Button to trigger momentarily.

The Modulation Matrix

The Modulation Matrix section in Ripley Space Delay is derived from the one in other Reason instrument devices, so
if you are familiar with Reason you will quickly find your way around in Ripley Space Delay’s Modulation Matrix.
There are three “Source –> Destination 1 –> Destination 2 –> Scale” buses.
A Source parameter can modulate two Destination parameters per bus. Each bus also has a Scale parameter that af-
fects the relative modulation Amount for the Destinations.

683 RIPLEY SPACE DELAY

 ! Note that the Source column is only available in the Macro & Matrix mode (see “Macro & Matrix”). In the other
modes the LFOs and Envelope Follower are the default modulation Sources.
q Note that it is possible to assign the same source parameter as Source in several buses. This allows you to
control more than one Destination parameter from the same Source.
1. In Matrix & Macro mode, select the desired Source parameters by clicking the corresponding Source box and
selecting from the list.
The following parameters can be used as modulation Sources:

| Parameter | Description
LFO 1/2 This allows you to modulate parameters from LFO 1/2.
Env. Follower This allows you to modulate parameters from the Envelope Follower signal.
Macro Knob This allows you to modulate parameters from the Macro Knob.
Macro Button This allows you to modulate parameters from the Macro Button.
Constant This sends out a constant value.
Ducker Follower This allows you to modulate parameters from the Ducker modulation signal.
CV1/2 Inputs This takes the current value on the CV 1 and CV 2 inputs on the rear panel and sends to the desired
destination(s).

Modulation Matrix Source parameters

2. Set the Amount for the Destination 1 by clicking and dragging vertically in the corresponding AMT box.
! Note that the Amount range is +/-100. This means that the Amount value can exceed the modulated parame-
ter’s range. When this happens, the modulated parameter simply stays at its extreme value until the Amount
value gets within the parameter’s range again.
3. Click the desired Destination box and select the Destination parameter from the list.
The following parameters can be used as modulation Destinations:

| Parameter | Description
Delay: Delay Time This affects the Delay Time parameter in “single delay” mode.
Delay: Time Offset L-R This affects the Time Offset L-R parameter.
Delay: Time Multiplier This affects the Time Multiplier parameter.
Delay: Feedback This affects the Feedback parameter.
Delay: Feedback Offset L-R This affects the “under the hood” Feedback Offset L-R parameter which allows the Left and Right
delay
signals to have their own separate feedback.
Delay: Ping Pong Pan This affects the Ping Pong Pan parameter (if Ping Pong is active).
Delay: Freeze This affects the Freeze button. The Freeze parameter is activated as soon as the Source modulation
level is >50% (>64).
Delay: Delay Level This affects the “under the hood” Delay Level parameter, which controls the output level from the De-
lay
section.
Delay: Wobbler Amount This affects the Wobbler Amount parameter (if Wobbler is active).
Delay: Delay Width This affects the Delay Width parameter, which affects the delay stereo spread.
Delay Dual: Delay Time L This affects the Delay Time L parameter in Dual Delay mode.
Delay Dual: Delay Time R This affects the Delay Time R parameter in Dual Delay mode.
Delay Dual: Delay Pan L This affects the “under the hood” Delay Pan L parameter in Dual Delay mode.
Delay Dual: Delay Pan R This affects the “under the hood” Delay Pan R parameter in Dual Delay mode.
Delay Dual: Delay Level L This affects the “under the hood” Delay Level L parameter in Dual Delay mode.
Delay Dual: Delay Level R This affects the “under the hood” Delay Level R parameter in Dual Delay mode.

684 RIPLEY SPACE DELAY

 | Parameter | Description
Feedback Filter: Lo Cut Frequency This affects the Lo Cut Frequency parameter in Low/High mode in the Feedback Filter.
Feedback Filter: Hi Cut Frequency This affects the Hi Cut Frequency parameter in Low/High mode in the Feedback Filter.
Feedback Filter: Hi & Lo Frequency Shift This affects the Frequency Band position in Low/High mode in the Feedback Filter.
Feedback Filter: Band Frequency Shift This affects the Frequency Shift parameter in Band mode in the Feedback Filter.
Feedback Filter: Band Offset L-R This affects the Offset L-R parameter in Band mode in the Feedback Filter.
Feedback Filter: Band 1-8 Level This affects the respective Band Level parameters in Band mode in the Feedback Filter.
Feedback Filter: Band Odd Level This affects the “under the hood” levels of the odd (1-3-5-7) Bands in Band mode in the Feedback
Filter.
Feedback Filter: Band Even Level This affects the “under the hood” levels of the even (2-4-6-8) Bands in Band mode in the Feedback
Filter.
Space: Space Amount Affects the Amount parameter in the Space section.
Space: Space Decay Affects the Decay parameter in the Space section.
Space: Space Width Affects the Space Width parameter in the Noise section.
Noise: Noise Amount Affects the Amount parameter in the Noise section.
Noise: Noise Character Affects the Character parameter in the Noise section.
Noise: Noise High End Affects the Hi End parameter in the Noise section.
Noise: Noise Stereo Affects the Stereo button in the Noise section.
Dist: Dist Dry-Wet Affects the Dry/Wet parameter in the Dist section.
Dist: Dist Tone Affects the Tone parameter in the Dist section.
Dist: Dist Drive Affects the Drive parameter in the Dist section.
Digital: Digital Dry-Wet Affects the Dry/Wet parameter in the Digital section.
Digital: Digital Bit Crush Affects the Bit Crush parameter in the Digital section.
Digital: Digital Rate Crush Affects the Rate Crush parameter in the Digital section.
EQ: Low Gain Affects the Low Gain parameter in the EQ section.
EQ: Mid Frequency Affects the Freq parameter in the EQ section.
EQ: Mid Gain Affects the Gain parameter in the EQ section.
EQ: Mid Q Affects the Q parameter in the EQ section.
EQ: High Gain Affects the High Gain parameter in the EQ section.
Modulation: LFO 1/2 Rate Affects the LFO 1/2 Rates
Modulation: CV 1/2 This sends out the source modulation value(s) to the CV1/2 Output on the rear panel.
Output: Ducker Amount Affects the Amount parameter in the Ducker section.
Output: Dry-Wet Affects the Dry/Wet parameter in the Output section.
Output: Output Gain Affects the Output Gain parameter in the Output section.

Modulation Matrix Destination parameters

4. Repeat the procedure from Step 2 for the AMT(2) and Destination 2 parameters.

685 RIPLEY SPACE DELAY

 5. If desired, click the Scale box and select a Scale parameter.
The available Scale parameters are the same as the Source parameters, see “Modulation Matrix Source parame-
ters”.
Selecting a source here means the modulation is scaled by that source.
For example, if you set up a modulation as follows:
Source AMT Destination 1 Scale
LFO 1 50 Delay: Feedback Macro Knob

then the LFO will modulate the Feedback parameter in the Delay section, but the modulation will be scaled by the
Macro Knob in the Macro & Matrix section. When the Macro Knob is at zero, there will be no modulation at all.
Raising the Macro Knob will gradually increase the modulation up to the set Amount (50).
D To clear an assigned Source, Destination or Scale parameter, hold down [Ctrl](Win) or [Cmd](Mac) and click
the Source/Destination/Scale box. Alternatively, click the Source/Destination/Scale box and select “Off” from
the list.
D To reset an Amount value to 0, hold down [Ctrl](Win) or [Cmd](Mac) and click the desired Amount box.
D To clear an entire modulation assignment (a whole row), click the circular X button to the right of the corre-
sponding Scale box.

Connections

Audio jacks
Input L&R
D Patch the main audio input signal(s) here.
If your input signal is in mono, connect only to the L (left) input.

Output L&R
D Patch the main audio output signal(s) here.
If your input signal is in mono, connect only to the L (left) input.

686 RIPLEY SPACE DELAY

 External Feedback Loop Send & Return L&R
D If you want to route the Feedback signal via external effects, connect the effect here.
The signal is tapped and returned before the Feedback Filter in Ripley Space Delay. Any added effects device(s)
are connected in series before the Feedback Filter, i.e. they will not bypass the Feedback Filter. If you want to tap
the signal in mono, connect only to the L (left) Send and Return input and output.

CV Inputs
These CV inputs can be used for modulating various Ripley Space Delay parameters from external sources (devices).
Each input also has an attenuator knob for tuning the modulation range.

Delay Time
D Modulate the Delay Time parameter (in “single delay” mode) by a CV signal patched to this input.

Feedback
D Modulate the Feedback parameter by a CV signal patched to this input.

Freeze
D Modulate the Freeze button by a CV signal patched to this input.

Time Multiplier
D Modulate the Time Multiplier switch by a CV signal patched to this input.

Dry/Wet
D Modulate the Dry/Wet parameter in the Output section by a CV signal patched to this input.

CV1/2 In
D Connect the CV signals you want to assign as Sources in the Modulation Matrix.

CV Outputs
LFO1/2, Envelope Follower and CV1/2 Out
D Patch these to the desired destinations to modulate parameters in external devices.
Note that all CV Outs can send bipolar signals, except for the Envelope Follower, which only sends positive signals.

687 RIPLEY SPACE DELAY

688 RIPLEY SPACE DELAY
Chapter 41
RV7000 Mk II
Advanced Reverb
 Overview

The RV7000 Mk II is a high quality reverb processor. It features ten different reverb and echo algorithms, ranging
from rooms and halls to special effects. The Mk II version also incorporates a high-quality zero-latency convolution al-
gorithm, which makes it possible to load sampled impulse responses - and even sample and use your own impulse
responses!
Since the RV7000 Mk II comes with a number of useful reverb presets, you could simply select one and tweak the
most important parameters on the main panel - or you could use the Remote Programmer panel to fine-tune the re-
verb in great detail.
The RV7000 Mk II also contains an equalizer and a gate section. Both of these are for processing the actual reverb
sound, making it possible to get virtually any kind of reverb character, including gated reverb.

About the Patch format

The RV7000 Mk II features programmable effect presets. In the Factory Sound Bank you will find a number of preset
Patches which can be used as they are or provide you with a good starting point for further tweaking.
Patches use the Windows file extension “*.RV7”. Loading and saving Patches is done in the same way as for instru-
ment devices.
! Songs and/or Patches that use Convolution (or have an Impulse Response sample loaded) can not be opened
in older Reason versions. The same goes for Combi Effect patches that contain RV7000 Mk II in Convolution
mode (or with a sample loaded, or with the Combi programmer referencing Convolution parameters).
! Convolution patches that use samples only contain references to the samples - not the sample itself. This is
the way all sampler devices in Reason work.

Connections
Typically you connect the RV7000 Mk II as a send effect, as this allows you to use it for processing several different
mixer channels. However, it’s also possible to use it as an insert effect - use the Dry/Wet control on the main panel
to adjust the balance between the dry, unprocessed sound and the reverb. Note:
D The RV7000 Mk II is a true stereo reverb (except in Convolution mode), which means that it will use the stereo
input information when processing both channels (without summing the input channels).
It’s also possible to use it as a mono in - stereo out effect. Which type of connection to use (mono or stereo in) de-
pends on the material. If the audio sources are in mono (or in stereo but with no important difference between the
left and right channel) using a mono input is sufficient.
D If you want to use RV7000 Mk II’s Reverse reverb effect, you should consider connecting it as an insert effect or
using a Send on the Mixer, with Pre-fader mode selected (and the channel fader lowered).
This is because you typically don’t want to hear the dry sound when using the Reverse effect. See “The Gate sec-
tion”.

690 RV7000 MK II ADVANCED REVERB

 The main panel

The RV7000 Mk II main panel.

When you create an RV7000 Mk II, only the main panel will be shown. This contains a section for handling patches,
on/off buttons for the EQ and Gate sections, the most important reverb parameters and a dry/wet mix control. To se-
lect a reverb patch and make coarse adjustments, this is all you need.

The Remote Programmer

Clicking the arrow button next to the “cable slot” on the main panel brings up the remote programmer panel.

This is where you make detailed settings for the reverb. Note:
• The Edit Mode button to the left determines which section to make settings for, Reverb, EQ or Gate.
• Settings are made with the eight dials around the graphic display. The functions of the dials differ depending
on the selected Edit Mode and the selected reverb algorithm. Next to each dial, the display shows the name
and value of the corresponding parameter.

• Not all modes and algorithms use all eight dials. If a dial isn’t used in the selected mode, nothing will be shown
next to it in the display.
• You cannot make settings in the graphic display itself - this is for showing a graphic representation of the se-
lected reverb.

691 RV7000 MK II ADVANCED REVERB

 Reverb algorithms and parameters
Common effect device parameters
While the specific parameters for the RV7000 Mk II effect device are described below, some features and proce-
dures are common to all effect devices. Please, refer to “Common effect device features” for information about the
Input meter, the Bypass/On/Off switch and Signal Flow Graphs on the effect device.

About the main panel parameters

On the main panel you find three parameters that are available for all algorithms:

| Parameter | Description
Decay This governs the length of the reverb or the feedback if an echo algorithm is selected.
HF Damp Controls how quickly the high frequencies should decay in the reverb. Raise it to gradually remove high frequencies, mak-
ing the reverb sound warmer and less bright.
HI EQ This is a high-shelving EQ that works much like a typical treble control on a mixer or amplifier. Lower the setting for a
softer reverb sound or raise it to get more high frequencies.

Selecting an algorithm
You select a reverb algorithm in the remote programmer panel:
1. Click the remote programmer arrow button on the main panel to display the remote programmer panel.
2. Make sure the Edit Mode button is set to Reverb.
3. Use the top left dial to select a reverb algorithm.
The selected algorithm is shown in the display next to the dial.

692 RV7000 MK II ADVANCED REVERB

 Here’s a quick overview of the ten algorithms - for details and parameter descriptions, see below.

| Algorithm | Description
Small Space Emulates a small enclosed space (a small room or a resonant body).
Room Emulates a room with adjustable shape and wall character.
Hall Emulates a hall.
Arena Emulates a large arena, with separate pre-delay for the left, right and center reverbs.
Plate Emulates a classic plate reverb.
Spring Emulates a spring reverb, as used in e.g. guitar amplifiers.
Echo An echo effect with gradually diffusing echo repeats. Can be synced to Reason’s tempo.
Multi Tap A multi-tap delay with four different delay lines and tempo sync.
Reverse A reverse reverb that “pushes” the dry sound to appear after the reverb. The result is a backwards reverb leading up to
the direct sound.
Convolution The zero-latency Convolution algorithm uses impulse response samples. The samples are used for generating the de-
sired reverb effect (or actually any type of effect - depending on what sample you use).

Small Space
This algorithm places the sound in a small enclosed space, ranging from a tiny resonant body to a room. The param-
eters are:

| Parameter | Description
Size The size of the emulated space.
Mod Rate The reverb can be randomly modulated for a more even sound (or for special effects). This parameter sets the rate of
modulation (the amount is set with Mod Amount).
Room Shape Select from four different room shapes, affecting the character of the reverb.
LF Damp Controls how quickly the low frequencies should decay in the reverb. Raise it to gradually remove low frequencies,
making the reverb sound “thinner” and less boomy.
Wall Irreg Adjusts the positioning of the emulated walls in the small space. The lowest setting emulates two directly opposed
walls while higher settings emulate more walls and angles, for a more complex resonance.
Predelay Sets the predelay time, i.e. the delay between the source signal and the start of the reverb.
Mod Amount Sets how much the reverb will be modulated. Use fairly low settings when emulating real rooms and resonant bodies,
and higher settings for special effects.

Room
Emulates a medium-sized room, with the following parameters:

| Parameter | Description
Size The size of the emulated room.
Diffusion At low Diffusion settings, you will hear the individual reverb “bounces” more clearly, while higher settings produce a
more “smeared”, dense and even reverb.
Room Shape Select from four different room shapes, affecting the character of the reverb.
ER->Late The first “answers” in the reverb are called early reflections (ER) and are typically more pronounced than the actual
reverb tail. This parameter sets the time between the early reflections and the reverb tail. This is set as a percentage
- the actual delay time depends on the Size setting.
ER Level Adjusts the level of the early reflections. “0” is normal level.
Predelay Sets the predelay time, i.e. the delay between the source signal and the start of the early reflections and reverb.
Mod Amount Sets how much the reverb will be modulated. Moderate modulation gives a natural, less static sound.

693 RV7000 MK II ADVANCED REVERB

 Hall
Emulates a hall. The parameters are the same as for the Room algorithm above (but the Hall algorithm offers larger
Size settings).

Arena
Emulates the ambience in an arena or concert hall, with long pre-delay times (separate for left, right and center):

| Parameter | Description
Size The size of the emulated arena or hall.
Diffusion At low Diffusion settings, you will hear the individual reverb “bounces” more clearly, while higher settings produce a
more “smeared”, dense and even reverb.
Left Delay The predelay time for the left side of the reverb.
Right Delay The predelay time for the right side of the reverb.
Stereo Level Adjusts the level of the left and right sides of the reverb. “0” is normal level.
Mono Delay The predelay time for the mono (center) reverb signal.
Mono Level Adjusts the level of the mono (center) reverb signal. “0” is normal level.

Plate
A classic plate reverb, excellent for vocals for example. The parameters are:

| Parameter | Description
LF Damp Controls how quickly the low frequencies should decay in the reverb. Raise it to gradually remove low frequencies,
making the reverb sound “thinner” and less boomy.
Predelay Sets the predelay time, i.e. the delay between the source signal and the start of the reverb.

Spring
An emulation of a spring reverb as can be found in guitar amplifiers, organs, etc. The spring reverb has the following
parameters:

| Parameter | Description
Length Sets the length of the simulated spring.
Diffusion At low Diffusion settings, you will hear the individual reverb “bounces” more clearly, while higher settings produce a
more “smeared”, dense and even reverb.
Disp Freq When sending a signal to a real-life spring reverb, the initial transient will produce a quick, characteristic sweeping
tonal noise. This is because different frequencies in the sound are delayed by different amounts (a phenomenon
called dispersion). This parameter controls the frequency of that sound.
LF Damp Controls how quickly the low frequencies should decay in the reverb. Raise it to gradually remove low frequencies,
making the reverb sound “thinner” and less boomy.
Stereo (on/off) Determines whether the output of the spring reverb should be in mono or stereo.
Predelay Sets the predelay time, i.e. the delay between the source signal and the start of the early reflections and reverb.
Disp Amount Sets the amount of dispersion effect (see Disp Freq above).

694 RV7000 MK II ADVANCED REVERB

 Echo
This is an advanced echo effect, with diffusion controls and tempo sync. When Echo is selected, the Decay control on
the main panel controls the echo feedback (the number of echo repeats). The parameters are:

| Parameter | Description
Echo Time Sets the time between each echo.
When Tempo Sync (see below) is off, the echo time is set in milliseconds (10 - 2000 ms); when Tempo Sync is on you
set the echo time as a number of 1/16 notes or 1/8 triplet notes, in relation to the current song tempo.
Diffusion When this is set to 0, the echo will sound as a standard delay with clear, precise repeats. Raising the Diffusion setting
will introduce additional echoes very close to the “main” echo repeats, causing a “smeared” echo sound. This will also
expand the echo stereo image.
Tempo Sync Determines whether the echo time should be freely set (“off”) or synchronized to Reason’s tempo (“on”).
LF Damp Controls how quickly the low frequencies should decay in the echoes. Raise it to gradually remove low frequencies.
Spread Adjusts the spacing of the additional echoes added by the Diffusion parameter. For a very smeared echo (sound more
like a reverb), set both Diffusion and Spread to their maximum values.
Predelay Sets an additional delay time before the first echo repeat.

Multi Tap
The Multi Tap delay produces up to four different delays with separate delay times, panning and level. The whole set
of four delay taps can then be repeated at a given rate. Again, the Decay control on the main panel controls the feed-
back (the number of repeats for the whole multi tap set). All delay times can be tempo synced.
Note: this algorithm is handled a bit differently since you make separate settings for each delay tap:
• The parameters to the left of the display are common for all taps.
• You use the Edit Select parameter in the top right corner to select which tap to make settings for - the three
parameters below affect the currently selected tap.

Tap 2 selected for editing.

• You can also set Edit Select to “Repeat Tap” - this is where you specify the repeat time for the whole multi tap
“package”.
With short Repeat times, the first tap may be repeated before the last tap has sounded. This can be used to create
very complex multiple delay effects.

The common parameters (to the left) are:

| Parameter | Description
Tempo Sync Determines whether the delay times and repeat times should be freely set (“off”) or synchronized to Reason’s
tempo (“on”).

695 RV7000 MK II ADVANCED REVERB

 | Parameter | Description
Diffusion Raising the Diffusion setting will introduce additional echoes very close to the “main” repeats, causing a “smeared”
delay sound.
LF Damp Controls how quickly the low frequencies should decay in the echoes. Raise it to gradually remove low frequencies.

When Tap 1 - 4 is selected with the Edit Select parameter, you can make the following settings for the selected delay
tap:

| Parameter | Description
Tap delay Sets the delay - the time from the source signal to the tap.
When Tempo Sync is off, the delay time is set in milliseconds (10 - 2000 ms); when Tempo Sync is on you set the
delay as a number of 1/16 notes or 1/8 triplet notes, in relation to the current song tempo.
Tap level Adjusts the level of the selected tap.
Tap pan Adjusts the pan of the selected tap.

When Repeat Tap is selected with the Edit Select parameter, there is only one parameter to the right in the display:

| Parameter | Description
Repeat Time Sets the time between each repeat of the whole multi tap set. The number of repeats is set with the Decay control
on the main panel.
When Tempo Sync is off, the repeat time is set in milliseconds (10 - 2000 ms); when Tempo Sync is on you set the
repeat time as a number of 1/16 notes or 1/8 triplet notes, in relation to the current song tempo.

Reverse
The Reverse reverb algorithm in RV7000 Mk II is special in that it actually “moves” the source audio as well. Sounds
fed into the Reverse reverb are “sampled”, a reverse reverb is created and played back and finally the “sampled” orig-
inal sound is played back. For example, if you feed a snare drum hit into the Reverse reverb, you will hear a rising
“backwards” reverb, followed by the snare drum hit.
Therefore, you probably don’t want to hear the first, original (dry) sound. Here is how to set this up:
D Connect the RV7000 Mk II as an insert effect and make sure the Dry/Wet control on the main panel is set fully
to “Wet”.
Note that with this algorithm, raising the Decay setting on the main panel will make the reverse reverb start earlier
and build up under a longer time. Similarly, the HF Damp parameter affects how fast the high frequencies are built up
in the reverse reverb. In the remote panel, the Reverse algorithm has the following parameters:

| Parameter | Description
Length This sets the time from when the source signal is fed into the reverb until it is played back again. It is during this
time you will hear the reverse reverb, as shown in the display.
The time can be set in milliseconds or as note values, depending on whether Tempo Sync is off or on.
Note: As stated above, the Decay setting determines the length of the actual reverse reverb - in essence how soon
it starts after the source signal. But of course, the reverse reverb cannot start before the original source signal! If
you set Decay to a longer time than the Length setting, the reverse reverb will start abruptly, immediately when the
source signal is fed into the reverb. If this sounds complicated, just take a look at the RV7000 Mk II display and try
the settings - you will soon see how it works.
Note also that very high Length settings demand a lot of processor power. This can be reduced by adjusting the
Density parameter, see below.
Density Density governs the “thickness” of the Reverse effect. If this parameter is turned down to zero, the effect produces
individual delays rather than a dense “wash”, which can be used as a special effect. Worth noting is that if Density
is set to around 50%, this can considerably reduce the CPU load without altering the sound of the effect too much.
Exactly how much the Density parameter can be reduced without altering the sound depends on the source mate-
rial.
Rev Dry/Wet Sets the balance between the “moved” source signal (“dry”, low values) and the reverse reverb (“wet”, high values).
Tempo Sync Determines whether the Length setting should be freely set (“off”) or synchronized to Reason’s tempo (“on”).

696 RV7000 MK II ADVANCED REVERB

 Convolution

The zero-latency Convolution algorithm uses impulse response samples to generate effects. Basically, “convolution
effects” are the results of multiplying the frequency spectra of the input signals with the frequency spectra of impulse
response samples, and thus generating a signal with the “character” of the impulse response sample. If the impulse
response sample is a recording of the reflections of a large room, for example, the resulting effect will be “the input
audio signal played back in a large room”.
RV7000 Mk II comes with three built-in preset impulse response samples. You can also use any other samples for
the convolution algorithm, to generate all kinds of reverbs and special effects. You can even sample your own im-
pulse responses and use in the convolution algorithm in RV7000 Mk II.
Note that in the Convolution algorithm the input signals for the effect are first summed to mono (except in Parallel
Stereo Mode (see “Stereo Mode”)) and then processed with the impulse response sample. The figures below shows
the signal routings in the Convolution algorithm:

Dry L Dry L
In L Out L In L Out L
Convolution Convolution
Wet L Wet L
IR L IR L
+
IR R IR R
Wet R Wet R

In R Out R In R Out R
Dry R Dry R
(All Stereo and Mono modes) (Parallel mode)

In the remote panel, the Convolution algorithm has the following parameters:

| Parameter | Description
Sample Preset Here you select one of the preset impulse response samples - plus your own sample (if you use that, see “Loading
impulse response samples”).
Length Sets the length (end point) of the currently used impulse response sample.
Size Simulates the “size” of the impulse response sample, in practice its pitch, in semitone steps. -12 means pitching up
the impulse response sample 1 octave and 12 means pitching it down 1 octave.
LF Damp Controls how quickly the low frequencies should decay. Raise it to gradually remove low frequencies, making the ef-
fect sound “thinner” and less boomy.

697 RV7000 MK II ADVANCED REVERB

 | Parameter | Description
Stereo Mode Determines how the output signal should be presented. Note that the stereo effect is derived from the impulse re-
sponse sample itself. So, the wider the stereo effect in the impulse response sample, the wider the stereo effect in
the output.

Note that if you use a mono impulse response sample, all effects below will be in mono!

Stereo: The input signals are summed and then a stereo effect with a spread of 100% is applied.
Stereo75%: The input signals are summed and then a stereo effect with a spread of 75% is applied.
Stereo50%: The input signals are summed and then a stereo effect with a spread of 50% is applied.
Stereo25%: The input signals are summed and then a stereo effect with a spread of 25% is applied.
Mono: The input signals are summed and then the effect is panned to a centered mono signal (0% spread).
M->S Slow: From Mono slowly panned out to full Stereo.
M->S Fast: From Mono quickly panned out to full Stereo.
S->M Slow: From full Stereo slowly narrowed down to Mono.
S->M Fast: From full Stereo quickly narrowed down to Mono.
Parallel: Individual processing of the two L and R stereo input signals (dual mono).
Predelay Sets the predelay time, i.e. the delay between the source signal and the start of the convolution effect. A negative
predelay masks the start of the sample, so if the sample has unwanted initial transients or silence, these can be “re-
moved”.
Gain Sets the amplification or attenuation of the effect signal.

Below are some things to keep in mind when you are working with the Convolution algorithm:
• If you want the impulse response sample to play back exactly like the original, make sure the Decay knob is at
max, the LF Damp knob at zero and the Hi EQ knob at its 12 o’clock position. Also, make sure the Length pa-
rameter is at 100% and the Size parameter at 0.
• The Length value of the impulse response sample is also affected by the Decay knob setting.
If the impulse response sample is quiet at the end, reduce the Length value to cut it off a little earlier. The Decay
parameter introduces a smoother “cutoff” at the end, which might be desirable in many situations.
• Changing some convolution parameters re-calculates the impulse response in real time. Therefore, modulat-
ing these parameters might give unexpected results. Specifically:

Using CV to modulate the RV7000 Mk II in Convolution mode is not recommended.

If you're using the Gate function (see “The Gate section”) in combination with Convolution, we recommend
setting the Decay Mod parameter to 0 (see “Decay Mod”).

Loading impulse response samples

Besides the preset impulse response samples, you can load any other samples into the RV7000 Mk II and use as im-
pulse responses. Loading a sample automatically switches the Algorithm to Convolution.
D Drag and drop a sample onto the RV7000 Mk II panel to automatically load it in the Convolution Algorithm.
Or, do as follows:
1. Click the Browse Samples button on the Programmer panel to set browse focus to the RV7000 Mk II.

698 RV7000 MK II ADVANCED REVERB

 2. Select the desired sample in the Browser and load it.

Here we have selected and loaded the “Fx_DubHead.WAV” sample to use as impulse response. You can see that
the sample name is now displayed to the right of the Preset knob.
! Note that the maximum length of a sample used as an impulse response is approximately 12 seconds. Longer
samples are automatically truncated to 12 seconds (non-destructive).
! Note that any embedded loop data in the samples are disregarded when loaded into the RV7000 Mk II!
! Note that the Programmer display only shows the first 4 seconds of longer samples.
q To achieve stereo effects in the Stereo Mode alternatives (see “Stereo Mode”) you have to use stereo sam-
ples.
3. Edit the Convolution parameters on the RV7000 Mk II panel until you are satisfied.
D If you want to save your RV7000 Mk II patch with your impulse response sample, click the Save Patch button on
the upper panel:

! Note that the impulse response sample is NOT saved in the patch itself - only a reference to the sample! If you
have loaded an external sample (that is not in the Factory Soundbank), be sure NOT to move/remove the sam-
ple from its original location on your computer.

699 RV7000 MK II ADVANCED REVERB

 The EQ section

The equalizer in RV7000 Mk II affects the wet reverb sound only and is used for shaping the character of the reverb.
There are two EQ bands, one for low frequencies (shelving) and one full-range parametric EQ.
D To activate the EQ, click the EQ Enable button on the main panel so that the indicator lights up.
D To make EQ settings, select “EQ” with the Edit Mode button to the left in the remote programmer panel.
D In this mode, the remote programmer display shows a frequency curve, indicating the settings you make with
the EQ parameters.
The parameters are:

| Parameter | Description
Low Gain The amount of cut or boost of the low-shelving filter.
Low Freq The frequency below which the Low Gain cut or boost is applied.
Param Gain The amount of cut or boost for the parametric EQ.
Param Freq The center frequency of the parametric EQ, e.g. at which frequency the level should be decreased or increased.
Param Q This governs the width of the affected area around the set center frequency. The higher the value, the narrower the af-
fected frequency range.

D Remember that you have a third EQ band at your disposal - the HI EQ parameter on the main panel.
The reason why this is on the main panel and not in the EQ section is simply that it’s a setting you may want to ad-
just often, without having to open the remote programmer panel.

700 RV7000 MK II ADVANCED REVERB

 The Gate section

The Gate section allows you to create gated reverb effects with a lot of options and possibilities. You can either trig-
ger the gate from the source audio signal or via MIDI or CV.
When triggering the gate from the source audio signal, it works like this:
• The gate “listens” to the source (dry) signal and opens whenever the signal reaches a certain threshold level.
• The reverb sound is sent through the gate - when the gate is closed you won’t hear the reverb.
• When the source signal level drops below the threshold level, the gate closes after a time that depends on the
Hold parameter and the level of the source signal (see the parameter table).
D If you need the gate to be open for an exact duration (time), you should trigger it via MIDI or CV.
In audio trigger mode, the actual gate time will vary depending on the source signal.
When triggering the gate via MIDI or CV, it works like this:
• The reverb sound is sent through the gate - when the gate is closed you won’t hear the reverb.
• Whenever the gate receives any MIDI note (sent to the RV7000 Mk II) or a gate signal (connected to the Gate
Trig CV input on the back of the RV7000 Mk II), the gate opens for the duration of the note or gate signal.
Note:
D To activate the Gate, click the Gate Enable button on the main panel so that the indicator lights up.
D To make Gate settings, select “Gate” with the Edit Mode button to the left in the remote programmer panel.
D In this mode, the remote programmer display shows two meters - one showing the signal level (with an indica-
tion of the threshold level) and one showing the status of the gate.
This is useful for checking what happens, how the gate triggers, etc.

701 RV7000 MK II ADVANCED REVERB

 The parameters for the Gate section are:

| Parameter | Description
Threshold When Trig Source is set to “Audio”, this determines the audio signal level at which the gate opens. If you raise
this setting, only very loud sounds will open the gate.
Decay Mod This modulates the reverb Decay parameter so that the decay time is lowered when the gate closes. When this
is set to zero, no decay modulation happens - this means that if the gate is closed and then opened again, you
may hear “previous” reverb tails that are still ringing. If you raise the Decay Mod setting, the decay will automat-
ically be lowered when the gate is closed, eliminating this effect.
Trig Source Determines whether the gate should be triggered by audio or MIDI/CV, as described above.
High Pass A high-pass filter that affects the audio that triggers the gate (only active when Trig Source is set to “Audio”). If
you raise this setting, sounds with low frequencies only will not open the gate. Note that this setting doesn’t af-
fect the sound of the reverb, only the triggering mechanism.
Attack Determines how long it takes for the gate to open after a triggering signal has been received.
Hold This parameter is only active when Trig Source is set to “Audio”. Hold affects how quickly the gate closes, in the
following way:
Internally, the gate is controlled by an envelope follower that analyzes the source signal level and generates a
“level CV signal” accordingly. This signal is compared to the Threshold level to determine whether the gate
should be opened or closed. The Hold parameter affects how quickly the envelope follower responds when the
source signal level drops - you could say that this is the decay control for the envelope follower. The higher the
Hold setting, the longer it will take for the envelope follower signal to drop below the threshold level and close
the gate. But the resulting time also depends on the source signal level - with a loud signal, it will take longer
time for the envelope follower to drop to the threshold level. Therefore, the actual gate time depends both on
the Hold setting and on the character of the source audio.
Release Determines how long it takes for the gate to close after the Hold time.

CV Inputs

On the back of the RV7000 Mk II you find three CV inputs. These are:

| Parameter | Description
Decay Controls the reverb decay or echo/delay feedback via CV.
HF Damp Controls the HF Damp parameter on the main panel.
Gate Trig Used for triggering the Gate section with a gate signal. The length of the gate signal determines the
length of the gated reverb.

! Using the CV inputs to modulate the RV7000 Mk II in Convolution mode is not recommended.

702 RV7000 MK II ADVANCED REVERB

Chapter 42
Scream 4
Sound Destruction Unit
 Scream 4 Sound Destruction Unit

Scream 4 is a very versatile stereo in/out sound destruction device, capable of warping any sound literally beyond
recognition, but also capable of producing more subtle musical effects. Scream 4 features a wide range of algorithms
for distortion and sound mangling which can be combined with an EQ and a resonant “Body” section to provide ev-
erything you need to add an edge to your sounds. This effect is most often used as an insert effect.

About the Patch format

Unlike most of the other effect devices, Scream 4 features programmable effect presets. Included are a number of
factory Patches which can be used as they are or provide you with a good starting point for further tweaking.
Patches use the Windows file extension “*.SM4”. Loading and saving Patches is done in the same way as for instru-
ment devices.

Parameters
Scream 4 contains three main sections; Damage (distortion and other types of sound destruction), Cut (EQ) and
Body (places the sound in a resonant environment - can serve as anything from a cabinet emulator to a wah-wah to
completely new special effects) which can be switched on or off independently. The parameters in each section are
as follows:

Common effect device parameters

While the specific parameters for the Scream 4 effect device are described below, some features and procedures are
common to all effect devices. Please, refer to “Common effect device features” for information about the Input meter,
the Bypass/On/Off switch and Signal Flow Graphs on the effect device.

Damage section controls

The “Damage” section is where you specify the basic sound mangling algorithm and make settings to inflict the de-
sired amount of damage to the sound. There are ten basic algorithms to chose from, ranging from classic distortion
effects to digital-sounding warping and modulation effects.

704 SCREAM 4 SOUND DESTRUCTION UNIT

 There are five controls in this section, with the following functions:

| Parameter | Description
Damage button This switches the Damage section on or off.
Damage Control knob This controls the input gain which in turn determines the amount of damage inflicted. The
higher the value, the more destruction!
When raising the Damage Control you may need to lower the Master level to maintain the
same output level (and vice versa).
Damage Type knob This selects the type of effect - see the table below for a description of the available
damage methods.
P1/P2 knobs The functionality of these knobs vary according to the selected Damage Type - see the
table below for a description.

Description of the various Damage Type algorithms

Here follows a basic description of the ten Damage Types available, and what parameters the P1/P2 knobs control
for each type:

| Type | Description
Overdrive This produces an analog-type overdrive effect. Overdrive is quite responsive to varying dynamics. Use
lower Damage Control settings for more subtle “crunch” effects.
- The P1 knob controls the basic tone of the effect. Turn clockwise for a brighter sound.
- The P2 knob controls Presence. Presence boosts frequencies in the high midrange before the distortion
stage which in turn affects the character of the distortion. Turn clockwise for more Presence boost.
Distortion Similar to Overdrive, but produces denser, thicker distortion. The distortion is also more “even” across the
Damage Control range compared to Overdrive.
- The P1/P2 knobs control Tone and Presence, respectively - see Overdrive for a description.
Fuzz Fuzz produces a bright and distorted sound even at low Damage Control settings.
- The P1/P2 knobs control Tone and Presence, respectively - see Overdrive for a description.
Tube This emulates tube distortion.
- The P1 knob controls Contour, which is somewhat like a high pass filter, changing the tone and character
of the distortion.
- The P2 knob controls Bias, which changes the “symmetry” of the tube distortion. Setting this to the min-
imum or maximum value will produce asymmetrical distortion (typical of a real-life tube amplifier), while a
12 o’clock setting will produce symmetrical distortion (odd harmonics only).
Tape This emulates the soft clipping distortion produced by magnetic tape saturation and also adds compres-
sion which adds “punch” to the sound.
- The P1 knob controls Speed, which simulates tape running at different speeds. The higher the Speed
setting the more of the original high frequency material in the signal. Turn clockwise for a brighter sound.
- The P2 knob controls the amount of Compression. Turning the knob clockwise increases the compres-
sion ratio.
Feedback This effect combines distortion in a feedback loop which can produce many interesting and sometimes
unpredictable results. Feedback is basically when a sound source is fed back to itself. An open micro-
phone picking up sound from a nearby loudspeaker that is also being used to amplify sound from the mi-
crophone will produce a feedback loop with the associated typical howling. For this effect the Damage
Control knob controls the gain of the feedback loop.
- The P1 knob controls Size, which could be described as the “length” (i.e. the distance between the micro-
phone and the loudspeaker in the above example) of the feedback loop.
- The P2 knob controls Frequency, which for this effect determines which overtones will “howl”.
Modulate Modulate first multiplies the signal with a filtered and compressed version of itself, and then adds distor-
tion. This can produce resonant, ringing distortion effects.
- The P1 knob controls Ring, which is the resonance of the filter. Turn clockwise for more pronounced
ringing effects.
- The P2 knob controls Frequency, which is the filter frequency. Turn clockwise to raise the filter frequency
which generally produces a sharper, more piercing effect.

705 SCREAM 4 SOUND DESTRUCTION UNIT

 | Type | Description
Warp Warp distorts and multiplies the incoming signal with itself.
- The P1 knob controls Sharpness. Lower values will produce a soft, compressed distortion, while higher
values produces more harmonics and a sharper sound.
- An effect of multiplying a signal with itself is that the fundamental pitch is removed from the signal, leav-
ing overtones only. The P2 knob controls Bias - raise this to reintroduce the fundamental pitch in the
sound.
Digital Lo-fi anyone? This reduces the bit resolution and sample rate for raw and dirty sounds or for emulating
vintage digital gear.
- The P1 knob controls (bit) Resolution. If the knob is turned fully to the right there is no bit reduction, fully
the left the resolution is 1-bit.
- The P2 knob controls the sample rate. If the knob is turned fully to the right the sample rate is not re-
duced; turning it to the left gradually reduces the sample rate.
Scream Similar to Fuzz, but with a bandpass filter with high resonance and gain settings placed before the distor-
tion stage.
- The P1 knob controls the basic tone of the effect. Turn clockwise for a brighter sound.
- The P2 knob controls the filter frequency. The high resonance setting of the filter makes it suitable for
wah-wah effects.

Cut section (EQ)

The sliders in the Cut section are tone controls, allowing you to cut or boost the level by up to 18dB in the low, mid
and high frequency areas. The Cut section is activated with the Cut button above the sliders.
Move the slider from the middle upwards to boost the level, and from the middle downwards to cut the level of the
corresponding frequency area.

Body section

The Body section is just what it says - it places the sound in a resonant “body”. Depending on the settings, the result
can be similar to a speaker cabinet simulator, an auto-wah effect, or effects with no real-world counterpart. The sec-
tion is based on 5 basic body types, which simulate how a sound is affected by different physical enclosures. The size
and resonance of the Body types can be changed, and the section also features an envelope follower.

706 SCREAM 4 SOUND DESTRUCTION UNIT

 The Body parameters are as follows:

| Parameter | Description
Body button This switches the Body section on or off.
Body Type knob This is used to select one of the five available Body types (A-E).
Body Reso knob This simulates the resonance of the selected Body. Turning the knob clockwise gives a more
resonant effect.
Body Scale The Body Scale parameter could be said to control the “size” of the Body. Note that this is “in-
verted” - turning the knob clockwise reduces the emulated size.
Auto knob Determines the amount of envelope follower effect on the Scale parameter - see below.

About the envelope follower

The Body section features an envelope follower for dynamic control of the Scale parameter. The envelope follower
analyzes the amplitude of the incoming signal and changes the Scale setting accordingly - the louder the incoming
signal the higher the value of the Scale parameter. The operating frequency (or size) range is set with the Scale pa-
rameter, and the envelope follower amount is set with the Auto parameter. A typical use for this is auto-wah effects
- try Body Type “B” for a pronounced wah effect.
• On the back of the Scream 4 you will find an Auto CV output - this delivers the CV signal from the envelope fol-
lower, allowing you to dynamically control parameters in other devices. See “Creating a real dynamic wah ef-
fect with the envelope follower” for an example.

About the Master level control

The Master level control should be used when you need to increase or decrease the output level, while retaining the
basic character of the effect. It can also be used to balance the level between the distorted sound and the “clean”
(unprocessed) sound if the effect is to be switched in and out in the mix.
If the output level is too high, turning down the Damage Control setting would lower the output, but it would also
change the character of the distortion, as would changing eq or presence settings.
Simply lowering the mixer channel level (for the channel that Scream 4 is connected to) would also work of course,
but this would also mean that the level difference between the unprocessed and processed sound would increase.
So if the clip indicator lights up on the Transport, or if the distorted sound is too loud compared to the normal sound,
the solution is to lower the Master output level.
As pointed out elsewhere in the manual, audio out clipping (indicated by the red clip indicator lighting up on the
Transport Panel) can only happen in the Reason Hardware Interface. In other words, you never have to worry about
levels passed internally from device to device. However, bear in mind that if you use high Master output settings (or
a lot of boost in the Cut section) Scream 4 can quite easily cause audio out clipping - and that is most likely not a dis-
tortion effect you want!

707 SCREAM 4 SOUND DESTRUCTION UNIT

 CV inputs and outputs

On the back of the Scream 4 you will find CV inputs for controlling the following four parameters:
D Damage Control
Use this for dynamically changing the amount of damage effect.
D P1
The use for this depends on the selected Damage Type. For example, if the Feedback effect is selected, this will
control the Size parameter - connect it to the CV Out on a Matrix or synth LFO for strange, flanger-like sweeps.
D P2
The use for this depends on the selected Damage Type. For example, if the Scream effect is selected, this will
control the Frequency parameter, producing a distorted wah wah sound.
D Scale
Lets you control the Scale parameter in the Body section from another CV source, for wah wah-like effects, etc.
In addition, you find a CV output from the “Auto” (envelope follower) function in the Body section. By connecting this
to a CV input for a parameter in another device, the level of the signal going into the Scream 4 will affect that param-
eter. See below for an example on how to use this.

708 SCREAM 4 SOUND DESTRUCTION UNIT

 Screamy tips and tricks
Don’t restrict yourself to using Scream 4 as a basic distortion stompbox, but try it in as many ways as possible - you
may be surprised to find how often Scream 4 can add power, warmth and color to your sounds. Here are some ex-
amples:

Creating a heavy drum sound

Scream 4 is ideal for processing drums. Try connecting it as an insert effect to a Redrum device and experiment with
the Damage Types and settings.
• For classic distorted drums, try the Tube, Tape or Distortion algorithms.
• The Scream algorithm is excellent for a really raw body or industrial drum sound.
• For more weird, synth-like effects, try the Modulation or Warp effects.
Remember that you don’t have to route the whole drum kit through the Scream device - sometimes it may be better
to route the individual outputs from the bass drum, snare and/or toms to a Spider Audio Merger (see “Spider Audio
Merger & Splitter”), connect the merged output of the Spider to the Scream 4 and route this to a separate channel in
the Mixer. That way, hi-hats, cymbals and similar are unprocessed.

Warming up a mix with the Tape effect

If you find your mixes a bit sterile, the Tape algorithm is excellent for providing some warmth and gentle distortion:
1. Create a Scream 4 device and connect it between the main outputs of the Mixer and the Audio Hardware de-
vice.
2. Set the Damage Type to Tape.
Make sure the Cut and Body sections are turned off.
3. Start with a low Damage Control setting and P1 (Speed) and P2 (Compression) at 12 o’clock.
4. Play back your mix and adjust the settings.
Raise the Damage Control for more tape saturation distortion, adjust P1 to get the desired brightness and raise P2
if you want a more controlled, compressed sound. If you like, you could also activate the Cut section and use the
three-band EQ to further adjust the sound.

Using the Body section as a sound enhancer/phaser/wah

Nothing stops you from using the Body section on its own, without Damage. Try this:
1. Create a sampler device (e.g. an NN-19) and select an electric piano patch.
2. Select the sampler and create a Scream 4.
It is added as an insert effect.
3. Turn off the Damage section and instead activate the Body section.
You will find that this adds a resonant character to the sound, which will make it more “alive” and help it stand out in
a mix. You should experiment with the Body settings to find the character that suits you best. You could also activate
the Cut section- if you e.g. find the sound too bassy, just lower the “Lo” slider a bit.
4. Now flip the rack around and connect the CV out from the sampler’s LFO to the Scale CV input on the Scream
4.
As you can hear, the Scale is modulated by the LFO.

709 SCREAM 4 SOUND DESTRUCTION UNIT

 5. Adjust the amount of Scale modulation with the pot next to the CV input on the back of the Scream 4, and the
speed (and waveform) of the modulation in the LFO section on the sampler device.
With this type of modulation setup, it’s easy to get lush, dreamy phaser effects. For a more wah wah-like sound, se-
lect Body type B and raise the Reso and Scale settings.

Emulating vintage digital gear

The first generations of digital instruments (drum machines, synths and samplers) used 8 or 12 bit sampling and pro-
cessing, at low sample rates. This “lo-fi” sound is still in high demand, e.g. in hip-hop and R’n’B. Try this:
1. Connect a Scream 4 as an insert for a Redrum, with a suitable kit selected.
2. Set Damage Type to Digital and turn P1 and P2 fully right.
3. Play back and lower P1 (bit depth) and P2 (sample rate) to get the sound you want.
You may also want to use the Cut section to emphasize or cut frequencies in the sound.

Creating a real dynamic wah effect with the envelope follower

As we have shown, you can get auto-wah-like effects with the Body section in Scream 4 (by using the Auto param-
eter). You could also use the ECF-42 envelope controlled filter and trigger this with a gate signal - this is after all a
“real” filter and sounds even more like a wah effect. However, to get a “real” auto-wah effect that responds to the sig-
nal level, you need to combine both these devices:
1. Create an instrument device that you want to process with an auto-wah.
It should be velocity responsive so that the harder you play, the louder it sounds.
2. Create a Scream 4 device and an ECF-42 device.
Both these should now be connected as insert effects to the instrument device.
3. Turn off all three sections in the Scream 4.
This is of course a matter of taste - but here we will show how to use the envelope follower in Scream 4, not its
sound destruction capabilities.
4. Flip the rack around and connect the Auto CV Output on the Scream to the Freq CV input on the ECF-42.
5. Lower the pot next to the CV input a bit - the envelope follower is rather sensitive and you probably don’t want
the filter to open too much.
You can adjust this later if needed.
6. On the ECF-42, select the BP 12 (bandpass) mode and set the Res setting rather high.
7. Play the instrument device and adjust the Freq setting on the ECF-42 to taste.
As you can hear, the harder (or the more notes) you play, the more the filter will open.
q If you find the auto-wah too responsive, you could add a compressor between the instrument device and the
Scream 4 to even out the level differences a bit.
q The Spider CV Merger and Splitter (see “Spider CV Merger & Splitter”) can be used to invert and split the Auto
CV output for even greater flexibility.

710 SCREAM 4 SOUND DESTRUCTION UNIT

Chapter 43
Sidechain Tool
 Introduction

Sidechain Tool is an all-in-one device for creating that characteristic sidechain pumping effect. Sidechain Tool does
both automatic pumping to tempo (no external routing needed!), triggered pumping, and regular sidechain
compression. Sidechain Tool is mainly suited as an Insert Effect - on a Mix Channel or Output Bus channel - for
rhythmically “ducking” the channel volume. However, you can also use it as a Send Effect, using the Send Mode
function.

Panel reference
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Auto Pump/Trigger/Sidechain mode switch

D Select which mode you want the Sidechain Tool to operate in:
Auto Pump uses the built-in ducking curve synced to an internal clock, which can be set to suitable tempo inter-
vals. The clock is automatically synced to the tempo of the main (Reason/DAW) sequencer.
Trigger mode also uses the built-in ducking curve, but you trigger the ducking either via MIDI notes, from the Side-
chain Audio inputs, or from the Trig CV input on the rear panel.
Sidechain mode uses the built-in compressor, for traditional sidechain compression, see “Sidechain mode specific
parameters”.

Ducking curve parameters (Auto Pump and Trigger modes)

The ducking curve display and parameters are common to the Auto Pump and Trigger modes.
Here is where you can shape the curve that ducks the incoming audio.

712 SIDECHAIN TOOL

 Ducking curve presets

D Click to select a ducking curve preset.

You could then edit and adjust each curve preset to your liking, as described below.

Attack, Hold, Release, Attack Shape, Release Shape and Slide

D Turn the Attack, Hold, Release, Attack Shape and Release Shape knobs to change the ducking curve shape.
Any changes are immediately reflected in the display.
D Alternatively, click the blue square points in the display and drag to shape the ducking curve.
The changes are also reflected by the Attack, Hold and Release knobs to the left of the display:

D Click and drag the Attack/Release Shape curves to change their shapes.
The changes are also reflected by the Attack Shape and Release Shape knobs to the left of the display:

D Turn the Slide knob to slide the entire curve back or forth, to affect the timing of the ducking:

Ducking Amount

D Set how much you want to attenuate (duck) the incoming audio.
The attenuation (ducking amount) is reflected by the blue shaded area above the curve in the display.

713 SIDECHAIN TOOL

 Auto Pump mode specific parameters
Rate

D Turn the Rate knob to set the desired repetition rate (time division) of the ducking.
Range: 8 bar, 4 bar, 3 bar, 2 bar, 4/4, 3/4, 2/4, 3/8, 1/4, 1/4T, 3/16, 1/8, 1/8T, 1/16, 1/16T and 1/32.

Trigger mode specific parameters

Audio/MIDI switch

D Select if you want to trig the ducking from an audio input signal connected to the a sidechain input(s) on the
rear panel (see “Sidechain Input Left & Right”) - or via MIDI Notes (see “MIDI Trigger”).

Audio Threshold

D Set the desired Threshold level by dragging the triangular “slider” up/down.
As soon as the audio on the sidechain input on the rear panel reaches above the set Threshold, the orange Trig in-
dicator lights up above the Rate knob.

MIDI Trigger
To be able to trigger the ducking curve from MIDI Notes, you first have to create a sequencer track for the Sidechain
Tool device. In Reason you do that as follows:
1. Right click the device panel and select “Create track for Sidechain Tool”.
A sequencer track for the Sidechain Tool has now been created in the sequencer.
2. Make sure the sequencer track has Keyboard Focus (is selected in the sequencer).
The default setting when you switch to MIDI Trigger is “All”, i.e. any MIDI Note# will trig the ducking curve.

714 SIDECHAIN TOOL

 If you like you could change this so that only a specific MIDI Note will trigger the curve:

1. Click the Note triangle and select “Note” from the popup.
2. Click the Learn button and play the desired note on the keyboard.
3. When you are satisfied with the selected note, click the Learn button again.
4. Play the selected note on the keyboard to trig the ducking.
The orange Trig indicator now lights up above the Rate knob.

Sidechain mode specific parameters

In Sidechain mode Sidechain Tool works as a regular sidechained compressor, i.e. it does not make use of the Duck-
ing Curve function. Instead it uses the Gain, Threshold, Attack Release and Ratio parameters to control the built-in
compressor.

Gain
D Set the ”drive” of the compression with the Gain knob.
This determines how much compression the signal will have in conjunction with the Threshold. Range: ±24 dB.

Listen
D Click the Listen button to listen only to the sidechain signal (input on the Sidechain Input(s) on the rear panel,
see “Sidechain Input Left & Right”).

Frequency band display

715 SIDECHAIN TOOL

 By filtering the sidechain signal you can make the compression more or less sensitive to a certain frequency range.
A typical application for this is to isolate the bass frequencies of a kick drum, to have the kick drum trigger the com-
pressor. Another typical application is de-essing - where harsh “S”-sounds in vocal material is reduced or eliminated.
E.g. for de-essing you should set the frequency “boundaries” to isolate the offending “S” frequencies as much as pos-
sible.
D Click and drag the frequency lines horizontally in the display to set the frequency band of the sidechain signal
that should trigger the compressor.

Threshold
D Set the level at which onset of compression should occur.
When the input level is below the Threshold setting the signal is unaffected. When the input level exceeds the
threshold, compression kicks in.
In practice, this means that the lower the Threshold setting (and the higher the Input Gain), the more compression
will be applied. Range: -52 dB to 0 dB

Attack
D Set how quickly the compressor should apply its effect when signals rise above the set Threshold.
If you raise the Attack, the response will be slower, allowing more of the signal to pass through the compressor un-
affected. Typically, this is used for preserving the attacks of the sounds. Range: 1ms to 1s.

Release
D Set how long it should take before the compressor lets the sound through unaffected - after the signal level
has dropped below the set Threshold.
Set this to short values for intense, “pumping” compressor effects, or to longer values for a smoother change of the
dynamics. Range: 20ms to 5.02s.

Compression display

This display shows how the signal is being compressed (attenuated) by the sidechain signal over time.

Ratio
D Set the amount of gain reduction that should be applied to the signals above the set Threshold.
The Ratio can be set from 1:1 (no reduction) to Infinite:1

716 SIDECHAIN TOOL

 Common parameters for all modes
Send Mode

D Click the Send Mode button if you are using Sidechain Tool as a send effect instead of an insert effect.
This will make Sidechain Tool send out only the processed signal, without any of the unprocessed signal.

Band Mode

With Band Mode active you can force Sidechain Tool to only duck a certain frequency band of the signal. A typical
example would be to have it duck only the bass frequencies when triggered/sidechained from a kick drum.
D Click and drag the blue dot horizontally in the display to define the desired frequency band to duck.
The ducked frequency band is indicated by the shaded blue area in the display.
D Click the triangular button to select “Lo Band” or “Hi Band”.
“Lo Band” ducks low frequencies and “Hi Band” ducks high frequencies.

Connections

CV Input
Pump Amount (Auto Pump and Trigger modes only)
D Modulate the Ducking Amount parameter by a CV signal patched to this input.

717 SIDECHAIN TOOL

 Attack & Shape (Auto Pump and Trigger modes only)
D Modulate the Attack time by a CV signal patched to the Attack input.
D Modulate the Attack curve Shape by a CV signal patched to the Shape input.

Hold (Auto Pump and Trigger modes only)

D Modulate the Hold time by a CV signal patched to this input.

Release & Shape (Auto Pump and Trigger modes only)

D Modulate the Release time by a CV signal patched to the Attack input.
D Modulate the Release curve Shape by a CV signal patched to the Shape input.

Band Freq
D Modulate the Frequency parameter (when Band Mode is active) by a CV signal patched to this input.

Trig CV (Trigger mode only)

D Modulate the Trig button (in Trigger mode) by a CV signal patched to this input.

CV Output
Trig, Hold, End (Auto Pump and Trigger modes only)
• These outputs send out high Gate CV pulses (in Auto Pump and Trigger modes) according to the graphic on
the rear panel.
Trig is sent at the start of the ducking curve, Hold is sent out at the end of the Hold stage and End is sent out
when the curve has finished.

Curve Out & Inverted

• These outputs send out continuous CV signals according to the current ducking curve or compression value.
The Inverted output sends the curve signal values inverted.
! Note that the Curve Out sends negative CV signals and the Inverted sends positive CV signals.

Sidechain Input Left & Right

D Patch the audio signal(s) you want to control the compressor (in Sidechain mode) here.
If your input signal is in mono, connect only to the L (left) input.

Input Left & Right

D Patch the audio signal(s) you want to duck/compress here.
If your input signal is in mono, connect only to the L (left) input.

Output Left & Right

These are the stereo audio outputs.

718 SIDECHAIN TOOL

Chapter 44
Softube Amps
 Introduction

The Softube Amp at the top and the Softube Bass Amp at the bottom
The Softube Amp and Softube Bass Amp are amplifier and speaker cabinet simulators based on the renowned mod-
elling algorithms developed by Softube. The two devices feature accurate simulations of some of the most coveted
vintage amplifier and cabinet models, that you can freely mix and match. Setting up a great amp tone for your guitar
or bass is easy, regardless of what type of sound you are looking for. In this chapter the two devices will be described
together using the collective name “Softube Amps”. Where parameters differ between the two devices this will be
duly noted. In Reason Rack Plugin the Softube Amp and Softube Bass Amp Rack Extension devices can be found on
the Effects palette.
! Please, note that this device is not available in Reason Lite Rack Plugin.

Basic usage
The Softube Amps can be used to process any signal but are best used “live”, i.e. as insert effects on an audio track
that a live instrument is connected to, so that you can monitor the effect as you play and record. When recording
electric guitar or bass, the basic tonal character of the amp/cabinet very much affects how the instrument responds,
which in turn affects your playing. The Softube Amps allow you to set up a good basic tone with a minimum of fuss
so you can start recording straight away!

Front panel
The Softube Amps front panel can be divided into two main sections; the display area with Patch/Amp/Cabinet se-
lectors at the top, and a set of standard amplifier tone controls below. The Amp parameters (tone controls) are de-
scribed in “Amp panel controls”.

Common effect device parameters

While the specific parameters for the Softube Amps are described later in this chapter, some features and proce-
dures are common to all effect devices. Please, refer to “Common effect device features” for information about the
Input meter, the Bypass/On/Off switch and Signal Flow Graphs on the Softube Amps.

720 SOFTUBE AMPS

 Using the Softube Amps
Loading and saving patches

Loading and saving patches in the Softube Amps is done in the same way as with any other Reason device - see
“Sounds, Patches and the Browser” for more details.

Selecting Amp and Cabinet model

There are six basic Amp models to choose from; four in the Softube Amp and two in the Softube Bass Amp. There
are also four Cabinet + microphone combination models in the Softube Amp and three Cabinet + microphone com-
bination models in the Softube Bass Amp.
You can use any Amp model with any Cabinet + microphone model. For general model descriptions see “About the
Amp and Cabinet models”.

Selecting Amp model

D To select Amp model, click the desired Amp button:

Amp selectors on the Softube Amp and Softube Bass Amp respectively.
D To bypass the Amp section, click the Bypass button.

Selecting Cabinet model

You can use any Cabinet with any selected Amp model.
D To switch Cabinet model, click the desired Cab button:

Cabinet selectors on the Softube Amp and Softube Bass Amp respectively.
D To bypass the Cabinet section, click the Bypass button.

721 SOFTUBE AMPS

 About the Amp and Cabinet models
Softube Amp
The following Amp models are available in the Softube Amp:

| Amp Model | Amp modeled on

Twang A silvery American classic, great for clean and slightly overdriven guitars. Pop-
ular for country, blues and more.
Crunch British 60's classic that keeps its chiming, present character whether you play
it clean or drive it so hard it's on the verge of breaking down.
Rock A yummy, warm, fat, tube distortion. Just like Mama used to make it.
Lead The boogie man has come in all his high gain glory! Screaming leads and
power chords that sustain for days.

The following Cabinet models are available in the Softube Amp:

| Cabinet Model | Cabinet + microphone modeled on

Bright A 2x12 cabinet with deep lows and heavenly highs. The oh-so-familiar dy-
namic mic is placed on axis but still lets a hint of room into the sound.
Room A more midrange present 2x12, with the small diaphragm dynamic mic placed
off axis and a wide stereo room added.
Fat Fat, solid 4x12 classic. Captured with a German condenser microphone on
axis for that full bodied sound.
Tight Slightly midrange scooped 4x12 cabinet. The large diaphragm dynamic micro-
phone gives an in your face presence.

Softube Bass Amp

The following Amp models are available in the Softube Bass Amp:

| Amp Model | Amp modeled on

Modern A contemporary version of a classic tube design from one of the most popular
bass amplifier brands. Modern has all the clarity and punch you'd expect from
a modern amplifier, but still with the pleasant three-dimensionality that well de-
signed tube amplifiers offer.
Vintage Got grit? Vintage gives you a dark, dirty, steamy bass sound that adds power
and attitude, and helps making the bass instrument cut through the mix.

The following Cabinet models are available in the Softube Bass Amp:

| Cabinet Model | Cabinet + microphone modeled on

Dark A classic 8x10 cabinet with a snappy response across the frequency spec-
trum. Captured with a German large membrane dynamic microphone, straight
on axis for a clear sound and a rock solid low end.
Bright A British 4x12 bass cabinet with great midrange presence. The dynamic mi-
crophone is placed off axis for a slightly roomy sound.
Room A 4x10 cabinet, captured with a German classic large membrane condenser
mic at a distance. Roomy with a nice depth and width that caters for a very nat-
ural sound in the mix.

722 SOFTUBE AMPS

 Amp panel controls
Softube Amp

The panel controls are used for tweaking the sound. The following controls can be found in the Softube Amp:

| Control | Description
Gate knob Here you can set a gate threshold for the Softube Amp. Signals below the set threshold are not let
through.
Boost switch This boosts the input gain, for more crunch.
Gain This controls the input gain. The higher the Gain value, the more crunch you get.
Bass/Mid/Treble controls These are tone controls which you can use to cut or boost the bass, midrange and high frequencies,
respectively.
Poweramp Gain This is the gain control for the modeled power amp.
Volume Controls the master volume of the Softube Amp.

Softube Bass Amp

The panel controls are used for tweaking the sound. The following controls can be found in the Softube Bass Amp:

| Control | Description
Drive This controls the input gain. The higher the Drive value, the more crunch you get.
Bass/Middle/Mid Freq/Treble controls These are tone controls which you can use to control the bass, midrange and high frequencies, re-
spectively. The Bass and Treble controls also have Ultra Lo/Hi buttons for emphasizing the low and
hight frequency ranges even further.
There is also a Mid Freq selector where you can select different Mid Frequency EQ characteristics.
Volume Controls the master volume of the Softube Bass Amp.

723 SOFTUBE AMPS

 Connections

• Both Softube Amps have Left/Right inputs and outputs.

If you only connect in mono to the Left input, the output will still be in stereo (if you connect both outputs).
• The Softube Amp has CV inputs for controlling Gate, Gain and Volume.
• The Softube Bass Amp has CV inputs for controlling Drive and Volume.

724 SOFTUBE AMPS

Chapter 45
Stereo Tool
 Introduction

The Stereo Tool is a stereo widener, which can take a mono source signal and turn it into a big, wide stereo signal
without causing any phase problems. You could for example use it for breathing new life into vocals - or for
generating stereo signals from mono instrument devices like Subtractor. The Stereo Tool can also be used to further
widening stereo source signals.

Panel reference
Widening

D Set the stereo spread amount with the Widening knob.

At 0%, stereo input signals will have the same stereo spread as the original input signals.
Turn up the knob to make the stereo output signals wider. As opposed to other stereo widening effects, Stereo
Tool also works on mono input signals. Note that the stereo widening is achieved without causing any phase prob-
lems.

Low Bypass

D Set how much of the low frequencies in the input signal(s) you want to bypass from the spread effect.
This function is great if you only want to widen the higher frequencies of the input signal(s) and keep the lower fre-
quencies unaffected.

Frequency Adjust

D Adjust the Frequency Adjust knob to change the built-in comb filter frequency.
This will change the frequency spectrum of the output signal. Tweak to taste if you, for example, should experience
that the signal is louder in one of the stereo channels.
q Connect a triangle LFO signal to the Frequency Adjust CV input on the rear panel to create kind of a stereo
chorus effect, see “Frequency Adjust CV In”!

726 STEREO TOOL

 Mono

D Click the Mono button to output a mono signal.

Stereo input signals will also be output in mono.

727 STEREO TOOL

 Connections

CV In
Widening CV In
D Modulate the Widening parameter by a CV signal patched to this input.

Frequency Adjust CV In
D Modulate the Frequency Adjust parameter by a CV signal patched to this input.
Connect a triangle LFO signal to create a stereo chorus effect.

Input Left & Right

D Patch the audio signal(s) you want to process here.
If your input signal is in mono, connect only to the L (left) input.

Output Left & Right

These are the stereo audio outputs.

728 STEREO TOOL

Chapter 46
Sweeper
Modulation Effect
 Introduction

The Sweeper Modulation Effect device is a great sounding Phaser/Flanger/Filter device.

By drawing your own unique modulation curve in the display and assigning this curve to the desired effect
parameters, you also get a very flexible system for repeatedly sweeping/modulating the effects parameters - in
perfect sync with the sequencer.
Sweeper is designed to be used mainly as an insert effect, for spicing up instrument sounds with nice sweeps and
modulations, but you could of course use it as you like!
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Panel reference
Global controls
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Volume

This is the master volume control.

730 SWEEPER MODULATION EFFECT

 Routing

D Click the Routing selector to select “Stereo” or Dual Mono” from the pop-up menu.
Stereo: With this selected the L+R input signals are mixed before being sent into the stereo effect. This means
you can connect a mono input signal and get stereo output signals.
Dual Mono: The L+R input channels are processed independently.

Spread

The Spread control detunes the stereo channels to generate a nice and wide stereo effect. Note, though, that the
Spread control works a little differently and has different ranges in the Phaser, Flanger and Filter.

Dry/Wet

This controls the mix of the dry and processed signals.

LFO -> Freq

This controls the modulation amount from the LFO (see “LFO”) to the Frequency control of the Phaser (see “Fre-
quency”), Flanger (see “Frequency”) and Filter (see “Frequency”) section.
The control is bipolar, which means that negative values will invert the modulation.

Mod -> Freq

This controls the modulation amount from the Modulator (see “The Envelope Modulator” and “The Audio Follower
Modulator”) to the Frequency control of the Phaser (see “Frequency”), Flanger (see “Frequency”) and Filter (see “Fre-
quency”) section.
The control is bipolar, which means that negative values will invert the modulation.

731 SWEEPER MODULATION EFFECT

 LFO -> Volume

This controls the modulation amount from the LFO (see “LFO”) to a separate built-in amplifier. The control is bipolar,
which means that negative values will invert the modulation.

Mod -> Volume

This controls the modulation amount from the Modulator (see “The Envelope Modulator” and “The Audio Follower
Modulator”) to a separate built-in amplifier.
The control is bipolar, which means that negative values will invert the modulation.

732 SWEEPER MODULATION EFFECT

 The Phaser

The Phaser consists of a number of all-pass filters (1 to 40) with feedback, which can be used for creating really nice
phasing effects. An all-pass filter lets all frequencies of a signal through - but phase inverted 180 degrees. The prin-
ciple is to split the input signal in two, run one signal dry and the other through a series of all-pass filters, and then
sum the two signals. The picture below shows the basic principle of a phaser:

Feedback
Bandwidth

Allpass filters + Output signal

Stage 1 Stage 2 Stage 3 Stage 40
Input signal

: audio signal
Frequency : control signal

Amplitude 3-stage phaser

Frequency
Frequency (log)

Frequency

Here you set the frequency of the all-pass filter(s) in the phaser.
Range: 37.6 Hz to 16.17 kHz

733 SWEEPER MODULATION EFFECT

 Bandwidth

This controls the bandwidth of the all-pass filter(s) in the phaser.

Amplitude 3-stage phaser Amplitude 3-stage phaser

Low Bandwidth High Bandwidth

Frequency Frequency
(log) (log)

Feedback

This controls the level/intensity of the phaser peaks and notches.

Amplitude 3-stage phaser Amplitude 3-stage phaser

Low Feedback High Feedback

Frequency Frequency
(log) (log)

Stages

In a phaser, a stage (also known as “pole”) is represented by an all-pass filter. Here you set the number of all-pass fil-
ters you want to use. Each all-pass filter contributes with one notch/peak in the frequency spectrum.

734 SWEEPER MODULATION EFFECT

 Range: 1-40 stages (notches)

Amplitude Amplitude Amplitude

1-stage phaser 2-stage phaser 3-stage phaser

Frequency Frequency Frequency

Polarity

Pressing this button will invert the polarity of the Phaser filter, so that instead of notches in the frequency spectrum,
there will be peaks:

Amplitude 3-stage phaser Amplitude 3-stage phaser

Inverted polarity

Frequency Frequency
Frequency (log) Frequency (log)

Mute Dry

Pressing this button mutes the dry signal in the Phaser section, turning the effect into a frequency-dependent delay.
Since no dry signal is mixed with the effect signal, there will be no notches in the frequency spectrum:

X
Feedback
Bandwidth

Allpass filters + Output signal

Stage 1 Stage 2 Stage 3 Stage 40
Input signal

: audio signal
Frequency : control signal

This will give more of a “tremolo” effect rather than phasing.

735 SWEEPER MODULATION EFFECT

 The Flanger

The Flanger is basically a Comb Filter with adjustable feedback, which can be used for creating a wide variety of cho-
rus effects and frequency swirls. The principle is to split the input signal in two, run one signal dry and the other
through a comb filter delay, and then sum the two signals. The picture below shows the basic principle of a flanger:

Feedback
+ Output signal
Comb Filter Delay

Input signal

: audio signal
: control signal
LFO Frequency

Amplitude

Frequency
Frequency (log)

Frequency

Here you set the comb filter frequency (in practice, the delay time between the dry and processed signals).
Range: 37.6 Hz to 16.17 kHz

736 SWEEPER MODULATION EFFECT

 Feedback

This intensifies the flange effect by increasing the resonance peaks via feedback.

Polarity

Pressing this button will invert the polarity of the Flanger, so that instead of peaks in the frequency spectrum, there
will be notches:

Amplitude Flanger Amplitude Flanger

Inverted polarity

Frequency Frequency
Frequency (log) Frequency (log)

Mute Dry

Pressing this button mutes most of the dry signal in the Flanger section:

Feedback
+ Output signal
Comb Filter Delay

Input signal

: audio signal
: control signal
LFO Frequency

737 SWEEPER MODULATION EFFECT

 The Filter

The Filter section features a selection of great sounding filters with various characteristics, derived from the Europa
Shapeshifting Synthesizer.

Drive

This amplifies and introduces an overdrive type of distortion to the signal in the filter.

Frequency

Here you set the cutoff frequency (for the HP and LP filter types) or the center frequency (for the BP and Notch filter
types).

Resonance

This controls the resonance amount, i.e. the amplification of the signal around the cutoff frequency.
! In the SVF Notch filter, the Resonance knob controls the width of the notch - from wide to narrow.

(Filter) Type

D Click the TYPE selector to select one of the following filter types from the pop-up menu:

738 SWEEPER MODULATION EFFECT

 • SVF HP 12dB
Amplitude

Resonance

Frequency
Frequency (log)
A state variable (SVF) highpass filter with a 12dB/octave slope.
• SVF BP 12dB
Amplitude

Resonance

Frequency
Frequency (log)
A state variable (SVF) bandpass filter with 12dB/octave slopes.
• SVF LP 12dB
Amplitude

Resonance

Frequency
Frequency (log)
A state variable (SVF) lowpass filter with a 12dB/octave slope.
• SVF Notch
Amplitude

Resonance

Frequency
Frequency (log)
A state variable (SVF) notch filter.

739 SWEEPER MODULATION EFFECT

 • Ladder LP 24dB
Amplitude

Resonance

Frequency
Frequency (log)
A ladder-type lowpass filter with a 24dB/octave slope. The resonance peak more narrow in this filter type than in
the MFB LP 24dB filter (see below). The filter can be driven to self-oscillate.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!
• MFB LP 12dB
Amplitude

Resonance

Frequency
Frequency (log)
A multiple feedback (MFB) lowpass filter with a 12dB/octave slope. If you turn up the Resonance high, additional
resonance peaks appear.
• MFB LP 24dB
Amplitude

Resonance

Frequency
Frequency (log)
A multiple feedback (MFB) lowpass filter with a 24dB/octave slope. The resonance peak is wider in this filter type
that in the Ladder filter (see above). The filter can be driven to self-oscillate. If you turn up the Resonance high, ad-
ditional resonance peaks appear.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!

740 SWEEPER MODULATION EFFECT

 • MFB HP 24dB
Amplitude

Resonance

Frequency
Frequency (log)
A multiple feedback (MFB) highpass filter with a 24dB/octave slope. If you turn up the Resonance high, additional
resonance peaks appear.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!
• K35 LP 12dB
Amplitude

Resonance

Frequency
Frequency (log)
An “early MS-20 type” of lowpass filter with a 12dB/octave slope. The filter can be driven to self-oscillate.
! Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!

LFO

The LFO can be used for cyclic modulation of the Frequency parameter of the Phaser/Flanger/Filter section - and/
or for modulating the Volume. The LFO Rate can also be synced to the Reason sequencer. You can also modulate
the LFO Rate from the Modulator (see “The Envelope Modulator” and “The Audio Follower Modulator”).

Waveform selector

D Click the up/down triangles - or drag the waveform display up/down - to select the desired LFO waveform.
Ten different LFO waveforms are available. Besides the standard waveforms (sine, triangle, pulse, etc.) there are
random, slope and stepped waveforms. The shape of the waveforms are shown in the display.
! Note that all waveforms except the “Decay” are bipolar, i.e., they generate both positive and negative levels.

741 SWEEPER MODULATION EFFECT

 Rate

Here you set the LFO Rate.

Range: 0.050-50.00 Hz
D Click the SYNC button to sync the LFO Rate to the main sequencer tempo.
Range in Sync mode: 8 Bars to 1/64.

Rate Mod

If you like, you can modulate the LFO Rate from the Modulator signal (see “The Envelope Modulator” below). If the
LFO is in SYNC mode, modulating the Rate will force the LFO to switch between the sync divisions.
D Set the desired LFO Rate Modulation amount with the knob.

The Envelope Modulator

The Modulator section features an Envelope and an Audio Follower. You can use either the Envelope or the Audio
Follower (but not together).
The Envelope is taken straight from the Europa Shapeshifting Synthesizer, so if you are familiar with Europa you will
find your way around easily. The Envelope is extremely flexible, and you can draw your own custom modulation
shapes by clicking and drawing in the display area. There are also a number of preset shapes that you can use as
starting points (or use as is). If you use Loop mode, you could turn the envelope into an advanced LFO and design
your own wave shapes.
The Envelope can then be used for modulating the Frequency parameter of the Phaser/Flanger/Filter section, for
modulating the Volume, and for modulating the LFO Rate.
• The envelope/loop playback starts as soon as there is audio present in Sweeper - or you can trigger playback
using the Audio Trig function (see “Audio Trig”) or a CV trig signal on the rear panel (see “Trig Envelope”) in-
stead.

742 SWEEPER MODULATION EFFECT

 Preset
1. Click the Preset button to bring up a palette of envelope preset curves:

2. Click the desired envelope preset curve to place it on the display.

Let’s select a standard ADSR style of envelope curve:

Adding and removing envelope points

D Double click, or hold down [Ctrl](win) or [Cmd](Mac) and click in the envelope display to add points to the
envelope curve:

D To remove a point, double click, or hold down [Ctrl](win) or [Cmd](Mac) and click, on an existing point on the
envelope curve.

Changing the envelope curve shape

D Click a line segment (between two points) and drag up/down to change the curve shape:

743 SWEEPER MODULATION EFFECT

 Looping the envelope
D Click the Loop button to turn the envelope into a kind of LFO.

Here we have edited a stepped curve from the Presets. We have also enabled Sync and set the rate to 4/4. This
means that each step in the curve now represents an 1/16th note.
• The loop playback is synced to the Reason sequencer and time line (so that the loop always start on the “one”)
and continues for as long as there is still audio present through the Sweeper device.

Editing levels only

D To restrict the editing to levels only, without affecting the time positions, click the Edit button:

In this mode you cannot change the time positions of the envelope points, only their levels (height). This is extra
useful with a stepped Preset curve, because dragging up or down will change the value of an entire segment, turn-
ing the Envelope into a pseudo-sequencer.
! To be able to adjust the level of a segment, the two points on either side of the segment have to be on the ex-
act same time positions. Otherwise, only the closest point will be changed. Also, any inclining/declining seg-
ment will automatically turn horizontal when edited:

Adjusting the level of a segment.

Creating “free form” envelope curves

In the Edit mode, you can also draw “free form” curves:
D To continuously add new consecutive points, hold down [Ctrl](win) or [Cmd](Mac) and drag in the envelope
display:

D To erase points, hold down [Shift] and [Ctrl](win) or [Cmd](Mac) and drag in the envelope display.

744 SWEEPER MODULATION EFFECT

 Audio Trig

It’s also possible to trigger the envelope from the audio running through Sweeper.
D Activate the Audio Trig function and set the Threshold value as desired.

When the audio level in Sweeper exceeds the set Threshold value, or quickly increases when above the Threshold
level, the envelope is triggered.
• If the envelope is not in Loop mode, one complete cycle is completed at the most. The cycle continues to play
back as long as there is audio present through the Sweeper device.
• If the envelope is in Loop mode, the loop plays back from the very beginning when an Audio Trig signal is re-
ceived. The loop continues to play back as long as there is audio present through the Sweeper device.

The Audio Follower Modulator

The other part of the Modulator section is the Audio Follower. This is basically an envelope follower, which tracks the
level of the audio running through Sweeper and outputs a control signal that can be used for modulating the Fre-
quency parameter of the Phaser/Flanger/Filter section, for modulating the Volume, and for modulating the LFO
Rate. The tracked (followed) audio level is shown in real-time in the display.

Gain In
Here you can attenuate or gain the modulation signal level, to adjust it to the audio signal level.

Attack
This controls how fast the envelope follower should react after the input signal level has increased from one value to
a higher.

Release
This controls how fast the envelope follower should react after the input signal level has decreased from one value to
a lower.

745 SWEEPER MODULATION EFFECT

 Connections

CV Input
Frequency
This CV input can be used for modulating the Frequency parameter in the Phaser/Flanger/Filter. The input accepts
bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Feedback/Reso
This CV input can be used for modulating the Feedback or Resonance parameter in the Phaser/Flanger/Filter. The
input accepts bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Spread
This CV input can be used for modulating the Spread parameter in the Phaser/Flanger/Filter. The input accepts
bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Dry/Wet
This CV input can be used for modulating the Dry/Wet parameter in the Phaser/Flanger/Filter. The input accepts
bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Trig Envelope
This CV input can be used for triggering the Envelope Modulator. The Envelope Modulator is triggered as soon as the
CV value goes from zero to a positive value.
If the Audio Trig function is active (see “Audio Trig”), the Trig Envelope function will co-exist with this.

CV Output
LFO
This sends out the LFO signal as a bipolar CV signal.

746 SWEEPER MODULATION EFFECT

 Fol/Env
This sends out the Audio Follower or Envelope signal as a unipolar CV signal.

Trigger
This sends out a unipolar CV trig signal as soon as the Audio Trig function (see “Audio Trig”) is triggered.

Input Left & Right

D Patch the audio signals you want to process here.
If your input signal is in mono, connect only to the L (left) input.
! Note that the result also depends on the current Routing setting (see “Routing”).

Output Left & Right

These are the stereo audio outputs.
! Note that the result also depends on the current Routing setting (see “Routing”).

747 SWEEPER MODULATION EFFECT

748 SWEEPER MODULATION EFFECT
Chapter 47
Synchronous
Timed Effect Modulator
 Introduction

The Synchronous Timed Effect Modulator device is a very flexible multi effect “loop” device with freely designable
effect parameter modulation curves. Synchronous features built-in Distortion, Filter, Delay and Reverb effects that
can be modulated simultaneously by up to three modulation curves.
By drawing your own unique modulation curves in the display and assigning these curves to the desired effect
parameters, you get a very flexible system for repeatedly modulating the effects parameters - in perfect sync with the
main sequencer.
The standard loop length is 2 bars. However, if you run Synchronous in half speed, you will get a maximum loop
length of 4 bars.
Synchronous is designed to be used mainly as an insert effect, on individual tracks, or as a “loop mangling” effect
hooked up to the outputs of a Dr Octo Rex or Redrum device, for example.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

750 SYNCHRONOUS TIMED EFFECT MODULATOR

 Panel overview
Below is a description of the different sections in Synchronous:

1
2

10

5 6 7 8 9

Synchronous panel sections

• 1. Patch selector (for browsing, loading and saving patches).
• 2. Curve design tools, rate, speed, phase, master offset and curve dimming controls.
• 3. Modulation curves display with selection, freeze, kill and loop length controls for each curve.
The display is where you draw your effect modulation curves.
• 4. Modulation control section.
Here is where you set how much each modulation curve should affect each effect.
• 5. Dist section.
Features two distortion effects, ring modulation, and a lo-fi effect.
• 6. Filter section.
Features lowpass, bandpass, highpass and comb filter types.
• 7. Delay section.
Features regular delay, ping-pong delay and roll (freezed delay) effects.
• 8. Reverb section.
Features a stereo reverb.
• 9. Level modulation section.
• 10. Master dry/wet and level controls.

751 SYNCHRONOUS TIMED EFFECT MODULATOR

 Using Synchronous
Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches”
and “Saving patches” for details.

Drawing and assigning modulation curves - a tutorial

To describe the basic principles with modulation curves, let’s have a look at an example of how you can design a
modulation curve and then have it modulate a couple of effect parameters:
1. Create a Dr Octo Rex device.
A Dr Octo Rex device is created and the default Rex loops are loaded in the device.
2. Select the Dr Octo Rex device in the rack and create a Synchronous device.
The audio outputs of the Dr Octo Rex device are automatically connected to the audio inputs of the Synchronous
device. The audio outputs of the Synchronous device are also automatically routed to the Dr Octo Rex’s Mix
Channel.
3. Select the Synchronous device in the rack and select “Reset Device” from the context menu or Edit menu.
All parameters in the Synchronous device are now reset to their default values. The yellow modulation curve is
automatically selected and reset to a straight line down in the display.
4. Click Play on the sequencer Transport Panel in Reason.
The default Rex loop in the Dr Octo Rex device starts to play back and the Position Indicator at the top of the
Synchronous display starts to advance in sync with the sequencer.

The Position Indicator advances in sync with the sequencer tempo.

5. Stop the sequencer.
6. Click the Positive Sawtooth Tool button and then the 1/8 Rate button:

7. Place the mouse pointer to the far left on the display, in the curve area.
8. Click-hold and then drag the mouse pointer to the right on the display.
9. Release the mouse button when you have reached the second thicker vertical grid line.
A sawtooth wave is now visible in the display. The rate of the sawtooth wave is 1/8th of a bar, since you selected
this when you clicked the 1/8 button. The amplitude of the sawtooth wave is increased or decreased according to
your vertical drawing direction.

752 SYNCHRONOUS TIMED EFFECT MODULATOR

 10.Now, click the Stepped Line Tool button and the Free button:

The Rate buttons have now disappeared from the panel, since they are of no importance when you use the Line
tools.
11.Click-hold the mouse pointer at the second thick line and then draw to the right throughout the rest of the dis-
play area.
Alter the amplitude (y-axis positions) as you draw.
12.Release the mouse button when you have reached the far right of the display.
Now, you have half a bar of a 1/8th sawtooth waves and one and a half bars of a stepped “free hand” curve:

Now, let’s assign the yellow modulation curve to the effect parameters to be modulated. In this example we will
assign the curve to the Freq and Resonance parameters of the Filter section:
1. Click the Filter button to switch on the Filter section:

2. Turn the Freq knob to a low value and the Resonance knob to a high value:

3. Raise the yellow Freq Modulation knob (above the Freq knob) to a positive value (past its 12 o’clock position).
Then, set the Resonance Modulation knob (above the Resonance knob) to a negative value.
Yellow modulation amount semicircles appear around the Freq Modulation knob and around the Resonance
Modulation knob, to indicate the modulation amounts and their “directions” (positive or negative modulation):

Setting the modulation amounts.

4. Start the sequencer.
Congratulations, you have now created your first looped Filter modulation in Synchronous!

753 SYNCHRONOUS TIMED EFFECT MODULATOR

 Editing modulation curves - a tutorial
Basically, there is one way you can edit an existing Modulation Curve: by redrawing the part of the curve you want to
change. The examples below describe two basic use cases.

Changing an existing curve’s amplitude

If you want to adjust the modulation curve’s amplitude anywhere throughout the modulation loop, proceed as follows:
1. Click the Curve Select button for the modulation curve you want to edit.
The curve is highlighted in the display.
2. Click the Tool button which corresponds to the existing curve in the display.
If the curve is a waveform, make sure you also click the correct Rate button.
3. Now, you have two options:
D To change the amplitude in a linear fashion (with a constantly increasing or decreasing amplitude), make sure
the Free button is deselected (off). Place the mouse pointer where you want the change to begin and then
draw to the right to where you want the change to end.
The amplitude starts at the level where you placed the mouse pointer and ends at the level where you released the
mouse button. The level change between the start and end point is linear.
D To change the amplitude individually between each vertical grid line, make sure the Free button is on. Place
the mouse pointer where you want the changes to begin and then draw to the right to where you want the
changes to end.
The amplitude starts at the level where you placed the mouse pointer, then follows your drawing directions, and
ends at the level where you released the mouse button. The levels changes are now individual between each
vertical grid line.
! Note that the shortest interval you can replace (or add) curves in is in 1/16th note “steps” of a bar, i.e. in be-
tween two vertical grid lines in the display (in Speed x 1 mode). The mouse pointer automatically “snaps” to
the closest leftmost vertical grid line when you draw.
q If you are not satisfied with your amplitude changes, just use the standard [Ctrl](Win)/[Cmd](Mac)+[Z]
command to undo the operation.

Replacing a part of a curve with another one

To replace the current modulation curve with a different curve anywhere in the modulation loop, use the same
principle as described above in “Changing an existing curve’s amplitude”. The only exception here is that you now
freely choose the new curve shape, regardless of the existing curve shape. You can replace an existing curve in as
many places throughout the loop as you like.
! Note that the shortest interval you can replace (or add) curves in is in 1/16th note “steps” of a bar, i.e. in be-
tween two vertical grid lines in the display (in Speed x 1 mode). The mouse pointer automatically “snaps” to
the closest leftmost vertical grid line when you draw.
q If you accidentally overwrite a curve part that you want to keep, just use the standard Undo command to undo
the operation.

754 SYNCHRONOUS TIMED EFFECT MODULATOR

 Panel reference
The display section

Tool buttons and the Free button

D Click the Stepped Line button when you want to draw a curve that has fixed levels between each vertical grid
line in the display:

The result when you draw will be a jagged curve with vertical steps.
D Click the Linear Line button when you want to draw straight lines between the vertical grid lines in the display:

D Click any of the Waveform buttons when you want to draw (repetitive) waveforms in the display:

In these situations, the Rate buttons define the waveform cycle lengths, see “Rate buttons”.
D Click the Free button in combination with a Line button to be able to freely define the amplitude between each
of the vertical grid lines in the display:

D Click the Free button in combination with a Waveform button to be able to freely define the waveform’s ampli-
tude between each of the vertical grid lines in the display:

Rate buttons

D Use the Rate buttons to define the waveform cycle length when you draw waveforms in the display.
You can alter between different time rates throughout the loop by clicking another Rate button and continue
drawing from where you stopped.
! When you have selected a Line tool, the Rate buttons automatically become invisible.

755 SYNCHRONOUS TIMED EFFECT MODULATOR

 Speed control

D Use the Speed x control to set the playback speed of the currently selected modulation curve.
The Speed can be set to 2, 1 or 0.5 times the main sequencer tempo, and are set individually for each of the three
modulation curves. The speeds are indicated by the Position Indicators’ advancement when the main sequencer is
running.
• In Speed x 1, the display covers 2 bars.
The vertical grid lines in the display represent 1/16th note.
• In Speed x 0.5, the display covers 4 bars.
The vertical grid lines in the display now represent 1/8th note.
• In Speed x 2, the display covers 1 bar.
The vertical grid lines in the display now represent 1/32th note.

Phase knob

D Set the phase of the currently selected modulation curve.

Range: 0-360 degrees.
The phase is changed individually for each curve type (waveform) used in the loop. The picture below shows the
same modulation curve with a Phase setting of 0 and ~180 degrees, respectively:

The same modulation curve at 0 degrees (top) and 180 degrees (bottom) Phase values.
As you can see in the picture above the first part of the curve, which is a sinewave with a 1/4th note rate is phase
shifted 180 degrees. The second part, which is a stepped line, is not changed at all. The third part of the curve,
which is a sawtooth wave with a 1/8th note rate is also phase shifted 180 degrees. However, since the third part
has a 1/8th note rate, the actual “distance” it has been moved is half as long as the 1/4th note rate sinewave in
the first part of the curve. This is because all waveform curves are phase shifted individually throughout the
modulation curve.
! Stepped Line and Linear Line curves are not affected by the Phase parameter - they stay fixed regardless of the
Phase value.
q If you want to “time shift” the entire modulation curves, including the Line curves, use the Master Offset
function described below.

756 SYNCHRONOUS TIMED EFFECT MODULATOR

 Master Offset knob

D Set the common offset for all three modulation curves.

This determines where in the modulation loop the playback should begin. Negative Master Offset values moves
the playback start position backwards, and vice versa.
! The Master Offset is indicated only by the Position Indicators - i.e. the curves are not displaced in the display.
Range: +/- 32 16th notes (in Speed x 1 mode).
Range: +/- 32 8th notes (in Speed x 0.5 mode).
Range: +/- 32 32th notes (in Speed x 2 mode).

Dim knob

D Turn the Dim knob to adjust the dimming amount of the currently unselected modulation curves.
At zero, all modulation curves are equally bright all the time.
At maximum level, only the selected modulation curve is visible in the display.
! Note that the Dim value is also saved with the patch.

Curve Select buttons

D Click one of these buttons to select the modulation curve you want to edit.

Setting the loop lengths

The three modulation curves can have their own individual loop lengths. The loop length for a modulation curve can
be adjusted, simply by repositioning its loop locator:
1. Click the Curve Select button for the modulation curve you want to edit.
2. Click-hold and drag the colored triangular loop locator horizontally to the new desired position.

Setting the loop length for the yellow modulation curve.

3. Release the mouse button to set the new loop length.
The loop locator will snap to the closest leftmost vertical grid line in the display. In Normal speed, this means it will
snap to a 1/16th note grid.
q By setting different loop Lengths for the three modulation curves, you can get very animated variations in the
modulations over longer periods of time.

757 SYNCHRONOUS TIMED EFFECT MODULATOR

 FRZ (freeze) buttons

1. Click a FRZ (freeze) button to stop the playback of the corresponding modulation curve and freeze its current
modulation value.
2. Click again to deactivate the freeze function.
The playback automatically continues at the position in the loop where it should have been if the freeze function
had not been activated.

Kill buttons

1. Click a Kill button to deactivate (mute) the corresponding modulation curve.

The modulation curve is preserved but does not modulate any parameter as long as the Kill function is active. Also,
the killed curve becomes invisible in the display.
2. Click again to deactivate the Kill function.
The modulation curve is activated and automatically continues with the modulation at the current position in the
loop.

Modulation controls

The Modulation Control knobs are used for setting a modulation amount (positive or negative) for the effect
parameter right below each Modulation Control knob. To set up a modulation, proceed as follows:
1. Select the modulation curve you want to use for the modulation by clicking its Curve Activate button.
All assignable Modulation Control knobs are automatically colored according to the selected modulation curve:

758 SYNCHRONOUS TIMED EFFECT MODULATOR

 2. Set the modulation amount for the corresponding effect parameter by turning the desired Modulation Control
knob.

Setting the modulation amount.

A colored semicircle appears around the Modulation Control knob. The semicircle shows how much the selected
modulation curve affects the corresponding effect parameter (below each Modulation Control knob) - and in which
direction (positive or negative).
D If you want the selected modulation curve to modulate other effect parameters, adjust the Modulation Control
knobs above the other desired effect parameters.
3. If you want other modulation curves to modulate the same - or other - effect parameters, click the desired
Curve Activate button and repeat the modulation assignment procedure from Step 1.
All Modulation Control knobs can be modulated by all three modulation curves at the same time if you like. The
modulation amounts and directions can be set independently for each modulation curve. The picture below shows
a Modulation Control knob which is affected differently by all three modulation curves:

Different modulation amounts set for each of the three modulation curves.

Dist section

Amount knob
D Set the distortion amount.
At 0, the signal is left unaffected.

Character knob
D Set the frequency content of the distorted signal.
The effect varies depending on the selected distortion type (see below).

759 SYNCHRONOUS TIMED EFFECT MODULATOR

 Distortion type selector
D Select distortion type by clicking the corresponding LED button.
In Ring Mod mode, the Character knob controls the ring modulator frequency.
In Lo-Fi mode, the Character knob controls the sample rate. There is no anti-aliasing filter in this algorithm, so
there will be a lot of nice lo-fi character to the sound.

Post Filter button

D Click the Post Filter button to route the Dist section after the Filter section (instead of before, which is the
standard routing).
The routing order can have a big impact on the frequency content in the sound. Since the Dist section adds
frequencies to the sound, you might want to preserve these by routing the Dist section after (Post) the Filter
section.

Dist On/Off button

D Click the Dist button to switch on/off the Dist effect section.
If switched off, the signal is bypassed unaffected.

Filter

Freq knob
D Set the cutoff frequency (for the HP, LP and Comb filter types) or center frequency (for the BP filter type).

Resonance knob
D Set the resonance amount of the filtered signal.

760 SYNCHRONOUS TIMED EFFECT MODULATOR

 Filter Type switch
D Click to select the desired filter type.
The filter types are:
• 12 dB highpass (HP):

Amplitude (log)

Resonance

Frequency
Freq (log)

• 6 dB bandpass (BP):

Amplitude (log)

Resonance

Frequency
Freq (log)

• 24 dB lowpass (LP):

Amplitude (log)

Resonance

Frequency
Freq (log)

• Comb filter with positive feedback:

Amplitude (log)

Resonance
Frequency
Freq (log)

761 SYNCHRONOUS TIMED EFFECT MODULATOR

 Lag knob
D Increase the Lag to get smoother frequency variations when the Freq parameter is modulated.
This works like a sort of “portamento” for the Freq parameter, i.e. the Freq parameter will move more slowly when
modulated if the Lag value is high.

Filter On/Off button

D Click to switch on/off the Filter effect section.
If switched off, the signal is bypassed from the Filter section.

Delay

Amount knob
D Set the amount of the delay signal.
At 0, the signal is left completely dry.

Time knob
D Set the time between the delay repetitions.
If the “Sync button” is on, the Time values can be stepped between time divisions (e.g. 1/1, 1/2,
1/4, 1/8 etc.) relative to the main sequencer tempo.

Feedback knob
D Set the feedback amount, i.e. the amount of repetitions, of the delayed signal.

Keep Pitch button

When you manually change the Rate during recording or playback, you will notice that the pitch of the delayed signal
also changes. If this effect is undesirable, you can enable Keep Pitch, which will ensure that the pitch remains fixed
regardless of changes in Rate.

Ping Pong button

With Ping Pong enabled, the stereo position of each delay repeat will alternate between left and right. The “Pan knob”
determines the stereo width as well as the position of the initial repeat. When the Pan knob is set to full Left, the first
delay bounce will be panned hard Left, the second will be panned hard Right, and so on. When the knob is set to full
Right, the order is reversed (R > L > R etc).

762 SYNCHRONOUS TIMED EFFECT MODULATOR

 Sync button
D Click the Sync button to sync the delay times to the main sequencer tempo.
Set the time divisions (e.g. 1/1, 1/2, 1/4, 1/8 etc.) with the “Time knob”.

Roll button
The Roll function works like a “freezed” delay, perfect for stutter, repeat and glitch effects. When the Roll button is on,
and you turn up the Feedback knob, the input signal to the Delay section is gradually suppressed, while the feedback
is automatically raised internally. In Roll mode, the Amount knob controls the level of the delay signal when the
Feedback parameter is turned up. When the Feedback is set to zero, the Amount value is disregarded.
1. Click the Roll button to switch from the regular delay settings to Roll mode.
2. Turn the Feedback knob to set the mix between the rolled delay signal and the dry signal.
D To get an instant stutter/repeat effect, turn the Feedback knob up quickly from zero at the point where you
want the stutter effect. Then, quickly turn the Feedback knob back to zero again when you want the effect to
disappear.
q Experiment by changing the Time parameter to get different “stutter” times.

Pan knob
D Use the Pan knob to set the stereo position of the delay repetitions. In Ping Pong mode, the Pan parameter de-
fines where the initial delay bounce should be placed in the stereo panorama, see “Ping Pong button”.

Send/Return switch
D Select whether you want the Amount knob to control the Send level or the Return level of the delay effect.
The picture below shows the different configurations schematically:

Send Return Send Return

Amount Amount
Delay Delay

Feedback Feedback

Audio In Audio Out Audio In Audio Out

The schematic placement of the Amount knob in Send and Return mode, respectively.

Delay On/Off button

D Click to switch on/off the Delay effect section.
If switched off, the signal is bypassed from the Delay section.

763 SYNCHRONOUS TIMED EFFECT MODULATOR

 Reverb

Amount knob
D Adjust the balance between the unprocessed and the reverberated audio signal.

Decay knob
D Set the decay time of the reverberated signal.

Size knob
D Set the emulated room size, from small room to large hall, with the Size knob.
Lowering this parameter results in a closer and gradually more “canned” sound. Raising the parameter results in a
more spacey sound, with longer pre-delay.

Damp knob
D Set the high-frequency damping amount of the reverberated signal.
Raising the Damp value cuts off the high frequencies of the reverb, thereby creating a smoother, warmer effect.

Send/Return switch
D Select whether you want the Amount knob to control the Send level or the Return level of the reverb effect.
The picture below shows the different configurations schematically:

Send Return Send Return

Amount Amount
Reverb Reverb

Decay Decay

Audio In Audio Out Audio In Audio Out

The schematic placement of the Amount knob in Send and Return mode, respectively.
q To create gated or “reversed” reverb effects, set the switch to Return and modulate the Amount Modulation
parameter from a modulation curve.

Reverb On/Off button

D Click to switch on/off the Reverb effect section.
If switched off, the signal is bypassed from the Reverb section.

764 SYNCHRONOUS TIMED EFFECT MODULATOR

 Level

Level knob
D Set the default level.
Range: - INF to + 12 dB.
! If the Level is modulated by a modulation curve, the volume will increase above the default level, according to
the modulation amount.

In & Out switch

D Select if the Level control should be routed before or after the rest of the effect sections in the signal chain.

Master Controls

Dry/Wet
D Set the balance between the dry input signal and wet signal for the entire effect chain.
At 0 the effects are completely bypassed and the input signal is passed through Synchronous unaffected.

Master Level
D Set the master output level from Synchronous.
Range: - INF to + 12 dB.

765 SYNCHRONOUS TIMED EFFECT MODULATOR

 About automation of display section parameters
As with other Reason and Rack Extension devices it is possible to automate most of the panel parameters and
control them from parameter automation lanes in the main sequencer. Besides the regular panel parameters in
Synchronous, parameters in the display section can also be automated:

The parameters in the display section that can be automated are:

• The Master Offset control.
• The Speed control for each modulation curve.
• The Phase control for each modulation curve.
• The Loop Locator for each modulation curve.
• The FRZ (Freeze) and Kill buttons for each modulation curve.
• The Modulation Control knobs for each modulation curve.

766 SYNCHRONOUS TIMED EFFECT MODULATOR

 Connections

CV In
Curve 1/2/3
These CV inputs accept bipolar control signals. Each input CV signal is added to the corresponding Modulation Curve
signal and the resulting signal then modulates the assigned effect parameters.
The input signals can be attenuated with the corresponding attenuation knobs.
! If the resulting modulation curve value should be negative (below zero), the assigned effect parameters are
modulated with reversed polarity.

Freeze 1/2/3
CV signals with levels >0 patched to these inputs will activate the Freeze function for the corresponding Modulation
Curve. When the CV signals drops to 0 or below, the Freeze function is deactivated. See “FRZ (freeze) buttons” for
more information.

Master Level CV In
A CV signal on this input can modulate the “Master Level” parameter. You can attenuate the input CV signal with the
attenuation knob.

767 SYNCHRONOUS TIMED EFFECT MODULATOR

 CV Out
Curve 1/2/3 and Curve 1/2/3 Inverted
These CV outputs send out positive unipolar control signal levels according to the shapes of Modulation Curves 1-3
respectively.
The “Inverted” CV outputs below each “regular” CV output send out inverted negative unipolar control signal levels
according to the shapes of Modulation Curves 1-3 respectively.
The picture below shows a Curve signal (dark red) and its corresponding Curve Inverted signal (orange):
CV Out level
Curve
+

0
Time
-
Curve Inverted
q By patching one of the positive control signals to one destination and the corresponding inverted negative
control signal to another destination, you are able to control the parameters in a “mirrored” fashion. For
example, patch the positive CV signal to the Filter Cutoff modulation of a synth device and the negative signal
to the Resonance modulation of the same synth.

Audio In L&R
D Patch the audio signals you want to process here.
If your input signal is in mono, connect only to the L (left) input.

Audio Out L&R

These are the stereo audio outputs.

768 SYNCHRONOUS TIMED EFFECT MODULATOR

Chapter 48
The Echo
 Introduction

The Echo is an advanced stereo in/out echo and delay device with a multitude of parameters for tweaking the color
and shape of the echo effect - diffusion, filtering, distortion and more. In addition to the normal mode where The Echo
behaves like a regular send or insert effect, there are two additional modes called Triggered and Roll which let you
automate momentary echo effects as well as create interesting stutter and repeat effects on the fly. The Echo also
features breakout jacks which allow you to insert any number of other effect units into the feedback loop - this opens
up endless possibilities for creative sculpting of the echo repeats.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

About the Patch format

The Echo features programmable effect presets. Included are a number of Factory Patches which can be used as-is
or provide you with a good starting point for further tweaking. The Patches use the file extension "*.echo". Loading
and saving Patches is done in the same way as for instrument devices.

Parameters
The Echo is comprised of six main sections; Mode (Normal, Triggered and Roll), Delay (time, tempo sync and stereo
behavior parameters), Feedback (including a Diffusion subsection), Color (Drive and resonant Filter), Modulation (En-
velope, Wobble and LFO) and Output (Dry/Wet control and Ducking).

Common effect device parameters

While the parameters specific to The Echo are described in detail below, some features and procedures are common
to all effect devices. Please refer to “Common effect device features” for further info about the Bypass On/Off
switch. It should however be noted that the Input Meter on The Echo differs from the standard 7-segment LED meter
found on other effect devices in Reason; The Echo features a 3-segment meter with the usual green, yellow and red
LED (the latter indicates clipping).

770 THE ECHO

 The Mode section
The Mode section acts as a signal gate with three different ways of passing the input signal on to the subsequent
main sections.

Mode
This switch has three positions:
• Normal
The standard send/insert effect behavior where the input signal is fed continuously into the device.
• Triggered
This mode keeps the signal unprocessed until you hit the Trig button. This mode is for situations where you only
want the echo effect momentarily, e.g. on every 4th snare hit, or individual words on a vocal track.
• Roll
In Roll mode, the unprocessed signal passes through unaffected until you turn up the Roll slider, see “Roll slider”.
This gradually suppresses the Dry signal while simultaneously raising the Feedback and mixing the delay/echo
output into the Wet signal. The Roll mode is useful for repeat, stutter and glitch effects. For Roll mode, we recom-
mend that the Dry/Wet parameter in the Output section (see below) is set to (or close to) Wet.
q See “Using the Roll function” for an example of how to use the Roll function.

Trig
This button is only functional when the Mode switch is set to the Triggered position. When you press the Trig button
it opens the Input signal gate, which stays open until you release the button again. Think of it as momentarily enabling
an effect send.

Roll slider
This slider is only functional when the Mode switch is set to the Roll position. The Roll slider does three things:
• Turns up the feedback (internally) to unity (100%) or to the Feedback setting on the front panel, whichever is
higher.
This starts happening during the slider throw, to catch a good amount of input signal.
• Closes the input to the delay (you don't want the delay to continue catching the input sound during the roll).
This happens late, with a slight delay - meaning it catches a little bit of sound after you've hit the Roll position.
• Changes the internal mix from dry to wet.
This also happens a bit late, so that if you hit a drum beat perfectly with the roll, you will hear that beat (dry) before
the delay repeats (roll).
! Note that the "internal mix" mentioned above is the signal sent to the "Wet" channel in Roll mode. The "Dry"
channel still carries the dry signal at all times.

771 THE ECHO

 The key thing here is that you shouldn't set the slider to a value in between - it should go from 0 to full Roll and back.
So why do we use a slider and not a switch? Well, things start happening during the slider throw (and actually a bit af-
ter you've moved the slider fully to the right).
The slider allows for some sloppiness, making it easier to catch a beat and roll it fully. The result is slightly different
depending on how fast you move the Roll slider - in most situations it works best to move it pretty fast and hit Roll ex-
actly on the beat.
q Although you will probably most often set The Echo to full Wet when Rolling, it might make sense to have a lit-
tle bit of dry signal leaking through during the roll (making it easier to keep time and return from the roll ex-
actly on the beat). Then you just set Dry/Wet balance to something like 85% Wet, see “Dry/Wet”.
q See “Using the Roll function” for an example of how to use the Roll function.

The Delay section

The Delay section features parameters relating to delay time, tempo sync and stereo behavior.

Time
This knob controls the delay time. The delay time range is 1 to 1000 milliseconds. When Sync is enabled, the range
is 1/128 notes up to ½ note.

Offset R
This unipolar knob controls the Right channel delay time offset. The higher the Offset R value, the more the Right
channel will be delayed in relation to the Left channel - perfect for creating stereo delay effects!
The Offset range is 1 to 1000 milliseconds. When Sync is enabled, the Offset range is “no offset” up to ½ note (ac-
cording to the same resolution table as for the Time parameter). However, the 1/128 notes offset value is replaced
by “no offset”.

Keep Pitch
When you manually change the Delay time during recording or playback, you will notice that the pitch of the delay
signal also changes. If this effect is undesirable, you can enable Keep Pitch, which will ensure that the pitch remains
fixed regardless of changes in Delay time.

Sync
This button enables Tempo Sync, which affects the Time and Offset R parameter.

772 THE ECHO

 Ping-Pong
With Ping-Pong enabled, the stereo position of each delay repeat will alternate between left and right. The Pan knob
determines the stereo width as well as the position of the initial repeat. When the Pan knob is set to full Left, the first
delay bounce will be panned hard Left, the second will be panned hard Right, and so on. When the knob is set to full
Right, the order is reversed (R > L > R etc).

The Feedback section

The Feedback section consists of a main section with Feedback and Offset R controls, as well as a Diffusion subsec-
tion which allows you to add a kind of ‘smearing’ effect to the echo.

Feedback
The Feedback knob sets the amount of feedback, i.e. the amount of wet signal fed back into the delay. This deter-
mines the number of repeats. At zero feedback there will only be a single repeat. Unity gain is achieved at 100%. If
you increase the feedback beyond this it will increase the gain so a distorted signal is produced.

Offset R
This bipolar knob controls the offset in delay Feedback on the right channel. By default, both channels have the same
amount of feedback (as determined by the Feedback parameter), but the Offset R knob allows you to add or subtract
feedback separately for the Right channel only. The practical result is that you will hear the echo gradually wander
from the center to the left or right side.
This control combined with the Offset R in the Delay section can be useful for controlling the length of the effect for
the right channel.

Diffusion
Diffusion introduces kind of a “smearing” effect, somewhat reminiscent of diffusion on a reverb. Raising the Amount
will introduce additional delay repeats very near the ‘original’ repeats, and raising the Spread value will spread these
repeats out wider.

773 THE ECHO

 The Color section

This section features a distortion/limiter and a resonant filter. Each echo repeat is colored before being fed back into
the loop, meaning that the distortion and filter effects will be more pronounced with each repeat.

Drive
This knob sets the amount of the selected distortion/limiter effect.

Type
The Type Switch lets you select between 4 different effect algorithms:
• LIM (Limiter)
This produces that typical “compression” effect which you would get from analog limiters.
• OVDR (Overdrive)
This produces an analog-type overdrive distortion effect.
• DIST (Distortion)
Similar to overdrive, but denser.
• TUBE
This emulates tube distortion.

Filter
This is a resonant bandpass filter, hence it lets you create filter effects where either the lower or the higher frequency
range is cut (or boosted, in case the Reso control has been turned up).

Freq
This control sets the change in frequency. For each delay repetition, the frequency content will shift in accordance
with the frequency setting.

Reso
This knob determines the amount of resonance on the delay repetitions. A different frequency range will be amplified
depending on the setting of the Freq parameter.

774 THE ECHO

 The Modulation section

This section features parameters for modulating the pitch and stereo image of the echo effect.

Env
The envelope parameter lets you create a kind of bend effect where the pitch of the echo repeats wanders down or
up, depending on whether you turn the knob left or right. The knob is bipolar, meaning that there is no Env effect in
the default middle (zero) position.

Wobble
This emulates a tape speed wobbling effect where the speed of the “tape” (and consequently, the pitch of the signal)
wobbles randomly.

LFO
The LFO subsection modulates the pitch of the left and right channels independently, meaning that it functions as a
kind of stereo spread at modest settings and warps the echo completely at heavier settings.

Rate
This knob sets the speed of the LFO.

Amount
This knob sets the amount of LFO.

775 THE ECHO

 The Output section

This section is the final output stage where the processed effect signal goes through the standard Dry/Wet control,
as well as the optional Ducking control.

Dry/Wet
This is a traditional dry/wet parameter for controlling the relation between the unprocessed and the processed signal.
When Roll mode is enabled (see the Mode section for more info), we recommend that the Dry/Wet control is set to
Wet only even when The Echo is used as an insert effect.

Ducking
Ducking attenuates the level of the Wet (processed) signal until the amplitude of the Dry signal drops, at which time
the Wet signal is faded back in. This is useful for adding a delay effect to the silence that comes after you have
played your lovely lead line. The delay will not be heard while you are still playing so you will avoid muddling up the
solo.

776 THE ECHO

 CV/Gate inputs

On the back of The Echo you will find the following CV inputs:

Trig
This is a Gate input for controlling the Trig function in the Mode section.

Roll
Use this for dynamically changing the Roll amount (corresponding to the Roll slider on the front panel) in the Mode
section.

Delay Time
Use this for dynamically changing the Delay Time in the Delay section.

Filter Freq
Use this for dynamically changing the Filter Frequency in the Color section.

The Breakout Jacks

The Echo features special Breakout jacks which allow you to insert other effect devices into the feedback loop. The
signal is processed externally and then fed back into the loop, meaning that each delay repeat will be more colored
by the effect(s) connected to the breakout jacks.

777 THE ECHO

 Tips and Tricks
Using the Roll function
Here's a quick example of how you can typically use the Roll function in The Echo:
1. Have some signal running through The Echo, e.g. a drum loop.
2. Set The Echo to Roll mode and full Wet output (see “Dry/Wet”).
The Roll slider is at 0. You will hear the drum loop unprocessed, since Roll isn't engaged yet.
3. On the beat you want to roll (or freeze, to use another term), move the Roll slider quickly all the way to the
right.
This replaces the dry signal with that beat, being rolled at whatever delay time you set.
q A nice thing is to set the Delay to Sync mode (see “Sync”) so the repeats will be in sync with the loop, e.g. to
1/16th notes or other desired resolution.
4. To go back to the (dry) loop, move the Roll slider quickly back to 0.

Creating “pitched” delay

1. Set the delay time to minimum (1ms).
2. Increase feedback to 100%.
3. Enable Diffusion and increase the Amount to maximum.
4. Play a drum loop or similar though The Echo.
Now you have set up a sonic playground that equals no other delay! Experiment with Diffusion Spread, Modulation
Env and Delay Time to achieve various interesting sonic results.

Distorted external feedback

1. Connect an instrument device to the input(s) of The Echo.
2. Connect a Scream 4 device in the feedback loop.
Connect the Scream 4 inputs and outputs to the Breakout jacks of The Echo, see “The Breakout Jacks”.
3. Play the instrument device and try different distortion algorithms on the Scream 4.
Be careful with the master volume in Scream 4 as this is very sensitive to the feedback level in The Echo.

778 THE ECHO

Chapter 49
Half-Rack Effects
 Common effect device features
While the specific parameters for each effect device are described below, some features and procedures are com-
mon to all effect devices:

The Input meter

This shows the level of the incoming audio signal, giving you an indication of which devices are active, connected and
playing. However, you don’t need to worry about clipping in effect devices, even if the meter goes into the red.

The Power/Bypass switch

This is located in the upper left corner of each effect device. The switch has three modes, according to the following
figure:

| Mode | Description
Bypass In this mode, the input signal is passed directly to the audio output, without being affected by the effect device.
This is useful when the effect device is connected as an insert effect, and you want to compare the effect
sound with the dry sound.
On This is the default mode, in which the device processes the incoming signal.
Off In this mode, the effect device is turned off and neither dry nor effect sound is sent out. This is useful when the
device is connected as a send effect and you want to turn it off temporarily.

About making settings

You adjust effect parameters using the regular editing techniques.
q A quick way to reset the parameters to their default values is to hold down [Ctrl](Win) or [Cmd](Mac) and click
the corresponding knob.

About Connections
• All effect devices have stereo inputs and outputs, and can be connected as send effects or as insert effects.
However, some effects are best used as one of these only. This is stated for each effect on the following pages.
See also the section about the signal flow graphs below.
• Most of the effect devices have one or several CV inputs on the back panel.
These allow you to control various effect parameters in real-time, from another device in the rack. See “CV/Gate
signals” for details about routing CV.

780 HALF-RACK EFFECTS

 The Signal Flow graphs

On the back of each effect device, you will find two or three small “graphs”. These indicate how the effect device han-
dles mono and stereo signals, depending on the connections. The selection of graphs for a device tells you how it
should be used, according to the following rules:

| Graph | Description
Can be connected as a mono-in, mono-out device.
(Of course, all effect devices can be connected in mono. However, if this graph isn’t shown for a device, this
means that a mono-in, mono-out connection may not give the proper results).

Can be connected as a mono-in, stereo-out device. This means that the device creates some sort of stereo ef-
fect (e.g. a reverb) or a mono effect that can be panned.

If you connect both inputs and outputs in stereo, the two sides will be processed independently (dual mono pro-
cessing).

If you connect both inputs and outputs in stereo, the two sides are summed before the effect processing. How-
ever, the actual effect is in stereo (and the dry signal will remain in stereo, if it is passed through the effect).

“True stereo” processing, or “stereo in - stereo out” processing. When you connect the inputs in stereo, each
channel in the effect uses the signal information from both inputs. However, the inputs are not summed - the
two channels are processed differently.
This mode is available on the RV7000 Advanced Reverb - see “RV7000 Mk II Advanced Reverb”.

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 DDL-1 Digital Delay Line

This is a mono delay (where the output can be panned in stereo) that can be synchronized to the song tempo. The
delay can be used as a send effect or an insert effect.

Parameters
| Parameter | Description
Delay time The display to the left on the device panel shows the delay time, either as note value steps (based on the sequencer
tempo and the Step Length parameter) or in milliseconds, depending on the setting of the Unit switch.
The maximum delay time is two seconds (2000 ms) while the maximum number of steps is 16.
Note that if the tempo is low, you may reach the maximum delay time at a lower number of steps than 16 (in which
case raising the steps value will not make any difference).
Unit This is where you select whether you want a tempo-based delay (“Steps” mode) or a free time delay (“MS” mode).
In the Steps mode, you specify the delay time in note value-based steps. This means that if you change the tempo in
the transport panel, the delay will maintain its rhythmic relation to the music (provided that the resulting delay time
doesn’t reach the maximum value). This mode is useful for creating rhythmic patterns.
If you change the tempo when using the delay in MS mode, the delay time will remain the same.
See also the note about switching Unit modes below.
Step length Governs whether each step in Steps mode should be a sixteenth note (1/16) or an eighth triplet note (1/8T).
Feedback Determines the number of delay repeats.
Pan Pans the delay effect to the left or to the right.
Wet/Dry If you are using the delay as an insert effect, you use this parameter to adjust the balance between the unprocessed
audio signal (dry) and the delay effect (wet).
If the delay is used as a send effect, this should be set all the way to wet only, since you can control the balance by us-
ing the AUX send controls in the Mixer.

CV Inputs
The following CV inputs are available on the back panel of the device:
D Pan CV.
This allows you to control the panning of the delay signal. Connect an LFO to this for moving delay effects, or use
a Matrix pattern to simulate random delay panning.
D Feedback CV.
This allows you to control the amount of feedback (the number of delay repeats) from another device. Useful for
dub-type echoes on certain beats or notes only.

Switching between Unit modes

When you switch between the two Unit modes (Steps and MS), the following rules apply:
D If you switch from Steps mode to MS mode, the delay will be set to the same actual delay time as was used in
the Steps mode.
This means that you can set up an exact rhythmic delay in Steps mode, and then switch to MS mode to adjust it
slightly.
D If you switch from MS mode to Steps mode, the delay is reset to the previously used Steps value.

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 CF-101 Chorus/Flanger

The CF-101 is a combined chorus and flanger effect. It adds depth and movement to the sound by adding a short
modulated delay to the audio signal. The delayed signal is then mixed with the original (either in the effect device or
manually by you - see below). The CF-101 can be used as an insert or send effect.

Parameters
| Parameter | Description
Delay This is a manual control for the delay time used to create the chorus/flanger effect. Usually, flanger-type effects use fairly
short delay times while chorus-type effects use medium long delays.
Feedback This governs the amount of effect signal fed back to the input, which in turn affects the intensity and character of the ef-
fect. Turning this to the extreme left (negative feedback) or right (positive feedback) produces different flanger effects
with a pronounced resonance “tone”, while keeping it in between produces a more gentle chorus effect.
LFO Rate This is the frequency of the LFO modulating the delay time. The higher the value, the faster the sound will oscillate.
LFO Sync This button lets you activate/deactivate LFO sync. When it is activated, the frequency of the LFO is synchronized to the
song tempo, in one of 16 possible time divisions. The LFO Rate knob is then used for setting the desired time division.
Turn the knob and check the tooltip for an indication of the time division.
LFO Mod Amount This determines the depth of the LFO modulation, i.e. by how much the delay time should be modulated. If you set this to
0, the effect will be “frozen” (most effective if you add some feedback).
Send Mode This determines whether the delayed signal and the dry signal should be mixed in the effect device or not. If you use CF-
101 as an insert effect, you should turn this off - the device will then output a mix of the dry signal and the modulated de-
lay signal.
If you use the device as a send effect, you should activate Send mode. Then, the device will only output the modulated de-
lay signal, allowing you to mix it with the dry signal using the AUX send controls in the mixer. See also the note below
about using the CF-101 as a vibrato effect!

CV Inputs
The following CV inputs are available on the back panel of the device:
D Delay CV.
Allows you to control the delay time from another device. This may give best results if you turn off the LFO modu-
lation in the device (turn LFO Mod Amount to zero). For example, by controlling the delay parameter from a Matrix,
you can create “stepped flanger” effects, in sync with the tempo.
q If you use the Delay CV input for “playing” the feedback tone, note that a higher delay value gives a lower
pitch.
D Rate CV.
Lets you control the rate of the modulating LFO from another device.

About Stereo and Mono connections

The CF-101 can be connected using mono or stereo inputs, as shown by the graphs on the back panel. Note the fol-
lowing:
D A “moving” stereo effect is only obtained when you use a mono input and stereo outputs.
With a stereo input, the two sides are processed in parallel, maintaining the stereo image of the original sound.

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 D When you are using a mono input and stereo outputs, there will only be a stereo effect if the internal LFO is
used.
If you turn LFO Mod Amount to zero, both stereo outputs will carry the same signal (mono). This is because the
“fake stereo” effect is produced by inverting the modulation for one of the output channels.

Tip: Using the CF-101 as a vibrato effect

The Send mode is intended for when using the CF-101 as a send effect. In this mode, the device will only output the
modulated delay signal - you get the actual “chorusing” by mixing this signal with the dry, unprocessed signal in the
Mixer.
However, if you activate Send mode while using the device as an insert effect, the result will be a pitch modulated
version of the original sound - in short, a vibrato effect. Along with a little feedback, this can be used for special ef-
fects.

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 Spider Audio Merger & Splitter

The Spider Audio Merger & Splitter is not an effect device, but a utility. It has two basic functions:
D To merge up to four audio input signals into one output.
D To split one audio input signal into four outputs.
There are no controls on the front panel of this device, only signal indicators.

Merging audio

Four audio input pairs Merged signal outputs

On the back panel of the Spider are several audio connectors. The left half of the panel contains four stereo audio in-
put connectors, and to the right of these, one merged stereo output.
D The principle is simple; all audio signals connected to any of the four inputs will be merged and output via the
output connectors.
If you connect a mono signal (to a L/Mono input, with nothing connected to the corresponding R input) it will be
output on both merged outputs. This way you can merge stereo and mono signals freely.
If you connect a signal to the R input only (with nothing connected to the corresponding L/Mono input) it will be
output on the R output only.

Practical uses of merging audio

There are many practical uses of merging audio signals together, for example:
D Process several audio signals with the same insert effect(s).
Perhaps you want to process certain channels in a mix with the same compressor, or use one ECF-42 to filter a
group of instruments in a mix. You can also set up a chain of insert effects and process the merged signals.
D Sub-grouping signals.
It may be practical to control several audio signals using one channel strip in the Mixer.
D Use merged signals as either carrier or modulator source for the BV512 Vocoder.
You could use several sounds as carrier signal, or modulate the carrier with several signal sources.

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 Splitting audio
The right half of the back panel contains two signal splitters, labeled “A (L)” and “B (R)”. The two splitters work inde-
pendently, in the following way:
D The signal fed to the input connector will be simultaneously output by all four outputs.
For splitting stereo signals you simply use both splitters with “A” for the left channel and “B” for the right channel.

Audio inputs Four split output pairs

Practical uses of splitting audio

There are many practical uses of splitting audio signals - here a few examples:
D Create “pseudo” stereo effects from mono signals.
For example, you could route the mono output of a Subtractor to the Spider and then send two split outputs (from
the same row) to different effects and on to different Mixer channels panned left and right.
D It provides a way to instantly switch between (or mix) different variations of the same signal.
This is a neat way of applying “spot effects” in a mix. An instrument output is split and sent to three different com-
binations of insert effect processing. The outputs from the three effects are routed to separate channels in the
Mixer, which could in turn have different send effects, eq, etc. You then have three different variations of the same
signal that can be easily switched in and out of the mix for drastic sonic changes - or combined for huge layered
sounds.

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 Spider CV Merger & Splitter

The Spider CV Merger & Splitter is not an effect device, but a utility. It has two basic functions:
D To provide one merged CV output from up to four CV input sources.
D To split CV or Gate inputs into several outputs.
Two inputs, A and B, are provided, each with four outputs, where one of the outputs will invert the polarity of the
control signal. One reason for having two splitable inputs is to make it possible to split Gate and Note CV, to con-
trol several instrument devices with one Matrix for example.
There are no controls on the front panel of this device, only CV signal indicators. The four horizontal indicators light
up to indicate signals connected to the corresponding merge input. The two indicators to the right indicate signals
connected to the corresponding split inputs.

Merging CV
Four CV inputs with The merged CV
trim controls signal output.

On the back panel of the Spider there are several CV connectors. The left half of the panel contains four CV/Gate in-
put connectors with associated trimpots, and to the right of these, one merged CV output.
D The merged CV output will produce a CV signal that represents the “sum” of all connected CV inputs.
A few things to note:
• Gate CV signals typically trigger notes or envelope cycles and are normally routed to a Gate input.
• CV signals typically control note pitch or for modulating parameters and are typically routed to CV Note or
Modulation inputs.
There are no strict rules involved, but the facts mentioned above means that it is generally better to stick to using ei-
ther Gate CV signals or CV signals but not a mixture when merging. simply because the CV/Gate signals usually go
to different input destinations.
For instance, merging Note CV and Gate CV from a Matrix does not make much sense if you want to use Matrix to
play melodic patterns via the Sequencer Control inputs of an instrument device. There would only be one merged out-
put whereas the instrument device would need a separate Gate and Note CV signal to work properly.

787 HALF-RACK EFFECTS

 Practical uses of merging CV
The practical applications of merging CV are perhaps less obvious compared to splitting CV. But there are numerous
applications for a merged CV control output, a few of which are listed below:
D You can create interesting modulation effects by merging several Modulation outputs from LFO’s and other
CV modulation sources.
For example, merging the Modulation outputs from several LFO’s would produce a “mixed modulation” output. This
merged output signal could be likened to a “super LFO” capable of generating several modulation cycles simulta-
neously, each with a different waveform and modulation rate! In addition to this, by using the trim control for each
CV input, you have full control over the amount of modulation applied by each LFO.
The above example could of course also include Curve CV outputs from a Matrix or Mod Outs from Malström etc.,
in short any CV Modulation output.
D Use the ECF-42 Filter to apply envelope controlled filter effects.
This can create the sound of “synthesized” percussion, and other interesting effects.
This is done using the following method:
1. Connect the audio outputs of a Redrum to a ECF-42 Envelope controlled filter.
2. Connect the Gate outputs from up to 4 Redrum drum channels to the merge inputs of a Spider CV.
3. Route the merged output to the Env Gate input on the ECF-42.
If you add a touch of velocity the connected Gate signals will trigger the ECF-42 filter envelope. Again, the trim
pots on the Spider allows you to adjust the amount of filter envelope applied.
D Create an “arpeggiator” using two Matrix devices and the Spider CV Merger & Splitter.
By merging the Note CV output from one Matrix with a Curve CV output of another Matrix, you can transpose the Ma-
trix pattern in real-time, a bit like an arpeggiator.
1. Create a Subtractor and a Matrix device.
Connect the Matrix Note and Gate CV outputs to the Subtractor Sequencer CV and Gate inputs, respectively.
2. Program a pattern for the Matrix.
In the following text this is referred to as “Matrix 1”.

788 HALF-RACK EFFECTS

 3. Now create a Spider CV and a second Matrix device and connect them as in the picture below.

Note that the Note CV output from Matrix 1, and the Curve CV output of Matrix 2 should be connected to the Spider. The merged
output is connected to the Sequencer Control Note CV input on the Subtractor.
4. On the Spider CV, turn the trimpot for the input connected to the Note CV output fully to the right.
This setting will retain the correct pitch relationship for the notes played by the pattern.
5. On the Spider CV, turn the trimpot for the input connected to the Curve CV output to “32”.
This will produce a Curve CV output that corresponds to semitone steps.

789 HALF-RACK EFFECTS

 6. Set the Curve type switch to “Bipolar” on the back of the second Matrix (Matrix 2).
7. Flip the rack around so that the front panels are showing, and make the following settings for the “Matrix 2”:
• Set the number of steps to “1”.
• Set the Curve/Keys switch to “Curve”.
8. Adjust the Matrix 2 curve for step 1 (the only step used) so that it is in the middle of the bipolar curve as the pic-
ture shows.

9. If you now activate Play from the transport, the pattern you programmed for Matrix 1 is played back. By care-
fully adjusting the Matrix 2 Curve step 1 up or down the Matrix 1 pattern is transposed in semitone steps.
By programming different values for the “pattern” played by Matrix 2 and saving them in different pattern locations,
you can use the Pattern selectors to transpose the Matrix 1 pattern to different keys!

Splitting CV
Two CV Split Inputs (A & B)

Each of the two Split inputs provide four

Split outputs. The lower right Split out-
puts will produce an inverted CV signal.

On the right half of the back panel you will find two split inputs “A” and “B”, each with four output connectors. The sig-
nal connected to a Split input will be output by all four corresponding outputs, where one is inverted.

Practical uses of splitting CV

There are many practical uses of splitting CV signals - here are a few examples:
D Connecting the CV Note and CV Gate outputs from a Matrix to Split Input A and B, allows you to connect the
Matrix to several instrument devices.
Simply route the CV and Gate outputs to the corresponding Sequencer Control CV/Gate inputs on the instrument
devices. Although this could also be done by copying the Matrix Pattern data to several sequencer tracks and rout-
ing the outputs to the desired devices, the advantage by using Split is if you are editing Matrix pattern data this will
be immediately be reflected in all the connected devices, without any copy/paste operations.
D Splitting modulation outputs from LFO’s, Curve CV data etc. allows you to apply modulation from one source
to several parameters.
By using the inverted output, you can create interesting modulation crossfades, where one parameter value rises
and another parameter value is lowered for example.

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 RV-7 Digital Reverb

! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.
Reverb adds ambience and creates a space effect. Normally, reverb simulates some kind of acoustic environment
such as a room or a hall, but you could also use it as a special effect.
• The Reverb device can be used as a send effect or an insert effect.
If several devices uses the same type of reverb, you should connect the reverb as a send effect, to conserve com-
puter power.

Parameters
The display to the left on the panel shows the selected reverb algorithm - the general type of reverb. By clicking the
arrow buttons you can change algorithm, with the following options available:

| Algorithm | Description
Hall Emulates a fairly large, smooth hall.
Large Hall Emulates a larger hall, with pronounced pre-delay.
Hall 2 A hall reverb with a brighter attack than “Hall”.
Large Room Emulates a large room with hard early reflections.
Medium Room Emulates a medium-sized room with semi-hard walls.
Small Room A smaller room, suitable for “drum booth”-type reverbs.
Gated A gated reverb, that is abruptly cut off.
Low Density A thinly spaced reverb, where you clearly can here the individual echoes. Useful for strings and
pads and as a special effect.
Stereo Echoes An echo effect with the repeats alternating between stereo sides.
Pan Room This is slightly similar to “Stereo Echoes”, but the echo repeats have soft attacks.

q If you need to conserve computer power, try using the Low Density algorithm. This uses much less power than
the other algorithms.

791 HALF-RACK EFFECTS

 The selected reverb algorithm can be tweaked using the parameters on the device panel:

| Parameter | Description
Size Adjusts the emulated room size. Middle position (value 0) is the default size for the selected algorithm.
Lowering this parameter results in a closer and gradually more “canned” sound. Raising this parameter
results in a more spacey sound, with longer pre-delay.
For the “Stereo Echoes” and “Pan Room” algorithms, the Size parameter adjusts the delay time.
Decay This governs the length of the reverb effect. Middle position is the default decay time for the selected al-
gorithm.
Note: Decay is not used for the “Gated” algorithm.
Damp Raising the Damp value cuts off the high frequencies of the reverb, thereby creating a smoother, warmer
effect.
Dry/Wet If you are using the reverb as an insert effect, you use this parameter to adjust the balance between the
unprocessed audio signal (dry) and the effect (wet).
If the reverb is used as a send effect, this should be set all the way to wet only, since you can control the
balance by using the AUX send controls in the Mixer.

CV Inputs
You can control the Decay parameter via the CV input on the back of the Reverb device.

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 D-11 Foldback Distortion

! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.
The D-11 is a simple but effective distortion effect, capable of producing anything from just a whisper soft touch of
distortion, to complete thrashing. This effect is most often used as an insert effect.

Parameters
The distortion has the following parameters:

| Parameter | Description
Amount This controls the amount of distortion. The higher the value, the more distortion.
Foldback This adjusts the character of the distortion by introducing foldback, which makes the waveform
more complex.
The default value is in the middle position. This produces a “flat” clipping distortion, which is the
most common type. Lowering the parameter makes the sound rounder and more gentle, raising it
makes the sound sharper and more evil.

CV Inputs
On the D-11 you will find a CV input for controlling the Amount parameter. This can produce very drastic effects, es-
pecially if you control parameters in the instrument device (such as filter frequency and resonance) at the same time.

793 HALF-RACK EFFECTS

 ECF-42 Envelope Controlled Filter

! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.
The ECF-42 is a multimode filter with a built in envelope generator. It is mainly designed to be used together with
pattern devices to create pattern controlled filter and envelope effects, but it can also be triggered via MIDI, or used
as a “static” filter for shaping the sound of an instrument device or a whole mix.

Usage
The Envelope Controlled Filter is best connected as an insert effect. However, unlike the other effects it is not a pure
“stand-alone” device. To make the most of the ECF-42, you need either CV/Gate from an external device or MIDI
notes from a sequencer track.
• If you connect a device to the ECF-42 using audio inputs/outputs only, it will simply act as a filter with no ve-
locity or envelope modulation.
Hence, all filter parameters are “static”, unless you manually turn the knobs or automate them in the sequencer.
• Connecting a gate signal to the Env Gate input on the back panel of the device allows you to trigger the enve-
lope generator for the filter.
Note that the ECF-42 envelope generator is not triggered by the audio itself - the envelope parameters won’t do
anything unless the device receives gate signals.
• By putting the ECF-42 in a Combinator (so it can receive MIDI), you can have the envelope triggered by MIDI
notes on the track.
The envelope is affected by the position, length and velocity of the MIDI notes (but not by their pitch).
q If you are unfamiliar with basic filter and envelope parameters, please refer to “Envelopes - General” in the
Subtractor chapter for a description of these.

The Filter Parameters

The ECF-42 filter section has the following parameters:

| Parameter | Description
Mode This button sets the desired filter mode. Three modes are available: 24dB/octave lowpass, 12dB/octave
lowpass and 12dB/octave bandpass.
Freq This is the filter cutoff frequency. When using the ECF-42 in “static” mode (without triggering the enve-
lope), this parameter adjusts the frequency content of the sound.
When using the envelope, the Freq parameter serves as the start and end frequency for the filter sweep.
Res This is the filter resonance. Raising this produces a more extreme, “synthy” effect.
Env Amt Determines how much the filter frequency should be affected when the envelope is triggered. The higher
the value, the more drastic the effect. Note though, that if the Freq parameter is set high, raising the Enve-
lope Amount will not make any difference over a certain value! This is because the filter is already fully
opened - try lowering the Freq parameter in that case.
Velocity This parameter determines how much the gate velocity value should affect the envelope amount.

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 The Envelope Parameters

This is a standard envelope generator with Attack, Decay, Sustain and Release parameters. It is triggered by a gate
signal connected to the Env Gate input on the back panel, or by MIDI notes on a sequencer track connected to the
ECF-42. The parameters have the following functionality:

| Parameter | Description
A (Attack) When the envelope is triggered, this is the time it takes before the envelope signal reaches its max
value.
D (Decay) After reaching its max value, this is the time it takes for the envelope signal to reach the sustain level.
S (Sustain) If the gate remains open (or the MIDI note is held), the envelope signal will remain on this level.
R (Release) When the gate is closed (gate CV goes back to 0) or the MIDI note ends, this is the time it takes for the
envelope signal to drop from its current value to the start value (set by the Freq parameter).

• The Gate indicator lights up when the device receives a signal to the Env. Gate input on the back panel or a
MIDI note from a sequencer track.

CV/Gate Inputs
On the back panel of the ECF-42, you can find the following CV/Gate inputs:
• Freq CV.
Use this for controlling the filter frequency from another device. For smooth filter modulation, try connecting an
LFO to this input.
• Decay CV.
For controlling the envelope decay parameter from another device.
• Res CV.
Allows you to control the filter resonance from another device. Can be very effective in combination with filter fre-
quency sweeps.
• Env. Gate.
This is where you connect a gate signal (e.g. from a Matrix or Redrum device) for triggering the envelope.

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 PH-90 Phaser

The PH-90 Phaser is a classic phaser effect with some special features for fine-tuning the sound. It can create the
classic sweeping phaser sounds suitable for pads or guitars, but also more extreme effects if you like. The phaser is
best used as an insert effect.

Theory
A phaser works by shifting portions of the audio signal out of phase, and then adding the processed signal back to
the original one. This way, narrow bands of the frequency range (“notches”) are filtered out. When these frequencies
are adjusted, a sweeping phaser sound is created.
The PH-90 is a four-stage phaser, which means that there are four “notches” in the frequency response curve (this
is a little like using four notch filters with different filter frequencies - see “Notch” in the Subtractor chapter for an ex-
planation of notch filters).
When the phaser frequency is adjusted (manually or by the built-in LFO), these notches will move in parallel in the
frequency spectrum. Furthermore, you can adjust the distance between the notches (Split) and their Width. Adding
feedback raises the filter gain just below each notch in the frequency range, creating a more pronounced effect.

Parameters
| Parameter | Description
Frequency Sets the frequency of the first notch. Adjusting this will move the other notches correspondingly. This is the parameter
modulated by the LFO to create phaser sweeps.
Split This adjusts the distance between the notches in the frequency range, thereby changing the character of the effect.
Width Determines the width of the notches. Raising the Width deepens the effect and simultaneously makes the sound
more hollow and thin. This will also have an effect on character of the feedback “tone”.
LFO Rate This is the speed of the LFO modulating the frequency parameter. The higher the value, the faster the phaser sweeps.
LFO Sync This button lets you activate/deactivate LFO sync. When it is activated, the frequency of the LFO is synchronized to
the song tempo, in one of 16 possible time divisions. The LFO Rate knob is then used for setting the desired time di-
vision. Turn the knob and observe the tooltip that appears for an indication of the time division.
LFO Freq. Mod This determines the depth of the LFO modulation, i.e. by how much the frequency parameter should be modulated.
If you turn this to zero, the effect will be a static, formant-like sound (most effective if you add a little feedback).
Feedback This is similar to the resonance control on a filter. Raising the feedback gives a more pronounced “tone” in the effect.
For “singing” phaser sounds, try raising this to the maximum.

CV Inputs
The following CV inputs are available on the back panel of the device:
• Freq CV.
Adjusts the frequency parameter. Use this e.g. for creating envelope controlled phasing (preferably with LFO Freq.
Mod turned off in the device).
• Rate CV.
Lets you control the speed of the modulating LFO from another device.

796 HALF-RACK EFFECTS

 About Stereo and Mono connections
The PH-90 can be connected using mono or stereo inputs, as shown by the graphs on the back panel. Note the fol-
lowing:
• A “moving” stereo effect is only obtained when you use a mono input and stereo outputs.
With a stereo input, the two sides are processed in parallel, maintaining the stereo image of the original sound.
• When you are using a mono input and stereo outputs, there will only be a stereo effect if the internal LFO is
used.
If you turn LFO Mod Amount to zero, both stereo outputs will carry the same signal (mono). This is because the
“fake stereo” effect is produced by inverting the modulation for one of the output channels.

797 HALF-RACK EFFECTS

 UN-16 Unison

! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.
The UN-16 simulates the sound of several detuned voices playing the same notes simultaneously. The voices are in-
dividually slightly delayed and also pitch modulated by low frequency noise. This produces a rich chorus effect with
the voices spread across the stereo field (given that stereo outputs are used).
The UN-16 can be used as an insert effect or a send effect.

Parameters
| Parameter | Description
Voice Count This switch sets the number of voices for the effect; 4. 8 or 16.
Detune This sets the amount of detuning for the voices. Turn clockwise for stronger detuning effects.
Dry/Wet If you are using the UN-16 as an insert effect, you use this parameter to adjust the balance between
the unprocessed audio signal (dry) and the effect (wet).
If the UN-16 is used as a send effect, this should be set all the way to wet only, since you can control
the balance by using the AUX send controls in the Mixer.

CV Input
One CV input is available on the back panel of the device. This controls the Detune parameter.

798 HALF-RACK EFFECTS

 COMP-01 Auto Make-up Gain Compressor

! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.
The COMP-01 compressor levels out the audio, by making loud sounds softer. To compensate for the volume loss,
the device has an automatic make-up gain, that raises the overall level by a suitable amount. The result is that the au-
dio levels become more even and individual sounds can get more “power” and longer sustain.
The COMP-01 should be used as an insert effect, either for a single instrument device or for a whole mix (e.g. in-
serted between a Mixer device and the Hardware Interface). There are no CV inputs for this device.

Parameters
| Parameter | Description
Ratio This lets you specify the amount of gain reduction applied to the signals above the set threshold. The
value is expressed as a ratio, from 1:1 (no reduction) to 16:1 (levels above the threshold are reduced by
a factor 16).
Threshold This is the threshold level above which the compression sets in. Signals with levels above the threshold
will be affected, signals below it will not.
In practice, this means that the lower the Threshold setting, the more the compressor effect.
Attack This governs how quickly the compressor will apply its effect when signals rise above the set threshold.
If you raise this value, the response will be slower, allowing more of the signal to pass through the com-
pressor unaffected. Typically, this is used for preserving the attacks of the sounds.
Release When the signal level drops below the set threshold, this determines how long it takes before the com-
pressor lets the sound through unaffected. Set this to short values for intense, “pumping” compressor
effects, or to longer values for a smoother change of the dynamics.
Gain meter This shows the amount of gain reduction or increase (in dB), caused by the combined compression and
make-up gain.

799 HALF-RACK EFFECTS

 PEQ-2 Two Band Parametric EQ

! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.
While there is a simple two-band shelving equalizer available for each channel in the mixer, the PEQ-2 gives you
much more precise control over the tone color. The device consists of two independent, fully parametric equalizers
and is most often used as an insert effect, in mono or stereo.

About the two EQ modules

The two independent EQs are labeled “A” and “B”.
• EQ A is always active (provided that the effect device is in “On” mode and that you have set the Gain to a value
other than 0).
• To activate EQ B, click the button next to the EQ B parameters, so that the LED lights up.
q If you only use one EQ, it’s a good idea to turn EQ B off, to conserve computer power.

Parameters
For both EQs (A and B), the following parameters are available:

| Parameter | Description
Frequency This determines the center frequency of the EQ, e.g. at which frequency the level should be de-
creased or increased. The range is 31 Hz to 16 kHz.
Q This governs the width of the affected area around the set center frequency. The higher the value,
the narrower the affected frequency range.
Gain Specifies how much the level of the selected frequency range should be boosted (positive values) or
lowered (negative values). The gain range is ±18 dB.

About the graphic display

The graphic display to the left in the device panel shows the frequency response curve as set by the EQ parameters.
This gives a visual feedback and helps you tailor the EQ settings.

CV Inputs
The following CV inputs are available on the back panel of the device:
• Freq 1 CV.
Allows you to control the frequency of EQ A from another device, creating subtle or dramatic EQ sweeps depend-
ing on the Q and Gain settings.
• Freq 2 CV.
Allows you to control the frequency of EQ B in the same way.

800 HALF-RACK EFFECTS

Chapter 50
The Combinator
 Introduction

A custom made Combinator front panel.

The Combinator is a special device that allows you to save and recall any combination of Reason/Rack Extension/
VST (Reason stand-alone only) devices (instruments, effects, mixers etc.) and their internal connections. A saved
Combinator setup can be loaded as a patch, called a “Combi”. The Combinator device itself acts as a container for the
devices in a Combi.
The basic idea behind the Combinator device is simple, but very powerful. Being able to save multiple devices as a
Combi enables you to instantly recall any type of setup, however complex, as simply as loading a patch!
The original version of the Combinator was first released with Reason 3. In Reason 12 we have made a major over-
haul, to make it even more powerful and flexible!
Here are some of the Combinator features:
D Design your own custom front panel with back-drops, control configurations and layouts.
Now you can choose what controls - and how many of each - you want to have on the Combinator front panel. You
can also select the height of the Combinator panel (1U to 6U) and add your custom back-drop image to use as
background on your Combinator.
D Create split or layered multi-instruments.
Add any number of instrument devices and play them as a single layered instrument. Instrument devices in a
Combi can also be assigned to specific keyboard/velocity zones.
D Save instrument/effect combinations.
Save an instrument together with your favorite effect(s).
D Create multi-effect devices.
You can create and save complex effect chains as Combis.

About the Combi patch format

The Combinator saves files in the Combi (.cmb) patch format. When you load a ready-made Combi patch, all devices
included in the Combi, their corresponding parameter settings and audio and CV connections are instantly recalled.
The Factory Soundbank includes many preset Combinator patches, divided into various categories.
There are two basic types of Combi patches; instrument and effect combis.
• Instrument Combis contain instrument devices and generate sound.
• Effect Combis contain effect devices and are used to process sound.

802 THE COMBINATOR

 Combinator overview
In the picture below an unfolded Combinator device is shown with its Editor and Devices container. The Combinator
in this example contains a pre-made patch, that’s why there are devices in it, and assigned controls.
1
2

The front of the Combinator consists of the following elements (from the top down):
• 1. The narrow panel at the top is always shown, even when the whole Combinator is folded.
It contains a display which (amongst other things) shows the name of the currently loaded Combi, and standard
Select/Browse/Save patch buttons.
• 2. Next is the Control panel, which is always shown if the Combinator is unfolded.
You can configure and design your own unique control panel, see “Configuring the Combinator panel”.
• 3. The Editor contains views for mapping devices and controls, and for configuring the Control panel.
Here are also parameters for Key and Velocity Zone mapping as, well as Modulation Routing settings.
The Editor can be shown/hidden by clicking the “Editor” button on the Control panel.
• 4. At the bottom are the devices included in the Combi.
At the top of the device section is an 8-channel stereo summing mixer, which allows for quick audio routing of de-
vices in the Combi. The summing mixer is always present and cannot be removed.
Devices can be shown or hidden by clicking the “Devices” button on the Control panel.

803 THE COMBINATOR

 Creating a Combinator device
Creating an empty Combinator device
D Add a Combinator device from the Utilities palette in the Device Palette.
Either double click the device or drag and drop from the device palette to the rack.
D Select “Combinator” from the “Utilities” sub-menu on the Create menu.
This will create an empty Combinator. Empty Combinator devices can be used as a starting point when creating
new Combi patches. It also allows you to browse for existing Combi patches.

Creating a Combinator by combining devices

You can create a Combinator device by combining existing devices:
1. Select two or more devices in the rack by [Ctrl]-clicking (Win) or [Shift]-clicking (Mac).
The devices do not have to be adjacent in the rack.
! Note that you should never select any Mix Channel or Audio Track devices (Reason stand-alone only), since
this will disable the “Combine” function described below.
2. Select “Combine” from the Edit menu or context menu.
This creates a Combinator device containing the devices that were selected according to the following rules:
• A sequencer track is created for the Combinator, just as when creating instrument or pattern devices.
• The new Combinator device appears at the position below the selected device that is farthest down in the rack.
• The selected devices are moved to be contained within the Combinator’s “container”.
Their internal order is not changed. Reason attempts to autoroute the first “input device” and first “output device”
to the Combinator Mixer To/From Devices connectors - see “About internal and external audio connections”.
Other connections are unchanged.
• Devices outside the Combi remain in the same order as before.
! Please see “Adding devices to a Combi” for details about auto-routing.

Creating a Combinator by browsing patches

Double clicking, or dragging & dropping a Combinator (.cmb) patch from the Browser to the rack will automatically
create the Combinator device in the rack and load the selected patch.

804 THE COMBINATOR

 About internal and external audio connections
Unlike other devices, the Combinator contains both external and internal audio connections.
• External connections are used to connect a Combinator to devices outside the Combi.
• Internal connections are how devices within the Combi are connected.

External audio connections

• The Combinator Input connectors are the inputs to the Combinator (used for effect Combis only).
Internally, these inputs are connected to the “To Devices” connectors of the Combinator Mixer (see below).
• The Combinator Output connectors carry the audio output of the Combinator Mixer.
This output connects with a device outside the Combi, normally a mixer. Internally, this output is connected to the
“From Devices” connectors. When you create a new Combinator this output will be auto-routed to the first avail-
able input channel in the I/O device.

Internal audio connections

• The “To Devices L/R” outputs on the Combinator Mixer connect to an input on a device in the Combi when the
Combinator is used as an effect.
• The “From Devices” inputs on the Combinator Mixer is where the outputs from devices in the Combi are con-
nected.
The “From Devices” outputs consist of 8 stereo output pairs.

805 THE COMBINATOR

 About External Routing
If an individual device in the Combi is directly connected to a device outside the Combinator - or if an external device
is connected to any of the “From Devices” connectors on the Combinator Mixer - this is known as an “External Rout-
ing” connection, which should generally be avoided. The reason for this is that such connections are not saved with
the Combi patch.
! Therefore, all connections to/from a Combi should pass via the To/From Device connectors on the Combinator
device if you wish to make the Combi “self-contained”.
If a Combi contains any external routing connections, this is indicated by the “External Routing” LED on the front and
back of the (unfolded) Combinator Mixer:

It is still possible to use a Combi with External Routing connections within the context of a song (where all routings
are saved with the song). Just keep in mind that the external routing connections will not be part of the Combi patch!

How to avoid external routing connections

As explained previously, all connections to/from a Combinator device should pass via the To/From Device connec-
tors in order to make the Combi self-contained. If you should have used up all “From Devices” connectors you need
to include a mixer device and route the outputs of that mixer to a free “From Devices” connector pair.
q If you should run out of “From Devices” connectors, or want more mixing capabilities, like panning and Effect
Sends, the Line Mixer 6:2 is ideally suited for mixing device outputs in Combis (see “The Line Mixer 6:2”).

806 THE COMBINATOR

 Adding devices to a Combi
Creating new devices in a Combi
To add devices to a Combi, you can drag and drop them from the Device Palette as usual.
If you want to add devices using the Create menu or by double clicking in the device palette, you need to make the
Combinator the target device in one of the following ways:
D Click in the empty area at the bottom of the open device container so that the red insertion line is shown:

If you create a device with the insertion line showing, the new device will appear at the bottom of the Combi con-
tainer.
D If the Combi already has devices in it, select a device in the Combi (but not the Combinator device itself).
When you add a device from the Create menu it will appear below the selected device (just like in the rack).
! No sequencer tracks will be automatically created for devices added to a Combi.

About auto-routing
The auto-routing of devices in a Combi is similar to devices in the rack:
• If a device in a Combi is selected, the new created device will appear below the selected device according to
standard rules.
• If an effect is selected and you create a new effect device, these will be connected serially.
• If an instrument device is selected and you create an effect it will be connected as an insert effect to the instru-
ment device.
• If a mixer is selected and you create an effect it will be connected as a mixer send effect.
• If an instrument device is selected and you create another instrument device it will be added below the se-
lected device and connected to the first available Combinator Mixer input channel.
• If you hold down [Shift] and create a new device, no auto-routing will take place.
• If you hold down [Option] (Mac) or [Alt] (Windows) and create a new device, a sequencer track will be created
for the device.
D If you add a device to an empty Combi, its output(s) will be auto-routed to the first available “From Devices”
connector(s) of the Combinator Mixer. For effect devices, the input will also auto-route to the “To Devices”
connectors.

807 THE COMBINATOR

 Adding existing devices to the Combi from the rack
You can move devices in the rack that are currently outside the Combi into the Combinator holder. This works as fol-
lows:
1. If you want to add more than one device at the same time, [Ctrl]-select (Win) or [Shift]-select (Mac) the devices.
2. Click on the panel area of a device.
You can click anywhere outside the actual parameters and displays.
3. Drag the device(s) into the Combinator.
A thick orange vertical line indicates where the device(s) will be positioned. Note that the orange line can be to the
left of a device (indicating that the moved device will be inserted before the other device) or to the right of a device
(indicating that the moved device will be inserted after the other device). You can also add devices to the empty
space at the bottom of the Combi holder.
• No auto-routing will take place.
The existing connections to the rack will be kept, so there may be external routing connections (see “About Exter-
nal Routing”). If there are, “External Routing” will be shown the Combinator name display (and on the back of the
Combinator a LED indicator will be lit).
• If you drag devices into the Combi with [Shift] pressed, the devices will be disconnected from the rack and
auto-routed according to the insert position.
The routing between dragged devices will be kept. If it isn’t possible to auto-route a dragged device (e.g. if there
are no free From Devices inputs on the Combinator Mixer), the device will become unconnected.
• If you drag devices into the Combi with [Alt](Win) or [Option](Mac) pressed, the devices will be copied. No
auto-routing takes place.
Using [Shift] at the same time will attempt to auto-route according to the same rules as described above.

Combining two Combis

• If you select two Combinators in the rack and then select “Combine” from the context menu the lower Combi
will be uncombined and the devices added to the upper Combi in the rack when combining.
Existing routing will be unchanged.

808 THE COMBINATOR

 Combinator handling
Moving the entire Combi
This works much the same as for other devices in the rack.
D Select the Combinator by clicking on the Control panel background and dragging to a new position.
An outline of the Combinator is shown when you drag, and an orange line shows the insert position. All connec-
tions are kept.

Moving devices within a Combi

Just as for devices in the rack, an outline of the devices is shown when you drag, and an orange line shows the insert
position. All connections are kept.

Moving devices out of a Combi

If you move devices out of a Combi the following applies:
D The routing in the Combinator is unchanged, and the External Routing indicator is likely to light up.
If you press [Shift] when dragging, the device(s) will attempt to auto-route to the rack according to standard rules.

Deleting devices in a Combi

This works exactly as for devices in the rack.
D Select the device and then either select “Delete Devices and Tracks” from the Edit menu or context menu, or
press [Backspace].

Uncombining devices
You can uncombine a whole Combi or selected devices within a Combi in the following way:
D If you select a Combinator and then select “Uncombine” from the Edit/context menu, the Combinator device
will be removed, and all devices contained in the Combi will be moved to the rack.
The devices previously connected to the To/From Devices connectors will now be connected to the rack in the
same way the Combinator device was (via the Combinator output and input).
D If you uncombine a few selected devices in a Combi these will be removed from the Combi and added to the
rack below the Combi.
Connections are unchanged, so external routing is likely to happen.

809 THE COMBINATOR

 Configuring the Combinator panel

This is the default Combinator panel - the one that appears after you have selected “Reset Device” from the context
menu. Like standard instrument devices it features Pitch and Mod wheels and various controls.

About the virtual controls

• The default four Rotary knobs and buttons in the middle of the Control panel are “virtual” controls that can be
assigned to parameters and functions in devices contained in the Combi.
These controls are by default not assigned to any parameters in new Combis.
• You can also add or remove controls, to design your own custom panel, see “Adding and positioning controls
on the front panel”.
• You assign parameters to the controls in the Modulation Routing section of the Editor panel (see “Using Mod-
ulation Routing”).
• Movements of the virtual controls can be recorded as automation.
• Each control can be assigned to any number of parameters.
• Up to four virtual controls can be modulated from the Control CV inputs on the rear panel, see “Control CV In”.
• Double clicking on the label for a control lets you type in an appropriate name for it.

The Pitch Bend and Modulation wheels

The Pitch and Mod wheels on the Control panel will mirror the corresponding actions on your master keyboard, just
like for standard instrument devices.
• When a Combinator device has Master Keyboard input and the Combi contains several instrument devices
these will all receive pitch bend and modulation data.
The settings in the instrument devices determines what happens when applying pitch bend or modulation.
• For example, moving the Mod wheel could apply vibrato for one device and change the filter cutoff frequency
for another device.
• Pitch Bend will also be applied according to individual settings in the Range field for all instrument devices in
a Combi.

Opening the Configuration panel

1. Open the Editor by clicking the Editor button at the top of the Combinator front panel:

2. Click the Configure button to open the panel Configuration panel:

810 THE COMBINATOR

 Setting the front panel size
D Select the desired panel height from the “Panel size” drop-down list.
You can choose from 1U (rack unit) to 6U.

Selecting front panel background color

D Select the desired panel background color from the Background Color drop-down list.

Selecting a backdrop image

This function allows you to change the “skin” of the Combinator front panel. You can design your own labels for the
assignable controls, and change the color and look of the whole panel.

811 THE COMBINATOR

 D Click the Backdrop drop-down on the Config panel in the Editor:

The Backdrop drop-down features the items “None”, “Default” and “Load...”
• “None” leaves the panel with the selected background Color (see “Selecting front panel background color”).
• “Default” is a semi-transparent texture which gives a higher quality look to the front panel.
Since the texture is semi-transparent the selected background Color will shine through.
• Selecting “Load...” will open the image browser.
Here you can select image file in JPEG (.jpg) or PNG format.
Instead of selecting “Load...” from the drop-down you could select the Combinator device and choose “Select
Backdrop...” from the Edit/context menu.
q Transparent PNG files are also supported, if you want to layer a backdrop image on top of a background color.
• For full hi-res support the width of the image should be 3770 pixels.
The height depends on what front panel size you have selected, see “Setting the front panel size”.
These are the optimal height dimensions for each of the front panel sizes:
1U: 345 pixels
2U: 690 pixels
3U: 1035 pixels
4U: 1380 pixels
5U: 1725 pixels
6U: 2070 pixels
If you select an image with other pixel dimensions, the image will be automatically scaled.
• The knobs, buttons, patch name display and patch buttons cannot be redesigned using a backdrop.
D If you wish to design your own text labels for the virtual controls, you should first remove the original text la-
bels:
Double click on a control label, remove the current text and press [Enter]. Alternatively, uncheck “Show Label” in
the Configuration panel in the Editor, see “Changing the appearance of the controls”).
D To remove a Backdrop, select “Remove Backdrop” from the Combinator context menu - or select “None” in the
Backdrop drop-down on the Config panel in the Editor.
The currently selected Background Color of the Combinator panel is restored.

About “Using Legacy Backdrop”

This is automatically turned on when you load a legacy Combi patch with a backdrop, and will show buttons in the
same way as on the original Combi (including Show Editor/Show Devices buttons on the panel) and hide the By-
pass/On/Off label as this is expected to be in the backdrop.

812 THE COMBINATOR

 About the “Bypass Label”

The “Bypass Label” item is available only when the “Using Legacy Backdrop” is de-selected.
D Select the desired color (or presence) of the Bypass/On/Off text from the drop-down.
The setting affects the color of the text on the top left of the Combinator panel:

About the “Combi Layout Templates” folder

In Factory Sound Bank/Combinator Patches/ there is a folder named “Combi Layout Templates”. This folder con-
tains a selection of pre-made Combinator panel configurations and layouts that can serve as starting points/inspira-
tion for your own Combinator designs.

Selecting, positioning and editing front panel controls

1. Open the Editor by clicking the Editor button at the top of the Combinator front panel.
2. Click the Configure button to open the panel Configuration panel.

Adding and positioning controls on the front panel

To the left on the Editor Configuration panel are a selection of pre-made knobs, faders, buttons and wheels that can
be freely laid out on the Combinator front panel:

Controls selector on the Configuration panel.

813 THE COMBINATOR

 1. Click a desired control icon on the Configuration panel.
The control shows up with a frame around it on the Combinator front panel.

Adding a new control to the front panel.

2. Click and drag the control to the desired position on the Combinator front panel.
Visual guides appear as you move the control so you can easily place the control aligned with other controls al-
ready on the panel.
D Use the [Arrow] keys on the computer keyboards to adjust the position, if desired.
D Double click the label and type in a new name, if desired:

3. Click a new control to add another one.

D You can also select multiple controls, by dragging or by [Shift]-clicking on the panel, and move them together
on the front panel.
Alternatively, [Command]/[Ctrl]-click the controls in the Panel Elements list to select multiple controls.

! The maximum number of controls that can be placed on a Combinator panel is 32 knobs/faders plus 32 but-
tons, 2 wheels (pitch and modwheel), 1 Run button and 1 Bypass FX button.

814 THE COMBINATOR

 It’s also possible to align and distribute a selection of controls, to get perfect position with even spacing on the panel.
D Select the group of controls you want to align/distribute, click the black arrow button on the control you want
to use as reference, and select action from the pop-up:

About the “Run” and “Bypass FX” buttons

The “Run” button can be used to start/stop all (Reason) pattern devices included in the Combi. This works exactly as
pressing the Run button on the pattern device panel. The on/off status of this button is not saved with the Combi
patch. Pressing play on the transport panel in the sequencer will automatically activate “Run Pattern Devices”.
q In the Reason Factory Sound Bank, Combi patches containing pattern devices have “(run)” at the end of their
patch names.
The “Bypass FX” button allows you bypass all effect devices in a Combi. It works as follows:
• All insert effect devices in the Combi are switched to Bypass mode.
• All effects connected as send effects to a mixer device are switched off.
• Clicking this button will not affect effect devices that were bypassed or turned off already.

815 THE COMBINATOR

 Changing the appearance of the controls
It’s possible to change the appearance of the controls.
1. Click the control you want to edit, either on the Combinator front panel or in the Panel Element list on the Con-
figuration panel.
The characteristics of the selected control is now displayed in the Settings section:

• The control type is shown in the Type display.

D The Name display shows the control name.
(To edit the name, double click the name label on the Combinator panel.)
D Select control style from the Style drop-down list.
D Select the Scale appearance (if the control features a scale)
D Click the check mark to show/hide the control name label on the Combinator front panel.
D Select the label text color from the Text Color drop-down list.
D Choose where you want the text label positioned around the control from the Position drop-down list.

Deleting controls on the front panel

1. Select a control (or multiple controls, by dragging or by [Shift]-clicking the controls on the front panel).
Alternatively, click in the Panel Element list ([Command]/[Ctrl]-click to select multiple controls).
2. Press [Delete]/[Backspace] on the computer keyboard to delete the selected control(s)
D Alternatively, click the X next to the control name in the Panel Elements list to delete one control at a time:

816 THE COMBINATOR

 Assigning panel controls to parameters in the Editor
The Editor is used for key mapping and setting velocity ranges for instrument devices, and for Modulation Routing;
assigning device parameters to the knobs/buttons/sliders on the Control panel.
D To bring up the Editor panel, click the “Editor” button on the unfolded Control panel.
The Editor appears below the Control panel.

A fully assigned Combinator.

• To the left on the Editor panel the devices in the current Combi are listed in the order they appear in the Com-
binator container - from top to bottom.
Clicking on a device in the list selects it for editing.
• To the right is the Modulation Routing section where you can assign parameters to the controls on the Control
panel.
See “Using Modulation Routing”.

Key Mapping instrument devices

About the Key (receive MIDI Notes) column

To the right of the Device name is a Key column.

D Click to select if the device should respond to MIDI Note input.
• If you deactivate the “Key” checkbox the selected device will not respond to incoming MIDI note messages.
For devices that cannot use MIDI notes (such as most effect devices), there is no checkbox in this column.

817 THE COMBINATOR

 Setting the Key Range
Each instrument device can have its own separate key range, the lowest and the highest key that will trigger the de-
vice. This allows you to create splits and layers for instrument devices in a Combi.
1. Make sure the “Key” checkbox is ticked for the instrument devices you want to edit the Key Range.
The Key Range is shown as a horizontal bar, and as note numbers on either sides of the bar. By default, the entire
note range is selected (C -2 to G 8).

There are several ways you can change the Key Range:
D By clicking in the Key Range value fields and moving the mouse up or down.
D By moving the handles of the horizontal bar.
D By dragging the horizontal bar itself you can also move entire key zones horizontally, thereby changing their
key ranges.
2. Using either method, set the desired key range for the desired device(s).
When done, the device(s) will only play back notes in the set key range.
D By setting up key ranges for devices in a Combi, you can create split instruments.
For example playing notes below C 2 could trigger a device playing a bass sound, whereas playing notes above C
2 could trigger a device playing a pad sound.
D Instrument devices in a Combi that share the same key range will be layered - i.e. play at the same time.
This given that no velocity ranges have been set up - see below.
D You can of course set up overlapping ranges where notes within a set key range will layer two (or more) de-
vices, but notes above and below the set range will play separate devices.

About the Transpose function

The Transpose field to the far right on the panel allows you to transpose the instrument devices. It will not shift the
key mapping, just the pitch of the device.
D Click and drag up/down in the Transpose field to edit the transposition.
The range is +/- 3 octaves, in semitone steps (+/-36 semitones).

818 THE COMBINATOR

 Enabling/disabling Performance Controllers

To the bottom right of the panel is a drop-down list that features all standard MIDI Performance Controllers (Pitch
Bend/Mod Wheel/Breath/Expression Pedal/Sustain Pedal/Aftertouch) for the selected instrument.
• The check boxes in the drop-down list allow you to control whether MIDI Performance Controller data is to be
received for each instrument device in a Combi.
• If you deactivate any of the Performance Controllers, the corresponding controller(s) will not be received by
the selected instrument device.
All Performance Controllers are on by default.

Setting Velocity Ranges for instrument devices

When instrument devices are set up so that their key ranges overlap – completely or partially – you can use velocity
switching to determine which devices should be played back depending on how hard or soft you play on your MIDI
keyboard.
This is done by setting up velocity ranges.
Each time you press a key on your MIDI keyboard, a velocity value between 1-127 is sent to Reason. If you press the
key softly, a low velocity value is sent and if you press it hard, a high velocity value is sent.
This velocity value determines which devices will be played and which will not.
1. Select an instrument device in the Device list (non-instrument devices, e.g. effects and mixers do not have ve-
locity ranges).
By default, the entire range is selected (0 - 127).
There are several ways you can change the current Velocity Range:
D By clicking in the “Lo” and “Hi” value fields and moving the mouse up or down.
D By moving the handles of the horizontal bar.
D By dragging the horizontal bar itself you can also move entire Velocity Range.
2. When you have set a range, the device will only be triggered by notes played within this velocity range.

819 THE COMBINATOR

 About overlapping Velocity Ranges
You can set overlapping velocity ranges. Here’s an example of how this can work:
• Device 1 has a velocity range from 1-45.
• Device 2 has a velocity range of 33-88.
• Device 3 has a velocity range of 76-127.

127

100

80

60

40
Device 3
20
Device 2
Velocity 0 Device 1

Now, velocity values between 33 and 45 will trigger notes from both Device 1 and Device 2. Likewise, velocity values
between 76 and 88 will trigger sounds from Device 2 and Device 3.

Using Modulation Routing

The Modulation Routing section to the right on the Editor panel allows you to assign any parameter or function in de-
vices included in a Combi to any of the virtual controls on the Control panel.
You can also control Combi parameters by connecting external CV modulation sources to any of the Source CV in-
puts on the back of the unfolded Editor panel, see “Wheel CV In and Source CV In”.

820 THE COMBINATOR

 About Rotary/Slider and Switch controls
The virtual Rotary/Slider and Switch controls operate much like the equivalent controls on the real devices:
• A Rotary/Slider control can either smoothly change parameter values (e.g. a level control), or step through
fixed values (like the Oscillator waveform spin controls on a Subtractor).
• A Switch control will switch between two set values like an on/off switch.
Worth noting is that there are buttons on several Reason devices that will step through a series of values, for ex-
ample LFO Waveform buttons. If LFO Waveform is assigned to one of the virtual Buttons you will only be able to
switch between two of the six LFO waveforms (which waveforms is determined by the Min/Max range).
• The available range for each selected parameter is shown in the Modulation Routing Min/Max fields.
Most sliders and rotary knobs on the actual devices have the standard 0-127 or -64 to 63 range. Selectors and
spin controls can have any value range.

About the CV Inputs

On the back of the Editor panel are CV Modulation Inputs for connecting external CV signals to modulate any of the
parameters that are accessible in the Target section in the Editor, see “Wheel CV In and Source CV In”.

Assigning panel controls to device parameters

Assigning Combinator panel controls to device parameters can be done in two different ways:
1. Right click the desired device parameter and select the desired Combinator panel control to assign it to from
the “Map to Combi Control” pop-up:

Assigning the “Master Volume” control on an Europa device to the “Level” control on the Combinator panel.
If you chose the method described above you can continue directly to step 4 below in this description.

821 THE COMBINATOR

 D Alternatively, select the device you wish to assign parameters on in the device list to the left:

The Modulation Routing section to the right contains four columns:

• In the Source column, any currently assigned Combinator panel controls are listed.
Each field can be changed to any Control/Slider/Switch/CV Input or Performance Controller by clicking the arrow
and selecting from the drop-down list.
• The drop-down lists in the Target column contain all parameters for the currently selected device.
Lastly in each Target drop-down list is the option to receive note data or not. This is useful if you want to tempo-
rarily enable/disable MIDI Note input for specific devices, e.g. to temporarily enable layered instruments - or switch
instrument playback.
• The Min/Max columns allow you to specify a value range for the virtual control.
It’s also possible to set the Min value higher than the Max value, meaning that the value will decrease when you
turn up the knob/slider.

822 THE COMBINATOR

 2. Click the small black triangle in the Source column and select the Combinator panel control from the drop-
down list:

Setting the “Level” Combinator panel control as Source.

3. Click the small black triangle in the Target column and select the desired Target parameter from the drop-
down list:
In the drop-down, all the available parameters for the device are listed.

Assigning the Europa “Master Volume” parameter as Target.

4. You can specify a range for the assigned parameter by clicking in the Min and Max columns and moving the
mouse up or down, or by dragging the handles of the horizontal Range bar, or by dragging the Range bar (to
offset the range).
By the maximum available range is set by default:

823 THE COMBINATOR

 • It’s also possible to “reverse” the range by dragging the Range handles past each other, or dragging the Min/
Max boxes so that the minimum value becomes max and the maximum value becomes minimum.
The result is that the Target parameter value will decrease when you turn up the Source knob/slider, and vice
versa. Here is an example of how to reverse the range for a parameter:

The Filter Reso parameter edited to have a reversed Range, as indicated by the light gray Range bar.
• It’s also possible to set a Source Range for each Source control, so that the knob/slider only affects the Target
parameter in a certain region/interval of the Source control:

The Filter Freq stays at Linear change from 200 Hz

200 Hz in this span. to 3 kHz in this span.

Source Range for Rotary 1 set to 50-100%.

5. If you select another device in the Device list to the left in the Editor, you can assign another parameter to the
same Rotary or Button control using the same basic methods.
This means that you can create multi-function controls that operate simultaneously on several parameters. E.g. if
you have two Subtractors and a Malström in a Combi you could create a “master” filter cutoff knob, that controls
this parameter for all three devices. See “Controlling multiple Target parameters differently from a single Source
control” for an example.
D If you want to clear all mappings and start from a clean slate, click the “Clear Mappings” X button:

824 THE COMBINATOR

 Controlling multiple Target parameters differently from a single Source control
You can also have the same Source control assigned to multiple Target parameters on the same device. In the exam-
ple below the “Eng1 Filter Freq” and the “Eng1 Filter Reso” parameters on the Spectral Filter in a Europa device are
assigned to the same FILTER control on the Combinator panel:

The “Eng1 Filter Reso” parameter Range is reduced and reversed (light gray Range bar), so when you turn up the
FILTER knob on the Combinator panel, the Filter Frequency is raised but the Resonance is decreased. The “Eng1 Fil-
ter Reso” parameter also has the Source Range set to 0-50%, which means the Resonance will decrease only during
the first half of the FILTER knob range.
The picture below shows how the “Eng1 Filter Freq” and “Eng1 Filter Reso” parameters are affected when you turn
up the FILTER knob on the Combinator panel. You can also see the differences in the Spectral Filter display:

Controlling both the Freq and Reso parameters from the FILTER control on the Combinator panel.

Controlling a single Target parameter from multiple Source controls

This is also possible, by simply selecting the same Target parameter from different Source controls in the Modulation
Routing section.

825 THE COMBINATOR

 CV Connections

The rear panel of the Combinator.

CV connections between devices in the Combi are saved with the Combi patches. This is also true for CV connec-
tions between devices in Combi and the Combi itself - e.g. if you have connected a Matrix in the Combi to one of the
CV inputs on the back of the Combi panel.
The following CV connectors can be found at the back of the Combinator:

Sequencer Control inputs

The Sequencer Control CV and Gate inputs allow you to play the Combinator from another CV/Gate device (typically
a Matrix or a RPG-8). The signal to the CV input controls the note pitch, while the signal to the Gate input delivers
note on/off along with velocity.

Control CV In

These CV inputs can be used for controlling four selectable controls on the Combinator front panel.
D Click the drop-down and select which panel control you want to modulate.
All currently available front panel controls are shown in the drop-down list.
D Attenuate the input signal with the corresponding attenuation knob.

826 THE COMBINATOR

 Wheel CV In and Source CV In
In this section you will find standard CV Pitch Bend and Mod Wheel modulation inputs, as well as eight modulation in-
puts which can be assigned as Sources in the Editor. Here you can connect external CV modulation sources for mod-
ulating any of the Target parameters in the Editor (see “Using Modulation Routing”).
D Click the Editor button to show the Wheel CV In and Source CV In connectors on the rear panel:

Next to each CV Input are one sensitivity knob and one polarity switch that work as follows:
• The sensitivity knobs can be used for attenuating the CV Input signal.
• The polarity switches should be used for defining the polarity of the CV Input signal.
For example, if you have connected a CV signal from an envelope generator, the switch should be set to Unipolar.
If you are modulating from a standard LFO, the polarity switch should be set to Bipolar.

827 THE COMBINATOR

828 THE COMBINATOR
Chapter 51
Gain Tool
 Introduction

Gain Tool is a utility for managing your signal levels, panning, and routing. It’s particularly useful at the end of a signal
chain, to do complex automation without having to use up the Mixer controls for this task. For Combinator builders
and Reason Rack Plugin users, Gain Tool acts as the fader part of a channel strip and also enables some clever x-
fading for more intricate patch building.

Signal flow
The picture below shows the signal flows for the different input and output modes in Gain Tool:

Gain Input section Output section

Width Pan

Gain Width/Pan

Main Aux Pan Left Pan Right

L R
Main Input Mix Dual Pan

Main X-Fade Aux Main/Split

L R L R L R
Main Input Aux Input X-Fade Router Main Out

L R L R L R
Main Input Aux Input Split Out
: audio signals
: control signal

Gain Tool signal flowchart.

Panel reference
The input mode switch

D Select which input mode you want Gain Tool to operate in:
In Gain input mode Gain Tool works as a signal attenuator and amplifier for a stereo or mono input signal.
In Mix input mode Gain Tool works as a mixer for two stereo or mono input channels.
In X-Fade input mode Gain Tool works as a crossfade mixer between two stereo or mono input channels.

830 GAIN TOOL

 The Gain input mode

D Click the Gain button to activate the built-in amplifier.

D Set the desired gain/attenuation with the Gain knob.
The gain/attenuation level is shown in the Gain Display. The display also shows a percentage figure of the ampli-
fication.
Range: -Inf to +18.1 dB (0-800%)

The Mix input mode

In Mix input mode you can connect two separate stereo input signals (Main and Aux) and control their individual gain.
D Set the desired gain/attenuation for the Main and Aux inputs with corresponding the Gain knobs.
The gain/attenuation is shown in the corresponding Gain Display.
Range: -Inf to +18.1 dB (0-800%)

The X-Fade input mode

In Mix input mode you can connect two separate stereo input signals (Main and Aux), control their individual gain and
also crossfade between the input channels.
D Set the desired gain/attenuation for the Main and Aux inputs with corresponding the Gain knobs.
Range: -Inf to +18.1 dB (0-800%)
D Set the desired crossfade between the input channels with the X-Fade slider.
Level

Main Aux

X-Fade slider position

831 GAIN TOOL

 The output mode switch

D Select which output mode you want Gain Tool to operate in:
In Width/Pan output mode you can set the stereo width of any connected stereo input signals, as well as control
the panning of the input signal(s).
In Dual Pan output mode you can pan the Left and Right channel(s) of the input signal(s).
In Router output mode you can crossfade the output mix to the Main Out and Split Out jacks on the rear panel.

The Width/Pan output mode

D Set the stereo width of the input stereo signal(s) with the Width knob.
At the 12 o’clock position the width of the stereo input signal(s) are unaffected. Turn counter-clockwise to turn the
signal(s) more towards mono. Turn clock-wise to enhance the stereo width of the stereo input signal(s). Since
Width affects the differences between the left and right channels, enhancing a stereo signal will not cause any
phase problems.
! Note that mono input signals are not affected by the Width parameter - mono signals will always remain in
mono.
D Turn the Pan knob to pan the input signal(s) Left or Right.
If you have both the Main and Aux inputs connected, it’s the combined signal that will be panned.

The Dual Pan output mode

D Pan the Left and Right stereo channels individually with the Pan Left and Pan Right knobs.
If both Pan knobs are at their 12 o’clock positions the output signal will be mono.

832 GAIN TOOL

 The Router output mode

D Move the Output slider to crossfade the output signal to the Main Out and Split Out jacks on the rear panel.
Level

Main Split

Output slider position

The Output buttons

To the right of the front panel are a number of buttons.

D Click the Mute button to mute the output signal(s) of both the Main Out and Split Out jacks on the rear panel.
D Click the Inv L and Inv R buttons to invert the phase of the corresponding stereo channel.
Note that the phases of the Split Out signals are also inverted.
D Click the Swap L-R button to swap the left and right output channels.
D Click the Mono button to get the output signal(s) in mono.

The output level meters

To the far right of the front panel are the output level meters.
• In the Width/Pan and Dual Pan output modes the meter shows the output levels of the Left and Right Main Out
channels:

• In the Router output mode the meter shows the output levels of the Main Out and Split Out channels respec-
tively:

833 GAIN TOOL

 Connections

Audio Inputs
Main Input L&R
D Patch the main audio signal(s) here.
If your input signal is in mono, connect only to the L (left) input.

Aux Input L&R

D Patch the additional audio signal(s) that you want to use in Mix or X-Fade mode here.
If your input signal is in mono, connect only to the L (left) input.

CV Inputs
The first three CV inputs control parameters in the Input section. Each CV input also has an attenuator knob:

Main Gain, Aux Gain and X-Fade

D Modulate the Gain parameter in Gain, Mix and X-Fade mode by a CV signal patched to this input.
D Modulate the Aux Gain parameter in Mix and X-Fade mode by a CV signal patched to this input.
D Modulate the X-Fade parameter in X-Fade mode by a CV signal patched to this input.

The last three CV inputs control parameters in the Output section. Each CV input also has an attenuator knob:

Width/Pan L, Pan/Pan R and Output Router

D Modulate the Width parameter in Width/Pan mode - or the Pan L parameter in Dual Pan mode - by a CV signal
patched to the Width/Pan L input.
D Modulate the Pan parameter in Width/Pan mode - or the Pan R parameter in Dual Pan mode - by a CV signal
patched to the Pan/Pan R input.
D Modulate the Output parameter in Router mode by a CV signal patched to the Output Router input.

Audio Outputs
Main Out L&R
D Patch the main audio output signal(s) here.

Split Out L&R

D Patch the Split Out audio output signal(s) that you use in Router mode here.

834 GAIN TOOL

Chapter 52
The Line Mixer 6:2
 Introduction

The Line Mixer 6:2 allows you to control the level, stereo placement (Pan) and effect mix (AUX Send) of each con-
nected audio device.
The Line Mixer is configured with 6 (stereo) input channels, which are combined and routed to the Left and Right
Master outputs.

Channel parameters
The channels are identical and contain an Auxiliary Send, Mute and Solo buttons, a Pan control, and a Level control:

| Item | Description
Level control This controls the output level of each corresponding channel, allowing you to set the desired mix (balance)
between different devices connected to the Line Mixer.
Channel label Each channel in the mixer that has a device connected to it, displays a read-only label with the name of the
device.
Channel meter The meter is a graphical representation of the channel output level. If the signal level pushes the meter into the
range of the red area, try lowering either the output level of the device connected to the channel, or the Level
control itself, to avoid distortion.
Pan control Use this control to set the left/right position of the channel in the stereo field. [Ctrl]-click (Win) or the Pan knob
to set Pan to the default “0” (center position).
Mute (M) and Clicking a channel’s Mute button silences the output of that channel. Click the button again to unmute the
Solo (S) Buttons channel.
Clicking a channel’s Solo button silences all other mixer channels, so that you only hear the soloed channel.
Several channels can be soloed at the same time. If this is the case, note that soloed channels can’t be muted
with the Mute button. To mute one of several channels in solo mode you simply “unsolo” it.
Auxiliary (AUX) Effect Send The AUX Send controls the amount of channel signal that is to be sent to other devices - typically effect
processors. The effect output is then normally returned to the Mixer via the AUX Return input where it is mixed
with the dry (non-processed) signal. If you create an effect device when the Mixer is selected, the effect is auto-
routed to the Send/Return connectors. You can then control the amount of effect that is to be applied to any
device connected to a Line Mixer channel via the AUX Send knob. The Send can be taken pre or post channel
level - see “Auxiliary (AUX) Send”.

The Auxiliary Return section

The AUX Return channels provide inputs for a connected send effect device. The Aux Return channels have a single
level control on the front panel.

Master level
The Master L/R fader controls the summed output level of all channels in the Mixer. Use this to change the relative
level of all channels, to make fade-outs etc.

836 THE LINE MIXER 6:2

 Connections
All input and output connectors are as usual located on the back panel of the Line Mixer.

Mixer Channel Connections

D Each mixer channel features stereo left/right inputs for connecting audio devices.
Use the left input when manually connecting a mono signal source.
D In addition, there is a Control Voltage (CV) input, for voltage controlling channel Pan from other devices.

Auxiliary (AUX) Send

D The Send connectors are used to connect to the inputs of effect devices.
To connect the send to a mono-input device, use the Left (Mono) output.
When a Send is connected to an effects device, the corresponding AUX Send knob determines the level of the signal
sent to the effect device for each channel.
The Send Output can be taken Pre or Post channel level by using the switch to the left of the Send connectors.

Auxiliary (AUX) Return

These are normally connected to the left and right outputs of effect devices.

Master Left/Right Outputs

• The Master outputs are auto-routed to the first available output pair in the I/O device.
If used in a Combinator, the Master Outs are normally connected to the “From Devices” connectors on the Combi-
nator.

837 THE LINE MIXER 6:2

838 THE LINE MIXER 6:2
Chapter 53
Matrix Pattern
Sequencer
 Introduction

The Matrix is a pattern-based device. Matrix doesn’t generate sound on its own, but has to be connected to another
instrument device. It basically works by sending pattern data in the form of Note CV (pitch) and Gate CV (note on/off
plus velocity) or Curve CV (for general CV parameter control) signals to a device or device parameter. The patterns
can be up to 32 steps, and there are 32 memory locations for storing pattern data. The Matrix is monophonic and can
control one voice in an instrument device.
Unlike most other devices in Reason, the user interface of the Matrix is not modeled on any existing hardware equiv-
alent. The hardware devices that could be said to have similar functionality are analog step sequencers, which usually
had rows of knobs that controlled the note pitch and gate values for each step.

About the three Output types

Note values are entered here

Gate values are entered here

Note and Gate CV values.

The Matrix can produce three types of output: Curve CV, Note (Key) CV and Gate CV.
• Note CV normally controls note pitch.
When connected to an instrument device Sequencer Control input, the values correspond to semitone steps.
• Gate CV represents a note-on/off value, plus a level value (that could be likened to velocity).
Both of these two outputs are typically connected to the Sequencer Control Gate and CV inputs on a compatible in-
strument device. For example, if you create a Matrix with either a synthesizer (Subtractor, Malström) or a sampler
(NN-19, NN-XT) selected, they will be auto-routed in this way, and will control one voice in the device.

840 MATRIX PATTERN SEQUENCER

 • Curve CV is a separate pattern, programmed separately from the Note/Key and Gate CV.

Curve CV values (upper window).

This is useful for programming CV curves that control other parameters other than note pitch (although you could do
this too). This way you could control the note pitch and triggering from the Key and Gate outputs for a device, then
add a second independent pattern using the Curve CV output that could control filter cutoff for example.
It should be stressed that all three outputs can be used in any number of ways. For example, you could use the Gate
CV to trigger a drum in Redrum, or let the Curve CV control the feedback parameter of a delay, etc.

Programming patterns
Pattern basics
Matrix contains a built-in pattern sequencer that repeatedly plays back a pattern of a specified length. The typical ex-
ample in the “real world” is a drum machine which plays drum patterns, usually one or two bars in length.
Having the same pattern repeat throughout a whole song may be fine in some cases, but most often you want some
variations. The solution is to create several different patterns and program pattern changes (automatic switching from
one pattern to another) at the desired positions in the song.

Selecting Patterns
The Matrix has 32 pattern memories, divided into four banks (A, B, C, D).

The Bank and Pattern buttons for the Matrix pattern sequencer.
D To select a pattern in the current bank, click on the desired Pattern button (1-8).

841 MATRIX PATTERN SEQUENCER

 D To select a pattern in another bank, first click the desired Bank button (A, B, C, D) and then click the Pattern
button.
Nothing happens until you click the Pattern button.
The pattern change takes effect on the next downbeat according to the time signature set in the transport panel or
on the transport track.

The Pattern Enable switch

The Pattern Enable switch.

Next to the Bank and Pattern buttons you will find an additional switch, which is normally activated. If you click this to
turn it off, the pattern playback will be disabled, starting at the next downbeat - exactly as if you had selected an
empty (silent) pattern. For example, this can be used for bringing Matrix patterns in and out of the mix during play-
back.

Steps
Matrix patterns consist of a number of discrete steps. For each step, you can enter a note, a CV value and a Curve
value. When you run the pattern, each step will be played back in turn and will play a sound or send out the informa-
tion programmed for this step. If you have ever used a drum machine, this will be obvious to you.

Clearing a Pattern
To clear (empty) a pattern, select it and use the Clear Pattern command on the Edit menu or device context menu.
! Note that clearing a pattern doesn’t affect the pattern length, resolution or shuffle settings!

Using Cut, Copy and Paste

By using the Cut, Copy and Paste Pattern commands on the Edit menu or device context menu, you can move or du-
plicate patterns. The following rules apply:
D Copy Pattern makes a copy of the currently selected pattern and places the copy on the clipboard.
D Cut Pattern moves the currently selected pattern to the clipboard.
This is the same as first performing Copy Pattern and then Clear Pattern.
D Paste Pattern copies the pattern on the clipboard to the selected pattern location in the selected device.
This overwrites the selected pattern with the one on the clipboard.

Transferring patterns between Reason Rack Plugin instances

If you want to copy patterns between different Reason Rack Plugin instances songs, you can use copy and paste:
1. Select the pattern you want to copy in the first instance.
2. Select Copy Pattern from the device context menu.
3. In the other instance select the bank and pattern location to which you want to copy the pattern.
Note that any pattern already stored in that location will be overwritten!
4. Select Paste Pattern from device context menu.

842 MATRIX PATTERN SEQUENCER

 Tutorial
The programming procedure of the Matrix is to input note and gate values into the upper and lower fields of the pat-
tern window respectively. You can input values by clicking or dragging in the pattern window. Proceed as follows:
1. Create a Subtractor synth.
You don’t have to use the Subtractor device to use the Matrix, in fact you don’t have to use an instrument device
at all, but for this basic tutorial we will use a “standard” setup.
2. With the Subtractor selected, create a Matrix Pattern Sequencer.
The Matrix Note and Gate CV outputs will now be auto-routed to Subtractors Sequencer Control Gate and CV in-
puts, as you can see if you flip the rack around.

3. Make sure that the switch to the left of the pattern window is set to “Keys” position.
As you can see, there are two rows of red rectangles. The one with horizontal rectangles at the bottom of the up-
per field in the pattern window represent note pitch, for each step in a pattern. At the moment they are all set to
the same note pitch. The row of vertical rectangles in the lower field represent Gate velocity values - currently
these are all set to a velocity value of 100 for all steps.

843 MATRIX PATTERN SEQUENCER

 4. Click inside the upper grid section of the Matrix pattern window.
An orientation line is displayed in the grid to make it easier for you to find the desired note, and the red rectangles
are placed according to where you click. You can drag to input continuous note values.

5. Click and drag in the lower area of the pattern window.

You can create vertical Gate velocity strips of varying heights. The higher the strip, the higher the velocity value.

6. Press the Play button on the Matrix.

The pattern you “programmed” in the previous steps is now repeated. At the top of the pattern window, a red rect-
angle indicates every step of the pattern.
D If you now click or drag in the upper grid section with the pattern playing, you can hear how the note pitches
change.
The note pitch corresponds to the keyboard printed to the left of the pattern window, in a one octave range, and as
previously mentioned, an orientation line is visible when clicking or dragging, making it easy to find the note pitch
on the keyboard.
D If you now click or drag in the lower gate section while the pattern is playing, you can hear how the timbre and
volume changes.
D If you drag some of the vertical rectangles down so that they disappear from view, the corresponding steps of
the pattern are completely silenced.
D By using the 5-way switch below the “Keys/Curve” switch you can input notes in other octave ranges (over five
octaves).
Note that there can only be one note for each step in the pattern.

7. By using a combination of the methods described in the above steps, you can program suitable note values for
each step, decide which steps should be played and set their velocity with the gate values.

844 MATRIX PATTERN SEQUENCER

 Using Curve Patterns
Curve patterns are independent patterns that can be applied separately to the note pattern programmed in “Keys”
mode. If you switch the Keys/Curve switch to “Curve”, the note, but not the gate steps, disappear from view, and
leaves the upper area of the pattern window empty. You can now start programming a curve pattern. Proceed as fol-
lows:
1. Draw a curve, using the same method as for notes or gates.
As you can see, the Curve pattern looks like large vertical gate steps.
D If you play the pattern, nothing has changed, i.e. the pattern sounds exactly like it did before the Curve pattern
was drawn.
This is because the Curve CV output hasn’t been connected to any parameter yet.
2. Flip the rack around so you can see the back panel of the Matrix.
3. Connect the Curve CV output to the Filter Cutoff Modulation Input on the Subtractor.
Now the curve pattern controls the filter frequency of the Subtractor.
q If the effect isn’t very noticeable, try raising the filter Q parameter, and lowering the filter frequency.
D The Curve CV output can be connected to any device CV or Modulation input.
Actually, Curve CV signals can also produce Gate triggers (used for triggering samples or envelopes for example).
D A Gate trigger is produced for each curve pattern step that follows a value of “0”.
If you look at the picture below, steps 2,4 and 6 will produce a trigger, because steps 1, 3 and 5 are set to zero, but
the rest of the pattern would not.

About Unipolar and Bipolar Curves

On the back panel of the Matrix you will find a switch, allowing you to select between “Unipolar” or “Bipolar” Curves.
The difference is as follows:
D A unipolar curve has values starting from “0” and up.
“0” is the value produced by all steps when they are “empty” (not visible). Unipolar is the default setting of this
switch when a new Matrix is created.

Unipolar curve

845 MATRIX PATTERN SEQUENCER

 D A bipolar curve is divided in the “middle”, with the middle representing a value of “0”.
The curve reflects this. If no curve has been drawn and you switch to bipolar mode, all steps go from the bottom up
to the middle of the scale printed to the left of the pattern window. Thus, all steps are at “0”, and the curve can be
drawn both up and down from the middle.

Bipolar curve
Bipolar curves are essential in some instances. If you want to use the Matrix to CV control the Pan parameter for a
mixer channel for example, a unipolar curve would start at zero - which for Pan equals center position. This means
that you would only be able to use the curve to pan in one direction from this center position. A bipolar curve however,
will have the zero value in the middle, allowing you to draw pan curves in both directions. Bipolar curves can also be
used for controlling parameters with positive and negative values.

Setting Pattern Length

You may want to make settings for Pattern length, i.e. the number of steps the pattern should play before repeating:
D The “Steps” spin controls are used to set the number of steps you wish the pattern to play.
The range is 1 to 32. You can always extend the number of steps at a later stage, as this will merely add empty
steps at the end of the original pattern. You could also make it shorter, but that would (obviously) mean that the
steps you remove won’t play back. The steps you remove aren’t erased though, if you set the step number back
again, anything recorded in the previously removed step locations will be played back.

Using Tied Notes

If you activate “Tie” to the left of the Gate pattern window, you can create longer notes (eighth notes, quarter notes
etc.). A quick way to draw tied gates is to hold down [Shift] when you input the gate values.

Entering tied gate values.

D Each step that has one tied gate value will be twice the length compared to a normal step.
Tied gate steps are indicated by being twice as wide in the pattern window.
D If two or more notes of the same pitch are tied together, the result will be even longer notes.
Tied notes are also essential if you want to create typical TB-303 “Acid”-type lead lines - see “Programming “Acid
Style” lead lines”.

846 MATRIX PATTERN SEQUENCER

 Setting Pattern Resolution
Matrix always follows the tempo setting on the transport panel, but you can also make Matrix play in different tempo
“resolutions” in relation to the tempo setting.

Pattern Shuffle
Shuffle is a rhythmic feature, that gives the music a more or less pronounced swing feel. It works by delaying all six-
teenth notes that fall in between the eighth notes.

Example of straight sixteenth note pattern (viewed in the main sequencer).

The same sixteenth note pattern with shuffle applied.

You can activate or deactivate shuffle individually for each pattern in a pattern device. However, the amount of shuffle
is set globally with the Shuffle control in the I/O device.

Pattern Mute
If you deactivate the “Pattern” button above the Pattern select buttons, the pattern playback will be muted, starting at
the next downbeat (exactly as if you had selected an empty (silent) pattern). For example, this can be used for bring-
ing different pattern devices in and out of the mix during playback.

Pattern Functions
When a pattern device is selected, you will find some specific pattern functions on the Edit menu (and on the device
context menu).

Shift Pattern Left/Right

The Shift Pattern functions move the notes and corresponding gate values in a pattern one step to the left or right.

Shift Pattern Up/Down

! This function does not alter the Curve CV. This is because the values produced by the Curve CV do not neces-
sarily correspond to semitone note steps at all.
The Shift Pattern functions will transpose all the notes in a pattern one semitone up or down.

Randomize Pattern
The Randomize Pattern function create random patterns. These can often be great starting points and help you get
new ideas. Both Note, Gate and Curve CV values will be created.

847 MATRIX PATTERN SEQUENCER

 Alter Pattern
The Alter Pattern function modifies existing patterns. Note that there must be something in the pattern for the func-
tion to work on - using an Alter function on an empty pattern will not do anything.
! Note that Randomize and Alter affects both the Gate, Note and Curve CV!

Chaining Patterns
When you have created several patterns that belong together, you will most probably want to make these play back
in a certain order. This Pattern selection can be automated in the main sequencer, just like any other parameter au-
tomation.

Copy MIDI Notes to a sequencer track

It’s possible to copy the MIDI Notes of the currently selected Pattern to a track in your DAW’s sequencer:
1. Click to select the desired Pattern.
2. Click the “Drag MIDI Notes to track in sequencer” icon and drag to the destination track in the sequencer:

! Be sure to disable the Enable Pattern Section function on the Matrix panel afterwards, to avoid “doubled
notes” during playback.
! Curve patterns cannot be converted to sequencer data! Only the note pattern and the gate values will be con-
verted.

848 MATRIX PATTERN SEQUENCER

 Example usage
As mentioned previously, the Matrix is a very flexible device. Here follows a few examples of how you can use the
Matrix Pattern Sequencer.

Using the Matrix for modulation

You can effectively use the Matrix as a modulation source, much like an LFO. Just like the LFOs in Reason’s instru-
ment devices, the Matrix can generate modulation that is synchronized to tempo, which has many advantages. Pro-
ceed as follows:
1. Create a Synthesizer (e.g. a Subtractor or Malström).
2. Create a Matrix Pattern Sequencer, or if one already exists, set it to an empty pattern.
3. Flip the rack around and disconnect the Gate and Note CV outputs.
Gate/Note CV is not used in this example
4. Flip the rack around and connect the Curve CV output on the Matrix back panel to the “Amp Level” Modulation
input on the synthesizer.
This parameter is used for modulating the output level (volume) of the synthesizer. Volume modulation is often re-
ferred to as Tremolo. You can use a unipolar curve (see “About Unipolar and Bipolar Curves”) for this example.
5. Flip the rack back again, and switch the Matrix to display the Curve pattern window.
6. Draw a curve like the one shown in the illustration below.
If you use fewer or more steps than16 (as shown in the picture), just draw the curve so that it roughly matches the
shape in the picture.

7. Activate Click on the Transport Panel.

8. Select the track that is routed to the synthesizer, so that you can play it from your MIDI keyboard.
9. Activate Play on the transport panel, and hold a chord down on your keyboard.
You should now hear the volume being modulated by the Curve pattern.
10.While still in play mode, you can use the Resolution knob to change the modulation “rate” in relation to the
tempo.
For each clockwise resolution step the modulation speed is doubled and vice versa, but it will always stay in sync
with the tempo.

849 MATRIX PATTERN SEQUENCER

 Programming “Acid Style” lead lines
By “acid style” lead lines we mean patterns that use a combination of Legato and slide (or portamento) effects to pro-
duce the widely used hypnotic “wavy” sound produced by the original Roland TB-303, and recreated in the Propeller-
head Software product ReBirth. To approximate this typical sound using Reason, proceed as follows:
1. Create a Synthesizer (Subtractor or Malström).
2. Create a Matrix Pattern Sequencer, or if one already exists, set it to an empty pattern.
3. Make sure that the Note and Gate CV outputs are connected to the synthesizers Sequencer Control CV and
Gate inputs, respectively.
4. For Subtractor, select either a Init Patch, or use the “TB Synth” patch in the Monosynth category of the Reason
Factory Sound Bank.
D If you use an Init patch, it is important that you make the following settings:
• Set Polyphony to “1”.
• Switch Trigger Mode to “Legato”.
• Set Portamento to a value around “50”.
5. Create a pattern in Matrix, and keep it playing back.
D If “Tie” (see “Using Tied Notes”) now is activated for a step, the note will be tied to the next and the pitch will
continuously “glide” to the pitch of the following step.
Please note that Tie should be activated on the note you wish to slide from, and not the note you slide to.
D If you have several tied notes, one after the other, they will play as one long legato phrase. This can be used to
create “wavy” lead lines with pitch bend effects.
6. Experiment with different Note, Tie and Gate values.
If you have ever used a TB-303 or ReBirth, you should now begin to get the hang of how you can create patterns
in that particular style by using the Matrix together with a synthesizer.
q Adding a DDL-1 (delay), and a D-11 (distortion) effect device will make it sound even more “ReBirth”-like, but
of course you are also able to get a much wider range of timbres by utilizing Reason’s other sound and modu-
lation capabilities.

Triggering samples
The Gate CV output can be used to trigger samples, either in Redrum or in the NN-19 or NN-XT Sampler.
D Connect the Matrix Gate CV out to the Gate (Sequencer Control) in on the NN-19/NN-XT or to one of the indi-
vidual Gate Channel inputs of Redrum.
Gate values will now trigger the sample on each step with Gate values above “0”.

850 MATRIX PATTERN SEQUENCER

Chapter 54
Mixer 14:2
 Introduction

The Mixer 14:2 allows you to control the level, stereo placement (Pan), tone (EQ) and effect mix (AUX Sends) of
each connected audio device.
If you have ever used a conventional hardware audio mixer, you will most likely find the Mixer very straightforward to
use. It is configured with 14 (stereo) input channels, which are combined and routed to the Left and Right Master
outputs. The vertical channel “strips” are identical and contain - from the top down - four Auxiliary Sends, an EQ sec-
tion, Mute and Solo buttons, Pan control, and a Level fader.
Every mixer parameter can of course be automated.
! Please, note that this device is not available in Reason Lite Rack Plugin.

The Channel Strip

AUX Sends 1-4

Pre-fader switch for AUX Send 4

EQ On/Off button

EQ Treble and Bass controls

Mute (M) and Solo (S) buttons

Pan control

Channel Fader

Channel Meter
Channel Label

Each channel strip in the Mixer 14:2 contains the items listed in “Channel Strip Controls”.

852 MIXER 14:2

 Channel Strip Controls
| Item | Description | Range
Channel Fader The channel fader is used to control the output level of each correspond- 0 - 127
ing channel.
By adjusting the faders, you can set the desired mix (balance) between
different devices connected to the Mixer.
Channel Label Each channel in the mixer that has a device connected to it, displays a N/A
read-only label with the name of the device to the left of the fader.
Channel Meter The meter is a graphical representation of the channel output level. If the N/A
signal level pushes the meter into the range of the red area, try lowering
either the output level of the device connected to the channel, or the
channel fader itself, to avoid distortion.
Pan Control Use this control to set the left/right position of the channel in the stereo -64 – 0 – 63
field. [Command]/[Ctrl]-click the Pan knob to set Pan to the default “0”
(center position).
Mute (M) and Clicking a channel’s Mute button silences the output of that channel. On/Off
Solo (S) Buttons Click the button again to unmute the channel.
Clicking a channel’s Solo button silences all other mixer channels, so that
you only hear the soloed channel. Several channels can be soloed at the
same time, but if this is the case, note that soloed channels can’t be
muted with the Mute button. To mute one of several channels in solo
mode you simply “unsolo” it.
EQ Treble and Bass controls The EQ Treble and Bass controls is used to cut or boost the higher and Treble: +/- 24 dB at 12 kHz.
lower frequencies of the signal, respectively. Click on the EQ button to Bass: +/- 24 dB at 80 Hz.
activate the EQ.
If you need more advanced EQ, you can always use a PEQ2 parametric
EQ as an insert effect for a device.
Note also the two EQ modes - see “About the EQ modes”.
Auxiliary (AUX) Effect Send 1-4 The four independent AUX Sends control the amount of channel signal 0 - 127
that is to be sent to other devices - typically effect processors. The effect
output is then normally returned to the Mixer via the AUX Return inputs
(see “The Auxiliary Return Section”) where it is mixed with the dry (non-
processed) signal. If you create an effect device when the Mixer is se-
lected, the effect is auto-routed to the first available Send/Return con-
nectors. You can then control the amount of effect that is to be applied
to any device connected to a Mixer channel via the corresponding AUX
Send knob. The AUX Send outputs are taken post channel fader, but you
have the option of selecting Pre-fader mode for AUX Send 4 (by clicking
the “P” button next to the send so that it lights up). In that mode, the send
level is independent of the channel fader.
The sends are in stereo but can be used in mono as well.

853 MIXER 14:2

 The Mixer signal flow
The basic signal flow for a channel in the Mixer 14:2 is as follows:
Aux 4 pre-fader
mode

Input EQ Pan Mute Fader AUX Sends

AUX Returns Effects Devices

Solo Master L/R Output

Note that the Solo function is true “in-place” solo, meaning that if the channel uses Auxiliary sends routed to effect
devices, the soloed output signal will also include the soloed channel(s) including any Aux Send effects.
Note also that if the pre-fader send mode is activated for Aux 4 the send is tapped after the EQ and Pan controls but
before the channel fader.

About the EQ modes

With Reason 2.5, the EQ modules in the Mixer were improved to get an even better sound and character. However, if
you want to emulate the character of older Reason versions, you may want to use the “old” EQ mode to ensure that
it sounds exactly the same.
On the back of the Mixer 14:2 you will find a switch for this - select “Improved EQ” for the new EQ types or “Compat-
ible EQ” for the old-style EQ. The parameters are exactly the same in both cases.

854 MIXER 14:2

 The Auxiliary Return Section

The Auxiliary Returns provide an “extra” four stereo inputs in addition to the Mixer 14:2’s 14 stereo channels. The
main function of Return channels is to provide inputs for connected Send effects devices. Each Aux Return channel
has a level control, and a read-only tape label that display the name of the device connected to the Return channel.

The Master Fader

The Master L/R fader controls the summed output level of all channels in the Mixer 14:2. Use this to change the rel-
ative level of all channels, to make fade-outs etc.

Connections
All input and output connectors are as usual located on the back panel of the Mixer 14:2. Special connectors are
used for “chaining” two or more Mixer 14:2 devices together. This is described on “Chaining several Mixer 14:2 de-
vices”.

Mixer Channel Connections

D Each mixer channel features stereo left/right inputs for connecting audio devices.
Use the left input when manually connecting a mono signal source.
D In addition, there are two Control Voltage (CV) inputs (with associated voltage trim pots), for voltage con-
trolling channel Level and Pan from other devices.

855 MIXER 14:2

 Auxiliary (AUX) Send Out

D There are four stereo Send Out connectors, which normally are used to connect to the inputs of effect devices.
To connect a send to a mono-input device, use the Left (Mono) output.
When a Send is connected to an effects device, the corresponding AUX Send knob determines the level of the signal
sent to the effect device for each channel. The Send Output is taken post-channel fader but you have the option of
selecting pre-fader mode for AUX Send 4.
D Note that some effects (for example the Comp-01 compressor or the PEQ2 parametric EQ) are effect types
which are not designed to be used as AUX Send effects, but rather as insert effects, where the whole signal is
passed through the effect.
Alternatively, you could use AUX Send 4 in pre-fader mode and lower the channel fader completely.

Auxiliary (AUX) Returns

D There are four stereo Return input connectors.

These are normally connected to the left and right outputs of effect devices.

Master Left/Right Outputs

• The Master outputs are auto-routed to the first available output pair in the I/O device.
If used in a Combinator, the Master Outs are normally connected to the “From Devices” connectors on the Combi-
nator.
• In addition to the Master Out connectors, there is a Control Voltage (CV) input (and an associated trim pot), for
voltage controlling the Master Level from another device.

856 MIXER 14:2

 Chaining several Mixer 14:2 devices

Two chained Mixer 14:2 devices are connected like this, the top Mixer being the “Master” Mixer.
If you want more Mixer channels, you can chain several Mixer 14:2 devices.
D Select the existing Mixer 14:2 device and choose “Create:Mixer 14:2” from the context menu.
The new Mixer is automatically connected via the “Chaining Master” and “Chaining Aux” connectors of the se-
lected Mixer.
• The newly created Mixer’s Master Output is connected to the original Mixer’s Chaining Master input.
The Master Out Level for the new Mixer is now controllable from the original Mixer’s Master fader - so that this
fader now controls the Master output level of both mixers.
• The newly created Mixer’s four stereo Aux Send outputs is connected to the original Mixer’s Chaining Aux con-
nectors.
The new Mixer will now have access to any Aux Send effects connected to the original Mixer, via the same corre-
sponding Aux Send(s).
This way, the two Mixers operate as “one”.
! One exception is the Mute/Solo function, which is not chained. Thus, soloing a channel in one of the Mixers,
will not mute the channels in the other Mixer.
You can create as many Mixers as you like, they will be chained in the same way, with one Mixer remaining the “mas-
ter” (i.e. it controls the Master level of all chained Mixers and supplies the Aux Send effect sources).

Partially or Non-Chained Mixers

You can also have several Mixers that are only partially or not chained at all.
• You may for example wish to have different Aux Send effects for one Mixer.
Then simply disconnect one or more of the Send Out to Chaining Aux connectors, and assign new Send effects.
D You could for example send the Master output of one Mixer to another output pair on the I/O device, instead
of the Chaining Master inputs.

857 MIXER 14:2

858 MIXER 14:2
Chapter 55
Pulsar Dual LFO
 Introduction

The Pulsar Rack Extension device is a very flexible and versatile dual LFO module. LFOs (Low Frequency Oscillator)
are used for generating cyclic modulation signals. A typical example is to have an LFO modulate the pitch of an os-
cillator to generate vibrato, but there are countless of other applications for LFOs.
Pulsar features two separate LFOs that can be used for modulating parameters in other rack devices. The two LFOs
can also modulate each other to generate complex modulation signals. The LFOs in Pulsar can reach way up in the
audible frequency range, which opens up for really interesting applications. As an additional feature the LFO rates
can also be tracked from a MIDI keyboard.
In Reason the Pulsar Rack Extension device can be found on the Utilities palette.
! Please, note that this device is not available in Reason Lite Rack Plugin.

Panel parameters
LFO 1&2 common parameters
Rate

This controls the LFO rate. The Rate range in Pulsar is very wide and can reach way up in the audio frequency range.
The rate is indicated by the lamp to the left above the Rate knob. The rate can also be synced to the sequencer
tempo by clicking the Tempo Sync button below the Rate knob (see “Tempo Sync” below). In sync mode, the Rate
knob controls the sync resolution.
The LFO 1 Rate can be modulated and/or synced by LFO 2, see “Rate (LFO 2 to LFO 1 Rate)” and “Sync”. The Rates
can also be modulated from the Envelope, see “Envelope”. As a special feature, the Rates can also be controlled from
a MIDI keyboard, see “KBD Follow”.
Range: 0.06Hz-1.05kHz (synced: 32/4 to 1/64th)
! When the Rate is modulated, it can reach far beyond the default frequency range.

Waveform selectors

Here you can select one of nine different LFO waveforms. Besides the standard waveforms (sine, triangle, pulse, etc.)
there are random, slope and stepped waveforms. The shape of the waveforms are shown in the display.
! Note that all waveforms are bipolar, i.e., they generate both positive and negative levels.

860 PULSAR DUAL LFO

 D Select waveform for each LFO by clicking the up/down arrow buttons or by clicking and holding in the wave-
form display and dragging up/down.

Level

Here you set the output level of the LFO signal. The LFO 1 Level can be modulated by LFO 2, see “Level (LFO 2 to
LFO 1 Level)”. The Levels can also be modulated by the Envelope, see “Envelope”.

Phase

The Phase control lets you offset the phase of the LFO cycle, i.e. decide where in the cycle the waveform should
start. The range of the Phase control is 0-360 degrees:

Phase=0 Phase=90 Phase=180 Phase=270

Shuffle

The Shuffle function affects two adjacent LFO cycles, in pairs. Increasing the Shuffle value lengthens the first cycle
and shortens the second one:

Shuffle=50% Shuffle >50%

Note that the total length of the cycle pair is always 2 regular cycles, which means that shuffling will work great also
in Tempo Sync mode, see “Tempo Sync”.
Range: 50% (no shuffle) to 75%

861 PULSAR DUAL LFO

 Lag

The Lag control acts like a lowpass filter on the LFO signal, making the signal smoother. This is especially noticeable
on waveforms with sharp edges or transients like the square, sawtooth and stepped waves. On the sinewave you will
barely notice any effect since it’s already smooth by nature.

Tempo Sync

Click the Tempo Sync button to sync the LFO Rate to the sequencer tempo. This is great for creating animated
effects in sync with the song’s tempo. In Tempo Sync mode, the Rate knob controls the sync resolution, see “Rate”.

LFO 1 specific parameters

ENV Sync

With ENV Sync active, whatever triggers the Envelope will also restart LFO 1, see “Envelope”.
If Tempo Sync is also active (see “Tempo Sync”) the LFO will continue to sync to the sequencer tempo. However,
when you actually press a key or hit the Trig button will affect the LFO cycle start. In some situations this could be
perceived as the LFO is losing sync but it’s not; it’s merely the LFO cycle start that is changing.

LFO 2 specific parameters

On/Off

Click this to activate LFO 2.

LFO 2 to LFO 1 modulation parameters

Rate (LFO 2 to LFO 1 Rate)

This controls the modulation amount from the LFO 2 signal level to LFO 1’s Rate. The result is frequency modulation
(FM) of LFO 1.

862 PULSAR DUAL LFO

 Level (LFO 2 to LFO 1 Level)
This controls the modulation amount from the LFO 2 signal level to LFO 1’s Level. The result is amplitude modulation
(AM) of LFO 1.

Sync
With the Sync button on, every new LFO 2 cycle automatically restarts LFO 1.

Envelope

This is an AR (Attack-Release) envelope. The Envelope can be triggered from any of these four sources:
• The Trig button, see “Trig”.
• LFO 2, see “LFO 2 Trig”.
• Envelope Gate In modulation input on the rear panel, see “Envelope Gate In”.
• MIDI Note On from a connected MIDI keyboard.
The Envelope can modulate the Levels and Rates of LFO 1 and/or LFO 2.
! If the envelope is retriggered before all envelope stages are completed, the envelope will simply restart at the
current level (similar to how a monophonic synthesizer works).

LFO 2 Trig
With the LFO 2 Trig button on, the Envelope is automatically synced by the LFO 2 signal. This means that every time
LFO 2 begins a new cycle, the envelope is triggered.

Trig
This is a non-latching gate button which gates/triggers the envelope.

Attack and Release

The attack and release times of the envelope when the gate opens/closes are set with these knobs. For a gated
effect set both controls to zero and control the gate time with the Trig button, CV Gate In modulation input or MIDI
Notes. You can also let LFO 2 trig/gate the envelope as described above.
Range: 0.1ms-3.00s (Attack) and 0.0ms-10.00s (Release)

863 PULSAR DUAL LFO

 Rate (LFO 1 and 2 Envelope modulation)
The Rate knobs control the modulation amount of the Rate parameters of LFO 1 and 2 respectively. With these you
can force the LFO rates to vary according to the envelope’s Attack and Release times:
Env start Env end

LFO signal

Env Attack = 0, Env Release > 0, Rate modulation = +50%

The Rate modulation controls are bipolar, with no modulation at 12 the o’clock position, negative modulation to the
left and positive to the right. Negative modulation means that the LFO Rate gets slower during the envelope stage
and then goes back to the set Rate:
Env start Env end

LFO signal

Env Attack = 0, Env Release > 0, Rate modulation = -50%

864 PULSAR DUAL LFO

 Level (LFO 1 and 2 Envelope modulation)
The Level knobs control the modulation amount of the Level parameters of LFO 1 and 2 respectively:
Env start Env end

LFO signal (50% Level)

Env Attack = 0, Env Release > 0, Level modulation = +50%

The Level modulation controls are bipolar, with no envelope modulation at 12 the o’clock position, negative
modulation to the left and positive to the right. Negative modulation means that the LFO signal decreases in level
during the envelope stages and then goes back to set Level:
Env start Env end

LFO signal (50% Level)

Env Attack = 0, Env Release > 0, Level modulation = -50%

KBD Follow

It’s possible to control the LFO Rates in Pulsar from a connected MIDI keyboard:
D Create a sequencer track for Pulsar by selecting “Create Track for Pulsar n” from the context menu (right-click
the device in the rack).
A sequencer track is created and is automatically selected.

865 PULSAR DUAL LFO

 With the KBD Follow knob you can define how the LFO Rates should be modulated by incoming MIDI Note data. The
KBD Follow parameter is bipolar:
• At zero (12 o’clock position), the Rates are not modulated at all.
• At 100 the LFO Rates track the incoming MIDI Notes 1:1, which means you can “play” the LFOs chromatically
from a connected MIDI keyboard.
• At -100 the LFO Rates tracks the incoming MIDI Notes “backwards” chromatically, which means the LFOs run
slower the higher up on the keyboard you play, and vice versa.
The center MIDI note, i.e. where the LFO Rates “intersect” with the MIDI Note, is C3.
If Tempo Sync is enabled for the LFO(s) you will be able to do cool wobble effects by playing various notes on your
MIDI keyboard, see “Tempo Sync”.

Modulation inputs and outputs

LFO 1&2 input sections

Rate
Use this for dynamically modulating the Rate of the corresponding LFO. Attenuate the input signal with the knob.

Phase
Use this for dynamically modulating the Phase of the corresponding LFO. Attenuate the input signal with the knob.

Shuffle
Use this for dynamically modulating the Shuffle amount. Attenuate the input signal with the knob.

Amount
Use this for dynamically modulating the Level of the corresponding LFO. Attenuate the input signal with the knob.

866 PULSAR DUAL LFO

 LFO 1&2 output sections

CV
There are two CV signal outputs for each LFO, plus two additional outputs for the CV signal phase inverted.

Audio
There are two Audio signal outputs for each LFO, plus two additional outputs for the Audio signal phase inverted.
The difference between the Audio outputs and the CV outputs is that the signals on the Audio outputs have higher
quality for use in e.g. audio processing applications.

Output LFO 1+2

CV
There is one CV signal output for the LFO 1 and 2 signals summed with each other.

Audio
There is one Audio signal output for the LFO 1 and 2 signals summed with each other at audio quality.

Envelope connections

Envelope Gate In
A CV signal with a value > 0 present on this input will gate the Envelope (see “Envelope”). When the CV input signal
decreases to zero or below, the gate is opened (deactivated).

Envelope CV Out
There is one CV signal output for the Envelope signal.

867 PULSAR DUAL LFO

 Tips and Tricks
Patch between LFO 1 and LFO 2 on the back for more flexibility
• By connecting one LFO CV output to the Phase CV input of the other LFO you can achieve some very interest-
ing rhythms. Try using a square wave and set the CV Trim rotary to around 25%.
• Try connecting an LFO CV output to the Shuffle CV input of the other LFO. Use the sawtooth waveform, at a
slow Rate, in the first LFO to gradually add more shuffle over time.
• With the Envelope CV Out you can control both Shuffle and Phase from the Envelope and with the Envelope
CV In you can trigger this effect with any CV signal.

Using Pulsar as a monophonic synth

Because the LFO 1&2 can work in audio rate and can be tracked from a MIDI keyboard, it's possible to use Pulsar as
a two oscillator mono synth with a lot of character. Here's how to set this up:
1. Create Reason Rack Plugin as an instrument and add a Pulsar device.
2. Make sure the level of the LFO(s) you want to use is completely turned down.
3. Connect the Audio output of the LFO(s) you want to use to the I/O device.
4. Turn KBD Follow up to 100%.
5. Turn up the Level knob(s) in the Envelope section for the LFO(s) you want to use.
This decides how much the level of the LFO(s) will be affected by the envelope when you play.
6. You can now play Pulsar from your MIDI keyboard!
! You have to tune Pulsar manually by adjusting the Rate(s) of the LFO(s) you are using. You can set the Rate
knob(s) to max (1.05kHz) to be completely in tune - however, the Pulsar will then be 2 octaves above played
MIDI note. Lower octaves can be found at around 524Hz and 263Hz (hold [Shift] to fine tune).
! Note that Pulsar is an LFO device and therefore doesn't feature the standard sound improvement features of a
synth device (anti-aliasing filters etc.).
q By using LFO 2 to LFO 1 Rate modulation you can modulate the frequency of LFO 1, resulting in frequency
modulation (FM).
q By using LFO 2 to LFO 1 Level modulation you can modulate the level of LFO 1, resulting in amplitude
modulation (AM).
q By modulating the Shuffle you can achieve a sound similar to pulse width modulation (PWM), great for harsher
tones!

868 PULSAR DUAL LFO

Chapter 56
RPG-8 Arpeggiator
 Introduction

An arpeggiator generates rhythmic note patterns (arpeggios) from notes or chords. The RPG-8 Arpeggiator doesn’t
generate sound on its own, but has to be connected to another instrument device (just like the Matrix). It works by
converting MIDI note data (input to the RPG-8) to Note CV (pitch) and Gate CV (note on/off plus velocity) signals.
These CV/Gate signals are sent to the corresponding Sequencer Control inputs of an instrument device.
In addition to standard arpeggiator features the RPG-8 is equipped with a 16 step pattern editor for creating rhyth-
mic variations.
The RPG-8 is monophonic and can control one voice in a connected instrument device.
! Please, note that this device is not available in Reason Lite Rack Plugin.

870 RPG-8 ARPEGGIATOR

 Using the RPG-8
Setting up
The basic procedure is to input note data, either live or recorded, to the RPG-8 device. This note data is in turn sent
to a target device via its Sequencer Control CV/Gate inputs. The resulting output from the target device can either be
arpeggiated notes or simply mirror what is played on your control surface device. Proceed as follows:
1. Create a Combinator device.
2. Drag an instrument device, e.g. a Subtractor to the Combinator “container”.
Select a suitable patch, preferably one with a short attack time.
3. Drag an RPG-8 Arpeggiator from the Utilities palette and drop below the Subtractor.
The RPG-8 Note and Gate CV outputs will be auto-routed to the instrument device Sequencer Control Gate and
CV inputs, as you can see if you flip the rack around. In addition, the Mod Wheel and Pitch Bend CV outs are also
auto-routed to the corresponding modulation inputs on the target device.

4. Flip the rack and click the “Show Programmer” button on the Combinator panel.

871 RPG-8 ARPEGGIATOR

 5. Click the Subtractor in the Combinator display and deselect “Receive Notes”.
Since all instrument devices and such automatically gets MIDI input in Reason Rack Plug, you would get MIDI
notes both to RPG-8 and Subtractor here. By deselecting “Receive Notes” for the Subtractor, only the RPG-8 will
receive MIDI.

6. Make sure the Arpeggiator Enable (“On”) button on the upper part of the panel is activated.

7. Play a few notes.

The notes in the chord you play are now arpeggiated for as long as you hold down the keys. The arpeggio will
change directly if you release all notes and play another note or chord. If you add notes while holding down a
chord, the arpeggio will continue with the added notes.

• The display shows the notes played by the arpeggio pattern, with small bars indicating pitch for each step. The
display is continuously updated as you play.

872 RPG-8 ARPEGGIATOR

 • The arpeggio will play in sync with song tempo by default in new RPG-8 devices.
• You do not need to start playback to generate arpeggios.
D Try changing the arpeggiator mode using the Mode knob.
The various modes govern how the notes are arpeggiated. For example “Up” means from lowest note to highest
note, Up+Down from lowest to highest note, then back down to lowest note again. For a description of all the
modes see “Mode switch”.
D By activating the Hold button the arpeggio will continue to play even if you release the keys.
If you play a new chord the arpeggio will continue to play, using the new notes. To stop the arpeggio, deactivate
the Hold button or hit Stop on the transport. See “Hold On/Off”.
D The “Insert” buttons can be used to introduce further variations to the arpeggio.
See “Insert buttons”.
D If you change the Rate parameter the rate of the arpeggiated notes will follow the song tempo at the selected
note value resolution.
Straight, triplet and dotted note values are available. The arpeggiator can also be free running, i.e. not synced to
tempo - see “Rate”.
D The Gate Length knob allows you to adjust the length of the arpeggiated notes.
If the knob is turned fully clockwise (“Tie”), the gate is always open (the notes will be played legato). If it is turned
fully counter-clockwise, the gate is closed (no sound). See “Gate Length”.
D By using the Octave buttons beside the Mode knob you can increase or decrease the octave range of the ar-
peggio.
With “1 Oct” selected the arpeggiated notes will be those that you press down on the keyboard. If “2 Oct” is se-
lected the range will be expanded so that the arpeggiated notes play over two octaves, and so on - see “Octave
range buttons”.
D You can transpose the arpeggio up or down in octave steps using the Octave Shift buttons.
See “Octave Shift”.
D By using the Velocity knob you can set the arpeggio to play with a fixed velocity value (selectable between 1 to
127) for all notes. Turning the knob fully clockwise to the “Manual” position means that the arpeggio notes will
use the same velocity as the MIDI notes you input.
By using the Manual mode and varying the velocity for individual notes the arpeggio will become more animated
and rhythmic. See “Velocity”.
That covers the basic principles of how to set up and use the RPG-8 to arpeggiate notes that you play in real time.
But to make full use of the RPG-8 there are many further applications:
D You can introduce rests for more complex rhythmical arpeggios by using the Pattern editor.
See “Pattern editor” for a description.
D You can use the RPG-8 as a MIDI to CV converter which allows you to freely assign common performance MIDI
controllers like Mod Wheel and Aftertouch to control parameters - see “Using the RPG-8 as a MIDI to CV con-
verter”.

873 RPG-8 ARPEGGIATOR

 RPG-8 Parameters
MIDI-CV Converter parameters

The MIDI-CV Converter section to the left contains parameters that affect the CV output from the RPG-8, regardless
of whether the Arpeggiator section is activated or not. The following parameters are available:

Velocity
The Velocity knob can be used to set a fixed velocity value for the notes that are output via the Gate CV Out jacks on
the back of the RPG-8. If you set the Velocity knob to a value between “0” and “127”, the Gate CV Out will be fixed
(at the set value) regardless of the velocity of the incoming MIDI notes.
Turning the knob fully to the right activates Manual (“Man.”) mode (a LED is lit when activated). In Manual mode the
velocity levels will be sent out via the Gate CV Out with the same velocity value as they are input, i.e. “what goes in,
will come out”. Manual mode is on by default in new devices.
There is also a “Velocity CV” input at the back. If this is connected to a controller source (a LFO modulation output for
example), the output will be a merge between the Velocity setting and the applied CV modulation by the LFO - see
“CV Inputs”.

Hold On/Off
If the Hold parameter is activated (lit button), an arpeggio will continue to run even if you release all keys. It will con-
tinue to arpeggiate the last notes played until a new note-on is received.
• If you continue to hold down at least one key when Hold is on, any new notes will be added to the existing ar-
peggio as opposed to starting a new arpeggio.
• If the Arpeggiator section is off, and the Hold function is activated, there will be no note-offs for incoming
notes played (i.e. the CV Gate stays open).
• The Hold On/Off status responds to Sustain Pedal messages - as long as the pedal is pressed down, Hold will
stay activated.

Octave Shift
This allows you to transpose the RPG-8 Note CV output in octave steps. You can octave shift up or down 3 octaves.
Octave Shift can also be CV controlled.

874 RPG-8 ARPEGGIATOR

 Arpeggiator parameters

The middle section contains the Arpeggiator parameters that govern how the arpeggio is played. The following pa-
rameters are available:

Arpeggiator “On” button

This switches the Arpeggiator on or off.

Mode switch
This determines the direction of the arpeggio notes.

| Mode | Description
Up This will generate an arpeggio that plays from the lowest note to the highest note.
Up+Down Notes are played from lowest note to highest, then from highest back down to the lowest. The very lowest
and the highest arpeggio notes are not repeated. I.e. the arpeggiator will play the lowest note to the sec-
ond highest note, then the highest note to the second lowest note.
Down Notes are played from the highest note to the lowest note.
Random The notes you input will be arpeggiated randomly.
Manual Notes are arpeggiated in the same order they are played when input.

Octave range buttons

The Octave buttons allow you to set the octave range of the arpeggio.
Use as follows:

| Octave range | Description

1 Oct The arpeggiated notes will be those that you press down on the keyboard.
2 Oct If you select this, the arpeggio will be extended to a two octave range, i.e, the arpeggio pattern is played
then repeated one octave up. In Random mode, the notes you input are played in random order over 2 oc-
taves.
3 Oct Same as 2 Oct but extended to a three octave range.
4 Oct Same as 2 Oct but extended to a four octave range.

875 RPG-8 ARPEGGIATOR

 Insert buttons
Insert can be used to add variations to the arpeggio by repeating certain notes in a predetermined order. It works as
follows:

| Insert mode | Description

Off No Insert repeat.
Low The lowest note is repeated in between every second note in the arpeggio.
Hi The highest note is repeated in between every second note in the arpeggio.
3-1 The arpeggio will play 3 notes forward, then step 1 note back and from there play 3 notes forward etc.
4-2 The arpeggio will play 4 notes forward, then step 2 notes back and from there play 4 notes forward etc.

Rate
This sets the rate of the arpeggio. There are two basic modes for the Rate parameter:
D If Sync is activated, the Arpeggiator will play in sync with the sequencer tempo. By changing the Rate you can
make the Arpeggiator play in different tempo resolutions in relation to the tempo setting.
Straight, dotted or triplet note values are available in 1/2 to 1/16 resolutions. In addition, there are also 1/32, 1/
64 and 1/128 (straight) note resolutions.
D If the “Free” button is activated, the arpeggio rate is free running, and not synced to tempo.
The Rate is then selectable from 0.1 to 250Hz.

Gate Length
This determines the length of the arpeggio notes. Minimum value is 0 (Gate closed - no output). Maximum value is
“Tie”, meaning the gate is open all the time. This parameter can be controlled via CV.

Single Note Repeat

Single Note repeat governs how the arpeggiator behaves when the user plays single keys or monophonic lines.
D When Single Note Repeat is on, a single key will retrigger the gate, meaning the note will repeat.
If the Octave setting is 1 Oct, the note will simply repeat (given the Gate Length setting is not set to “Tie”). If the
Octave setting is set to anything else, the note will repeat according to the Octave, Mode and Insert settings.
D When this is off, single notes will not repeat and RPG-8 will play arpeggios when the user plays more than one
key (chords).

876 RPG-8 ARPEGGIATOR

 Shuffle
Shuffle is a rhythmic feature, that gives the arpeggio a more or less pronounced swing feel. It works by delaying all
sixteenth notes that fall in between the eighth notes.
In the RPG-8 you can switch Shuffle on or off using the corresponding button. However, the amount of shuffle can
be set globally (for all devices that incorporate this feature) with the Shuffle control in the I/O device.

Pattern editor

The Pattern editor allows you to introduce rests for arpeggio steps which can produce more rhythmic results. The
Pattern editor has 16 step buttons at the top, and a main grid display where the arpeggio notes are represented as
horizontal bars for each step in the arpeggio. The pitch of the arpeggio notes are shown on the vertical axis. Notes
within the C-1 to C7 octave range are shown. Notes cannot be edited in the display, they are only a visual represen-
tation of the arpeggio.
D The Pattern editor is activated with the “Pattern” button.
When activated, the Pattern button and the 16 Step buttons light up.
D When you play a chord (or in case you have recorded notes, when you start playback) the arpeggio will play ac-
cording to the current Arpeggiator parameter settings, as normal.
The only difference is that a pattern will be repeated in the display so that all 16 steps play the pattern.

A three note chord with Pattern off...

...and with Pattern on.

D If you click on a step button it goes dark. This means that this step will insert a rest in the arpeggio pattern.
Note that no arpeggio notes are “skipped”. Inserting a rest means that the step will be silent and the next active
step will play the next note in the arpeggio pattern.
D The “Steps -/+” buttons can be used to set the number of steps in the Pattern editor.
E.g. if you press the “Steps minus” button four times the last four step buttons will go dark and the Pattern editor
cycle will start over after step 12.

877 RPG-8 ARPEGGIATOR

 Pattern functions
When the Pattern editor is activated, you will find some specific pattern functions on the Edit menu (and on the de-
vice context menu). These are as follows:

| Function | Description
Alter Pattern The Alter Pattern function modifies existing step patterns. Note that there has to be a pattern to
start with - using the Alter function if all step buttons are active (or inactive) won’t do anything.
Randomize Pattern The Randomize Pattern function creates random patterns.
Invert Pattern This will invert the pattern, i.e. active steps will become rests and vice versa.
Shift Pattern L/R The Shift Pattern functions move the pattern one step to the left or right.

CV connections

On the back of the RPG-8 you can find a number of useful CV connectors. These are as follows:

CV Inputs
There are five CV inputs, of which four can be used to control RPG-8 parameters that have associated controls on
the front panel. These parameters are Gate Length, Velocity, Rate and Octave Shift.
If you use an external source to modulate these parameters, the incoming CV is merged with the setting on the front
of the device.
An example: Velocity is set to 50 on the front panel. A Matrix (bi-polar curve) that varies between +- 20, with the volt-
age trim pot set to 64 (50%) is connected to the Velocity CV input. The resulting Velocity should then vary between
40-60.
In addition to the above CV inputs, there is a “Start of Arpeggio Trig In” connector. This restarts the arpeggio figure
from step 1 when this input receives a gate trigger. See “Triggering arpeggios” for a tip on how this can be used. If
something is connected to this input the RPG-8 will not generate arpeggios unless a Gate trigger is received.
Note: If you are modulating the arpeggio using the CV Input jacks, this will not affect the rendered arpeggio notes.

878 RPG-8 ARPEGGIATOR

 CV Outputs
The following CV outputs are available:
• Gate CV Out (Velocity) transmits a gate/velocity value that corresponds to the Velocity parameter setting.
This is normally auto-routed to the Sequencer Control Gate input on an instrument device.
• Note CV Out transmits the notes generated by the Arpeggiator (or if this is off, the notes you input to the RPG-
8).
This is normally auto-routed to the Sequencer Control CV input on an instrument device.
• Mod Wheel/Pitch Bend Outputs will normally connect to the corresponding inputs of the instrument device
controlled by RPG-8.
• The common MIDI performance controllers Aftertouch/Expression/Breath can be used to control any parame-
ters using these CV outputs.
• Start of Arpeggio Trig Out - every time the arpeggio figures start over this output will send a gate signal.
This can be used to trigger filter, amp and mod envelopes in other devices, e.g. Subtractor. The length of this gate
is the same as the length of the first note in the arpeggio.
• Sustain Pedal Gate Out - the sustain pedal can be used to trigger envelopes in other devices.
If you plug a cable into the “Sustain Pedal Gate Out” the normal pedal to the Hold parameter connection is broken
and the sustain pedal function is changed to send out a gate signal with velocity equalling the Velocity setting
knob (if Manual mode is on then velocity defaults to a value of 100).

879 RPG-8 ARPEGGIATOR

 Tips and tricks
Using the RPG-8 for modulation
You can use the RPG-8 as a modulation source, much like an LFO. The RPG-8 can generate stepped modulation
that is both synchronized to tempo and controlled by note input. In this and in following examples we assume you al-
ready have a RPG-8 connected to an instrument device.
D With the RPG-8 selected in the rack, create a Spider CV Merger/Splitter.
The RPG-8 Note and Gate CV outputs are auto-routed to the Spider Split A and B inputs, and the Spider first split
Note and Gate outputs are connected to the instrument device (as they were before creating the Spider).
You now have 2 (and 1 Inverted) additional Note and Gate CV outputs via the Spider.
D Try using the Note CV split outputs to modulate other parameters in the instrument device, e.g. filter frequency.
The filter frequency will then track the notes generated by the Arpeggiator.
D You can of course also connect the Note CV out to various parameters in any device, not just the device con-
nected to the RPG-8.
D You can connect the Spider split CV/Gate outputs to the corresponding Sequencer Control CV inputs to an-
other instrument device so that the RPG-8 controls two (or more) devices. (This can of course also be done by
connecting the RPG-8 to a Combinator device.)

Triggering arpeggios
On the back panel there is a “Start of Arpeggio Trig In” CV connector. This restarts the arpeggio figure from step 1
when this input receives a gate trigger. You could use this in the following way:
D One or more Redrum channels Gate out can reset the step pattern to create rhythmic patterns in sync.
D You could use the Matrix in the same way - each positive Gate signal will restart the arpeggio figure.
! Note that no arpeggio will be generated unless a Gate trigger is received when something is plugged in to the
“Start of Arpeggio Trig In” CV connector.

Triggering samples
The Gate CV output can be used to trigger samples, either in Redrum or Kong or in the NN-19 or NN-XT Sampler.
D Connect the RPG-8 Gate CV out to the Gate (Sequencer Control) in on the NN-19/NN-XT or to one of the indi-
vidual Gate Channel inputs of Kong or Redrum.
Gate values will now trigger the sample on each step with Gate values above “0”.

Using the RPG-8 as a MIDI to CV converter

You can also use the RPG-8 as a stand-alone MIDI to CV converter without generating arpeggios. In this mode (Ar-
peggiator Off) you can play instrument devices just like as if MIDI input was directly connected to the target device.
The main benefit of this mode is that you can use note pitch and velocity to control parameters, not only in the target
device but in any device.

880 RPG-8 ARPEGGIATOR

Chapter 57
Working with Players
 About this chapter
This chapter describes the procedures and techniques for working with the Players device type.

Overview
A Player is a special type of device that automatically processes, filters and generates MIDI Notes, based on input
MIDI Notes to an Instrument device in the rack. Players can also play back MIDI on their own, without any MIDI input;
this could for example be pattern sequencers.
The Player devices can be found in the Players palette next to Utilities in the Device Palette:

The Players palette in the Device Palette.

The basic idea behind Players is that you first create an Instrument device, then hook up one or more Player devices
to the Instrument device. If the Player device is a chord generator, for example, you can have it generate chords in the
desired key and scale by just playing single notes on your MIDI Control Keyboard. Absolutely great for inspiration!
Player devices are a little narrower than regular devices, and are always created directly above their corresponding In-
strument device(s):

Two Player devices (in series) attached to an ID8 Instrument device in the rack.

882 WORKING WITH PLAYERS

 General recording principle
Apart from sequencer Players, recording Instrument tracks using Players is done as follows:
• By recording the notes exactly as you play them on your MIDI Control Keyboard.
This means that you simply record the notes you play on your MIDI keyboard straight onto the sequencer track,
just as you would a regular MIDI track (without any Player). During playback the recorded notes are then run
through the Player, which generates the Player notes in real-time.
The pictures below illustrates the recording principle:

MIDI Out from Master Keyboard

Patch name Chords

Scales 3 2
OCT UP
2 4 1 3
G Pollyprygan
P ll ADD OCT DOWN
1 5 0 4
KEY SCALE FILTER COLOR
NOTES NOTES INVERSION OPEN CHORDS ALTER

Output Fm7

MIDI data generated by Player

Grand A Delay
ID8 1

Piano Upright
Dance
B
C
Vibes D Chorus instrument device
PITCH MOD VOLUME

Live and playback MIDI data from Instrument Track in sequencer

MIDI In ID8 1

S M

The recording principle.

Using Players
Creating Players
Creating a Player device is very similar to creating other device types, with a few exceptions:
• A Player device can not exist on its own, without an associated instrument device.
• A Player device can only be created for devices in the Instrument palette.
All Instrument types are supported: native devices as well as Rack Extensions.
Here are the different ways you can create a Player device:
D Drag a Player from the Players palette and drop above an Instrument device, to attach it to the Instrument de-
vice.
The insertion point is indicated by the standard orange insertion line with the + symbol:
D Select the Instrument device you want to attach a Player to in the rack, and then double-click the Player device
in the Players palette.
The Player device is attached to the Instrument device.
D Right-click an Instrument device in the rack and select “Create > Players > Player device” from the context
menu.
The Player device is attached to the Instrument device.
• If you only create a Player device, a MIDI Out Device will be automatically created and attached to it.
You could then replace the MIDI Out Device with another instrument device if you like.

883 WORKING WITH PLAYERS

 Chaining Players
If you like, you can attach several Player devices to a single Instrument device in the rack. Just follow the instructions
in “Creating Players” above and make sure you insert the additional Player(s) somewhere in between the Instrument
device and the top section of the topmost Player device.
! When you chain several Player devices the MIDI is processed from the top device and down.
See “Tips & Tricks” for some interesting combinations!

Re-ordering chained Players

You can also rearrange the order of the chained Players:
D Drag the desired Player and drop at the new position.
The insertion point is indicated by the standard orange insertion line with the + symbol.

Replacing Players
Replacing a Player device with another Player device can be done in exactly the same way as when replacing other
device types, see “Replacing devices”.

Deleting Players
Deleting Players is done in exactly the same way as when deleting other device types, see “Deleting devices”.

Naming Players
Naming Players is done in exactly the same way as when naming other device types, see “Naming devices”.

About Players in Combinators

Players can also live inside Combinators, if there are instrument devices there. A Combinator device can also have
Players attached to it (outside of it).

Common Player device parameters

Fold and Bypass All

At the top of the Player device - or above the topmost Player device, if you are using several chained Players - are the
Fold and Bypass All buttons.
D Click the triangular Fold button to fold/unfold all Player devices attached to the Instrument device.
D Switch on Bypass All to send the MIDI from your MIDI Control Keyboard straight to the instrument device, by-
passing any Player devices.

884 WORKING WITH PLAYERS

 On button
D Click to activate or deactivate the Player.
When deactivated (off), the MIDI notes from your MIDI Control Keyboard bypass the Player.

Patch section
Here is where you can browse, load and save your Player patches. The functionality is exactly the same as for other
patch based devices, see “About patches”.

Inputs and Outputs

Player devices can also feature CV In/Out connectors as well as Audio In connectors on the rear panel.

Getting the Player MIDI output onto a track in your DAW

To get the generated MIDI notes onto a track in your DAW you will have to use the MIDI Out Device:
1. Drag a MIDI Out Device from the Instruments palette in the Browser and drop in the rack.
The MIDI Out Device is created in the rack.
2. Click the Players palette and drag e.g. a Dual Arpeggio Player and drop above the MIDI Out Device:

The Dual Arpeggio Player is automatically attached to the MIDI Out Device.
3. Create a MIDI track in the DAW sequencer.
4. Select Reason Rack Plugin as MIDI Input port for that track (refer to the DAW manual).
5. Arm the destination track in your DAW for recording and record the Player MIDI in real-time.

885 WORKING WITH PLAYERS

 Dual Arpeggio

The Dual Arpeggio device accepts one or several notes as input and generates rhythmic patterns based on these
notes. You can use it as a traditional “monophonic” arpeggiator, where arpeggio lines are played back one note at a
time, based on the notes you play/hold on the MIDI Control Keyboard.
You can also use it in Pattern Mode, where you can create polyphonic (up to 4 notes) rhythmic patterns, depending
on how many notes you play/hold on your MIDI Control Keyboard.
You can control the Velocity of the arpeggiated notes from your MIDI Control Keyboard, or by activating the Velocity
switch and drawing in Velocity values in the display.
Dual Arpeggio has two independent arpeggiator sections, that are identical. The sections can either be played by the
same input notes (parallel) or you can set up separate key ranges for them (key split). Input notes that are outside the
set range(s) will be throughput as is, which means you can play melodies on top of arpeggiated chords etc.
The Dual Arpeggio device plays back in sync with the main sequencer, as soon as you input notes to it.
! Please, note that this device is not available in Reason Lite Rack Plugin.

Arp1/Arp2
These are the “on/off” buttons for each of the sections.

Input Range
Here you can define the input note range, to which the arpeggiator should respond. Input notes outside the set
range(s) are bypassed and let through unaffected. This is very useful if you, for example, want to play melodies on top
of arpeggiated chords.
D Set the lowest and highest input note that should be routed to the corresponding arpeggio section, by click-
holding the notes in the displays and dragging up/down.
You can also set the Input Range by using the LRN (Learn) buttons:
D Press a key on your MIDI Control Keyboard and then click the desired LRN button to input the note number for
the pressed key.
Repeat the procedure in the other display for the other note in the range.

886 WORKING WITH PLAYERS

 Rate
D Set the time division to which to play back the arpeggiated notes.
Values: 8 bars, 4 bars, 3 bars, 2 bars, 7/4, 6/4, 5/4, 4/4, 3/4, 2/4, 3/8, 1/4, 3/16, 1/8, 1/8T, 1/16,
1/16 Shuffle*, 1/16T, 1/32, 1/32T, 1/64.
* The Shuffle amount is set with the Shuffle knob in the I/O device.

Octave
D Set how many octaves the arpeggiated notes should cover.
Range: 1-4 octaves.
Provided the Pattern function (see “Pattern”) is not active, this is what will be played back:

| Octave range | Description

1 Oct The arpeggiated notes will be those that you press down on the keyboard.
2 Oct If you select this, the arpeggio will be extended to a two octave range, i.e, the arpeggio pattern is played then re-
peated one octave up. In Random mode, the notes you input are played in random order over 2 octaves.
3 Oct Same as 2 Oct but extended to a three octave range.
4 Oct Same as 2 Oct but extended to a four octave range.

Repeat
D Activate this to repeat each note in the generated arpeggio.
Since each note is played back twice at the set Rate, the total length of the arpeggio will be twice as long.

Direction
D Click-hold in the display and drag up/down to select the direction of the arpeggio/pattern notes.
Values: Up, Down, Up+Down and Random.
Provided the Pattern function (see “Pattern”) is not active, this is how it works:

| Mode | Description
Up This will generate an arpeggio that plays from the lowest note to the highest note.
Down Notes are played from the highest note to the lowest note.
Up+Down Notes are played from lowest note to highest, then from highest back down to the lowest. The very lowest
and the highest arpeggio notes are not repeated. I.e. the arpeggiator will play the lowest note to the sec-
ond highest note, then the highest note to the second lowest note.
Random The notes you input will be arpeggiated randomly.

If the Pattern function is active (see “Pattern”), the Direction setting only determines in which direction the steps
should advance - regardless of the pitches of the notes in the pattern.

Hold
D Click the Hold button to keep the arpeggio running even after you release all keys.
It will continue to arpeggiate the last notes played until a new note-on is received.
• If you continue to hold down at least one key when Hold is on, any new notes will be added to the existing ar-
peggio as opposed to starting a new arpeggio.

887 WORKING WITH PLAYERS

 The Display sections

In the display sections you can design your arpeggios/patterns.

Normal (Arpeggio) mode

If none of the switches to the left of the display (Steps, Pattern or Velocity) are activated, the display only shows the
orange moving step indicator (if you hold down more than one key on your MIDI Control Keyboard). The rest of the
display area is dark.
• The step indicator travels the same number of steps as the number of notes you play and hold on your MIDI
Control Keyboard.
So, if you hold down a 5-note chord, the step indicator advances five steps before it starts over again.

Steps
If you want the arpeggio to always play a certain number of steps, regardless of how many keys you hold down, you
can activate the Steps function. This is perfect for maintaining a steady “beat” in your song.
1. Click the Steps switch to activate the Steps function.
The default value is 4 steps.
2. Click or click-hold and drag on the right side of the orange bar on the area above the step indicator, to increase
or decrease the number of steps:

The arpeggio (or Pattern, see below) will restart after the set number of steps, regardless of how many notes you
are playing.
Range: 1-16 steps.

888 WORKING WITH PLAYERS

 Pattern
In the default “Arpeggio mode” (Pattern switch off), the arpeggiator(s) work in the standard “monophonic” fashion,
where the notes are played back one by one according to the settings you have made in the left section of the device.
With Pattern activated, the arpeggiator turns into a combined pattern player + arpeggiator. Pattern mode allows for
up to four-note polyphonic “arpeggios”.
Here is how it works:
1. Click the Pattern switch to activate the Pattern function.

All currently active pattern steps (green boxes) light up in the display.
The four green boxes in the example above represent the playback pattern if you hold a 4-note chord on your MIDI
Control Keyboard. The lowest row in the display represents the lowest held note in the chord you play.
2. Set the Direction to “Up” (see “Direction”), to make the examples below work as described.
If you, for example, hold down the keys C4, E4, G4 and B4, the pattern plays back C4 in step 1, E4 in step 2, G4
in step 3 and B4 in step 4. Then the pattern starts over again:

Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4 Step 5
If you hold down fewer keys, the pattern steps gets equally fewer steps and only plays back the new lowest to the
new highest notes in the chord:

Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4 Step 5

889 WORKING WITH PLAYERS

 3. Click in any of the matrix boxes to edit the pattern:

Editing the pattern.

• Each step can have a note that can be 1/3, 2/3 or 1/1 of the step’s total length, depending on where in the box
you click.
This way you could draw in “staccato” notes for desired steps, if you like. If you click the full 1/1 of a step (click in
the right area of the box), and then click another consecutive box on the same row, the notes will be tied together
(played in legato).
If you hold down the keys C4, E4, G4 and B4, the pattern above now plays back C4 and B4 in step 1, E4 and B4
in step 2, G4 in step 3 and B4 in step 4. Then the pattern starts over again:

Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4 Step 5

D Click anywhere on a filled box to delete the note.

D To fill several consecutive boxes on the same row with identical note lengths, click and hold on the desired po-
sition in the initial box and then drag to the right:

Adding consecutive notes by click-holding and dragging.

Here are some general rules regarding Pattern Mode:
• A pattern can be up to 16 steps long (if Steps is activated, see “Steps”).
• The notes you play are then mapped to the four note positions for each step.
If you play a four note chord, some steps may play only the lowest note, others may play a higher note or a combi-
nation of notes or all four.
• If you hold down more than four notes, note 5 and up are ignored completely in the pattern.
Note that if the Steps function is not active (see “Steps”), note 5 and up will add additional silent steps in the pat-
tern, making the pattern longer.

890 WORKING WITH PLAYERS

 • If you play fewer notes than the number of rows you have programmed in you pattern, this is what happens:
Let’s take the pattern below (with the Steps function off) as an example:

Pattern Mode with the Steps function off.

If you only hold down key C4, note C4 is repeated (since the pattern length is only one step with one key pressed):
Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4 Step 5
If you hold down C4 and E4, the pattern plays back C4 and E4 alternating (since the pattern length is now two
steps with two keys held):
Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4 Step 5
If you hold down C4, E4, and G4 the pattern plays back C4, E4 and G4 alternating (since the pattern length is now
three steps with three keys held):
Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4 Step 5

891 WORKING WITH PLAYERS

 • If we now do the same procedure with the Steps function on - and set the number of steps to 4 - things work a
little differently.
This is what happens:

Pattern Mode with the Steps function on.

If you only hold down key C4, note C4 is played back on step 1 and note C5 is played back on step 2. Steps 3 and
4 are silent. In other words, a new note one octave above is automatically generated on the second step:
Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4
H ld Pl d b k
If you hold down C4 and E4, the pattern plays back note C4 on step 1, note E4 on step 2, note C5 on step 3 and
note E5 on step 4. So, the two held notes are played back one octave higher on steps 3 and 4:
Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4
If you hold down C4, E4, and G4 the pattern plays back C4 on step 1, E4 on step 2, G4 on step 3 and C5 on step
4. As you can see, the lowest held note is played back one octave higher on step 4:
Held notes: Played back notes:

C4
Step 1 Step 2 Step 3 Step 4

Velocity
In default mode, when the Velocity function is off, the velocities of the input notes set the velocity of each generated
note in the arpeggio/pattern. If you want to define the velocity values and draw them in manually, you can activate the
Velocity function.
1. Click the Velocity switch:

2. Click or click-hold and drag the blue Velocity bars to manually set the velocity values.
When you play your chords, the input velocity is disregarded and replaced by the values you drew in the display.
Range: 0-127.

892 WORKING WITH PLAYERS

 Shift Step

D Change where in the arpeggio/pattern you want the playback to begin.

Range: -16 to + 16 steps.

Transpose

D Transpose all the generated arpeggio/pattern notes equally, in semitone steps.

Range: -24 to + 24 semitones.

Gate Length

D Change the lengths of the generated arpeggio/pattern notes.

Here you can change the “gate/hold” time for the generated arpeggio/pattern notes. 100% means the gate time
is exactly one full Step. A Length of 25% results in a gate time of 1/4 of a Step and a Length of 400% means a
gate time of four Steps.
Range: 25%-400%.

893 WORKING WITH PLAYERS

 Note Echo

Note Echo simulates a MIDI delay effect, by adding repeats of the incoming MIDI notes. The Note Echo device is
polyphonic, so you can get repeats of entire chords if you like.
! Please, note that this device is not available in Reason Lite Rack Plugin.

Step Length
The Step Length knob controls the echo time, and consequently also the time between the repeats. The step length
range is 0 to 1000 milliseconds. When Tempo Sync (see below) is enabled, the range is 1/128 notes up to ½ note.
q A Step Length of 0 makes it possible to create chords and clusters, since all steps are played back simultane-
ously. See “Creating parallel chords” for a practical example.

Tempo Sync
Switch on to sync the Step Length (see above) to the main sequencer tempo.

Repeats
This knob sets the number of repeats, from 1-16. The number of repeats is also shown in the display.

Velocity
The Velocity knob controls the linear increase or decrease of the Note Velocity value for each step.
The range is from 10% to 200%, where 100% means the velocity value is left unchanged throughout the repeats.
Values below 100% means a decrease of the velocity values, and values above 100% means increasing velocity with
each repeat.
The velocity values are represented by blue bars in the display. The blue line in the bars indicates the 100% value.

Pitch
With the Pitch knob you set how many semitones each step should be transposed relative to the previous step. The
function is linear for all repeats. The Pitch values are represented by orange bars in the display.
The range is –12 to +12 semitones. 0 is the default value and means no transposition of the input notes.
! If you use very high or very low Pitch value settings together with many Repeats, the note range of the associ-
ated instrument might be exceeded. If this happens, any remaining repeats will be silent.

Display
Aside from showing number of Repeats, Velocity and Pitch (transposition) information, the display can also be used
if you want to mute individual repeats.
D Click the desired green circles in the display, to mute the corresponding repeat.
This way you can create interesting rhythmic repeats. The first green circle (to the left in the display) represents
the incoming “dry” MIDI Note. This can be muted as well, if you like. By default, all repeats are on (unmuted).

894 WORKING WITH PLAYERS

 Scales & Chords

The Scales & Chords device either transposes incoming notes to fit a set scale, or transposes notes and generates
chords that fit the desired scale. Scales & Chords has a number of built-in preset scales, and you can also create your
own custom scales!
Here are some examples of how you can use it:
• to get chords from single note inputs.
• to get new chord voicings and progressions that feel fresh and inspires your music making.
• to experiment with existing note lines, by transforming them to different scales.
• to get the notes you play to automatically fit the music.
• to easily be able to play chords without using a MIDI Control Keyboard.
You could write single MIDI notes in a Note clip and have the Scales & Chords Player generate the chords for you.
• to get chords that sound right, without knowledge of tonality/scales.

Scales
In the Scales section you choose which key and scale you want.
q Note that all parameters in the Scales section can be automated, so you could change key and scale anywhere
throughout your song if you like!

Key
D Click to select the desired Key from the list that appears.
12 Keys are available, from C to B.

Scale
D Click to select the desired Scale from the list that appears.
13 scales are available, plus a Custom scale that you can create yourself (see below). The preset Scales are:
Major, Minor, Lydian, Mixolydian, Spanish, Dorian, Phrygian, Harmonic Minor, Melodic Minor, Major Pentatonic,
Minor Pentatonic, Hemi Pentatonic and Chromatic.

895 WORKING WITH PLAYERS

 • The notes that fit the selected Key and Scale are displayed in light blue in the keyboard display in the Scales
section:

The Dorian scale in the key of E.

The notes in the selected Key and Scale are then automatically repeated throughout all octaves of the entire MIDI
Note range.
• Notes you play on your MIDI Control Keyboard/On-screen Piano Keys are indicated in orange in the Scales
keyboard display.

Creating a Custom Scale

D Create a custom Scale by clicking the desired keys in the keyboard display in the Scale section:

Selected notes are colored light blue. The selected notes are then automatically repeated throughout all octaves
of the entire MIDI Note range.
• You can easily switch between your Custom Scale and any of the preset Scales whenever you want, by clicking
the “Scale” display and selecting the desired scale from the list that appears.
• Your Custom Scale is automatically saved with the song.
Your Custom Scale is also saved in the Scales & Chords patch, should you choose to save your settings as a
patch.

Filter Notes
• When the Filter Notes switch is off (default), incoming notes outside the set Scale (see “Scales”) are trans-
posed to fit the scale.
Wrong notes are automatically transposed to the closest correct note. If the wrong note is equally near two correct
notes, the wrong note is transposed to the lowest of the correct notes.
• When the Filter Notes switch is on, incoming notes that are outside the set scale will be filtered out (silent) in-
stead of transposed.
The display below shows a hand symbol whenever wrong input notes are filtered out (silent).

896 WORKING WITH PLAYERS

 Chords
D Switch on the Chords function to generate chords out of single notes.
When you play a note, you will get a chord that fits the set Scale (see “Scales”), and contains the number of notes
set with the “Notes” knob. The generated notes are shown in real-time in the keyboard display at the bottom of the
Chords section:

! If you play several notes at the same time you will get several chords, which might be cool or just weird. If
notes in the generated chords should coincide, only one instance of the notes are played back (to avoid dou-
bled/layered notes).
q If you want to create “parallel chords”, i.e. chords that contain notes with the same relative intervals, the Note
Echo device is better suited for this, see “Creating parallel chords”.

Notes
D Set the desired number of notes in the generated chords.
Range: 1-5 notes.

897 WORKING WITH PLAYERS

 Inversion
The Inversion knob defines which note in the generated chord should be the lowest. The function tries to keep the
lowest generated note in the chord as close to the played root note as possible. Below is an example of inversions of
a generated 4-note chord in the Scale C Major, played from a C note of the MIDI Control Keyboard:

Inversion 0:

C C C

Inversion 1:

C C C

Inversion 2:

C C C

Inversion 3:

C C C

Inversions of a generated 4-note chord played from the (orange) C note.

! Note that the Inversion function is only relevant if the Inversion number is less than the selected “Notes” value.
So, if the Notes value is 4, only a maximum Inversion value of 3 is useful.

Open Chords
When the Open Chords function is active, some notes in the generated chords are transposed in octave steps, so
that the notes are not so close to each other. The transposition is different depending on the “Inversion” and the
“Notes” settings.

Add
The Add functions can be used for adding notes to the generated chords:
• The Add Oct Up function adds a note one octave above the root note.
• The Add Oct Down function adds a note one octave below the root note.

898 WORKING WITH PLAYERS

 • The Add Color function adds a note two thirds above the highest note in the tertian chord structure.
If the Notes value in the Chord section is 3, you will get a regular triad (e.g. a C major). Adding a sixth above top will
mean adding a 9 (e.g. the note D). Similarly, if the Notes value in the Chord section is 4, this will add an 11; if the
Notes value in the Chord section is 5, this will add a 13.
q All three Add functions can be used together, simultaneously.
! If the added note is already used in the chord (due to inversions, open chord etc), the note is transposed up an-
other octave (or down, in case of Oct Down).

Alter
The Alter button is a momentary button that turns off as soon as you release the mouse button (like the Trig button
on The Echo, for example). It alters the chord slightly so that it goes outside the current scale. The result depends on
the Key (as set in the Scale section), whether the current scale is major or minor and what note you play (relative to
the Scale Key).
The chord is altered by having one of the chord notes changed. For example, a major chord may be changed to a mi-
nor chord, or vice versa.

899 WORKING WITH PLAYERS

 Beat Map

The Beat Map Algorhythmic Drummer is a Player that generates drum patterns based on built-in beats, algorithms
and simple but powerful controls. It's normally used with a drum instrument such as Kong, Rytmik or Umpf, but can
also be used with melodic instruments for interesting results.
! Please, note that this device is not available in Reason Intro Rack Plugin or in Reason Lite Rack Plugin.

Included content
To browse the content of Beat Map, click the Browse Patch button on the BeatMap panel to open the Browser,
where all patches are listed.
Beat Map comes with a number of patches, both for the Player itself and for several different drum devices. The drum
device patches are tailored to work well with Beat Map, but you can of course use any patch from the Factory Sound
Bank, or make your own.
If you want immediate results, start by loading one of the patches in the Combinator Style Patches folder! These
combine Beat Map with drum devices and effects for instant rhythmic goodness.
If you want a taste of what can be done with multiple Beat Maps in combination with other devices, check out the
Demos and Song Starters folder.
! Note that some of the combi patches use Rack Extensions included with Reason+.

900 WORKING WITH PLAYERS

 The front panel

The central item on Beat Map's panel is the Map display. This is where you select a basic rhythm, by clicking and
dragging the cross-hair.
• The cross-hair position is also indicated with the X and Y position parameters, which you can edit directly if
you like.
Each integer XY position corresponds to a different beat. The map graphics are there for visual reference and for the
nice looks – there is no particular rhythmic significance to the elements on the map. However, as a general rule, mov-
ing the position to the right will make the bass drum pattern more syncopated and moving the position upwards will
make the snare pattern more syncopated.
• Clicking the Spawn XY button drops you at a random position:

901 WORKING WITH PLAYERS

 Map Select
1. Click the Map Select button to show the four included maps:

They contain beats in different styles – click on a map to see some info about it in the display.
2. To select a map for use, either double click it or select it and click Map Select again.

Density

The beats consists of four different rhythmic patterns, for kick, snare, hi-hat and percussion. Once you've found a
beat that you like, you can adjust the Density of each drum. This means reducing or increasing the number of notes.
Turning Density down to zero will mute that drum. The Density parameters can be automated or CV controlled, for
continuous variations.
• Clicking the Random Density button to the left will change all four Density controls to random values, useful
for live variations:

902 WORKING WITH PLAYERS

 Lock Position

You may find a beat where for example the kick is perfect, but not the other drums. Then you can click Lock Pos for
the kick. This will put a "pin" on the map – the kick beat will stay at this position but you can continue exploring the
map for the other instruments.
This can be done independently for all four drums, locking them to different map positions. This way you can combine
rhythms from different basic beats, for infinite variations.
! You can still adjust Density for locked drums.

Mirror notes

Each of the four drums has a "Mirror" function, which plays a note in between two note positions from it's main drum.
For example, if the kick beat plays like this:

... then its Mirror notes (dark red) will play like this:

This means that the Mirror notes will be affected by the Density of the main drum, creating interesting counter-
rhythms or ghost notes.
Use the Mirror knobs to set the velocity (typically level) of the Mirror notes. Turning the knob down completely will
turn off the Mirror note.

903 WORKING WITH PLAYERS

 Setting MIDI note numbers
Each drum and Mirror function sends out notes on a separate MIDI note number, meaning that Beat Map can control
eight different drum sounds.
• Clicking the red signal LED for a drum will send out that note, for quick auditioning of drum sounds:

The Perc drum sends out MIDI Note F#1

• You can set which note a drum should send by clicking and dragging the note number on the panel:

Alternatively, use the Learn function:

1. Click Learn Key:

2. Click the note number value for a drum or Mirror to highlight it.

3. Play the desired note on your MIDI controller.

4. Repeat for the other notes.
5. Click the Learn Key again to exit Learn mode.
This is required for the changes to stick when you save the patch or song.

904 WORKING WITH PLAYERS

 Global settings

To the right are three settings that affect the overall playback:
• Rate is the note value of each step. Normally this is set to 1/16th notes.
• Shuffle allows you to add a swing feel to the beats by delaying the offbeat 1/16th notes.
You can set it manually from 50-75% or select “Global” to use the Global Shuffle setting (found in the I/O device
in Reason Rack Plugin).
• Reset Step determines the number of steps before the beat starts over from the beginning.
The default is 64, but it can be useful to lower this to make a beat less varying or to impose some structure on a
chaotic rhythm.

Beat Map and the main sequencer

You can click the Run button manually for auditioning the beat, but the Run function is normally controlled by the Play
function in the main sequencer.
However, if you want Beat Map to play in certain parts of your song but not in others, it's possible to control this by
automating the Beat Map “On” button in your host DAW:

In addition, most Beat Map parameters can be automated. For example, you can record map movements or Density
changes.
A couple of special features:
• The Locked XY positions can be automated for each drum.
This allows you to lock a single drum to a rhythm but change it with automation:

• The Map selection ("Beat Map" parameter) can be automated or controlled from a Combinator.
Switching to another map can be useful as a break or fill.

905 WORKING WITH PLAYERS

 Editing the drum notes
To manually edit the beats from Beat Map you need to connect Beat Map to a MIDI Out Device and record the MIDI
output from Reason Rack Plugin to another track in your DAW, see “Getting the Player MIDI output onto a track in
your DAW”.

Using CV

On the back of Beat Map, you'll find a comprehensive selection of CV inputs and outputs.
• Density CV in and Mirror Velocity CV in for each drum.
Using these with a Pulsar CV LFO can create longer, varying rhythmic patterns or random variations. There are
plenty of examples of this among the Combinator Style Patches.
• Gate Out from main drums and Mirror sections, allowing you to trigger other sounds or effects in parallel.
• Reset Step Trig In.
Whenever this receives a positive CV value, the beat will start over from the first step.
• Map XY CV Inputs, for automatically modulating the map position during playback.
• Map XY CV Outputs.
These send out the X/Y coordinates as CV values, letting you use the Map display as an XY controller for other
devices in Reason.

906 WORKING WITH PLAYERS

 Tips & Tricks
Generating scale-correct arpeggios from single notes
By chaining a Scales & Chords device with a Dual Arpeggio device you will be able to generate scale-correct arpeg-
gios by playing single notes.
1. Create an Instrument device.
In this example we create an ID8.
2. Select the ID8 device in the rack and double-click a Scales & Chords device.
The Scales and Chords device is automatically attached to the ID8.
3. Select the ID8 device in the rack and double-click a Dual Arpeggio device.
The Dual Arpeggio device is attached between the Scales & Chords device and the ID8 device:
4. On the Scales & Chords device, select Key and Scale, then switch on the Chords function set the Notes knob to
4.
5. Play single notes on your MIDI Control Keyboard and hear how four-note scale-correct arpeggios are now gen-
erated.

Generating chord arpeggios

By chaining a Dual Arpeggio device with a Scales & Chords device you will be able to generate arpeggiated chords.
1. Create an Instrument device.
In this example we create an ID8.
2. Select the ID8 device in the rack and double-click a Dual Arpeggio device.
The Dual Arpeggio device is automatically attached to the ID8.
3. Select the ID8 device in the rack and double-click a Scales & Chords device.
The Scales and Chords device is attached between the Dual Arpeggio device and the ID8 device:
4. On the Scales & Chords device, select Key and Scale, then switch on the Chords function set the Notes knob to
3.
5. Play chords on your MIDI Control Keyboard and hear how arpeggiated three-note chords are now generated.

Creating parallel chords

The Scales & Chords Player is great for generating chords that stay within a set scale, but sometimes you may want
chords that keep their form when you play different notes. For example, you might want a minor 9th chord that is
transposed with the notes you play, but stays a minor 9th chords (like with the Chord Memory function of some syn-
thesizers, for example). This can be called “parallel chords”, and is actually best done with the Note Echo device:
1. Set Step Length to 0.
2. Raise the Repeats value to maximum.
3. Set the Pitch to +1 semitone.
This means that each repeat is 1 semitone higher than the previous - and since Step Length is 0, they are all
played at the same time. If you play a note now, you get a dense cluster of notes - not very musical or useful, but
let’s continue:

907 WORKING WITH PLAYERS

 4. Click the Repeat circles in the display to turn off the notes you don't want.
This is how you would set up a minor 9th chord for example:

5. Play different (single) notes on your MIDI Control Keyboard/On-screen Piano Keys to get parallel, transposed
chords.
You could also add a Dual Arpeggio Player after the Note Echo for cool arpeggios!
q The Note Echo device comes with a number of ready-made chord patches. You find them by clicking the
Browse Patch button on the Note Echo device and opening the “Chords” sub-folder in the browser.

Using a Scales & Chords device as a “MIDI Note monitor”

Sometimes it might be interesting to see in real-time what notes/chords a Player or “Player chain” actually generates.
1. Add a Chords & Scales device last (at the bottom) in the Player chain.
2. Make sure Scale is set to “Chromatic” and that Filter Notes and Chords are Off.
When you play through your Player(s) the generated notes/chords are now displayed in the Scales & Chords key-
board display:

The “Creating parallel chords” example monitored in a Scales & Chords device when playing the note C.

908 WORKING WITH PLAYERS

Chapter 58
Settings
 The Reason Rack Plugin Settings dialog
D Click Settings at the top right of the Reason Rack Plugin window to open the Settings dialog:

Browsing
Open with Browser shown
Check this item to always show the Browser when creating a Reason Rack Plugin instance in your DAW.

Load default sound in new devices

When this is activated and you create a new device (of a device type that supports patches), a default patch will be
loaded. This way you can be sure that a new device generates sound right away. The default sound will also deter-
mine the default location in the Factory Sound Bank when you browse for patches for the new device.
If you turn this off, new devices will be initialized - parameters are reset to their default values and no samples are
loaded in sample-based devices.
! Note that “Sound” also implies patches for effect devices such as the RV7000 and Scream 4.

Do not index these folders

Here you can exclude folders that you have in the Locations and Favorites section of the Reason Browser from scan-
ning and indexing. This is especially useful if you have cloud (e.g Dropbox) folders on your computer.
D Click the Add button to add folders to be excluded from the indexing.
D Select a folder you want to remove in the list and then click the Remove button.

910 SETTINGS
 User Interface
Show parameter value tool tip
Normally, if you hold the mouse pointer over a parameter on a device panel for a moment, a Tool Tip appears display-
ing the name and the current value of the parameter. If you uncheck this option, Tool Tips will not be displayed.

Hide cables function

The selected alternative determines how cables on the back of the rack should be displayed when “Hide Cables” is
selected on the top menu in the Reason Rack Plugin window:
D Select “Hides auto-routed cables” to only display manually routed cables.
All devices with auto-routed cables will have “transparent” cables to make it easier to distinguish the manually
routed cables.
D Select “Shows cables for selected devices only” to only display the cables for selected devices.
All other (non-selected) devices will have “transparent” cables to make it easier to distinguish the cables of se-
lected devices.
D Select “Hide all cables” to hide all auto-routed and manually routed cables.
All cable connections on devices will be indicated with colored dots in the jacks, and no cables will be displayed.
See “Hiding cables” for more details.

Mouse Knob Range

This lets you adjust the response sensitivity of the various knobs in Reason when manipulating them with the mouse.
A higher sensitivity gives a higher degree of precision. You can choose between Normal, Precise and Very Precise.

Theme
Here you can choose from two visual themes (Light or Dark), i.e. how the user interface should be visually presented.
The selected theme affects the Browser areas. See “About different Themes”.
! Note that you have to restart Reason Rack Plugin for the new Theme to take effect.

Reason Rack Plugin info

Online Help
Clicking this button opens up the on-line Help system in your default web browser. The Reason Rack Plugin Help
contains detailed information about all functions in Reason Rack Plugin. You can choose to browse for information,
either from the Table of Contents (TOC), Index or Search tabs in the Help system.

Download Manuals
Clicking this takes you to the support section on Reason Studios, where you can download manuals in pdf format.

About
Clicking this opens up a dialog that informs you about the Reason Rack Plugin version and the people behind it.

911 SETTINGS
 Rack Extensions and Content
In this section you can manage your Rack Extensions and your User Account.

Open Companion
Clicking this button opens the Reason+ Companion application, where you can install Rack Extensions, download
Sound Packs etc.

Open Companion on Startup

Check this box to automatically open the Reason Companion application when you launch Reason Rack Plugin.

Go to Shop
Clicking this button starts up your default web browser and takes you to the Reason Studios Shop, where you can get
more Rack Extension devices for your Reason Rack Plugin rack and also purchase additional ReFills.

Audio
Render audio using host buffer size setting
The “Render audio using host buffer size setting” function should be selected (checked) for best plugin performance.
When selected, the audio batches are rendered internally in Reason Rack Plugin according to the set buffer size of
the host.
For example, if you are using a buffer size of 512 samples, each audio batch will be 512 samples also in Reason
Rack Plug. Raising the buffer size will let Reason Rack Plugin process larger audio batches in one go, which is often
more efficient.
If unchecked (off), all audio batches are rendered internally in Reason Rack Plugin at a fixed buffer size of 64 sam-
ples - regardless of the audio buffer size setting in your host DAW. This might be desirable if you are using feedback
signal routings and CV connections in your songs, and want the internal latency of those connections to be fixed at
a short value all the time. This might result in performance problems, though.

Account and Authorization

Here you can see the current status of your Reason Rack Plugin authorization. For example, here is displayed if you
are running Reason Rack Plugin with long-term authorization, and which Reason Studios account you are using.
Please refer to “About the Authorization system” in the Introduction chapter for more details regarding authorization.

Account page
Clicking this button launches your default web browser and brings you to your products page on the Reason Studios
web site. Here you can view your list of owned products, including Rack Extensions and download ReFills.

912 SETTINGS
Index
 A Channel EQ 611
About 911 Chorus 783
Account and Authorization 912 Chunk Trig (Kong) 252
Account page 912 Combi Patches 802
Add Long-Term Authorization 28 Combinator 801
ADSR 518 Adding Devices 807
All Locations (Location in Browser) 93 Bypass FX 815
All Notes Off (MIDI Out Device) 301 Combining Devices 804
Alligator 570 Creating 804
Dry Ducking 578 Creating by Browsing Patches 804
Effects 576 Creating New Devices 807
Filters 574 CV Connections 826
Mix Controls 578 External Routing 806
Patterns 572 Key 817
Alt Group (Dr. Octo Rex) 114 Key Mapping 817
Alter Run 815
Pattern (Redrum) 479 Select Backdrop 811
Pattern (RPG-8) 878 Setting Negative Parameter Ranges 824
Alter Pattern (Matrix) 848 Setting the Source Range 824
Audio In (to Reason Rack Plugin) 45 Setting Velocity Ranges for Devices 819
Audio Out (from Reason Rack Plugin) 45 Uncombining Devices 809
Audio Settings 46 Using Modulation Routing 820
Audio Signal Connectors 76 Using the Editor 817
Audiomatic Retro Transformer 584 Comp-01 Compressor 799
Authorizing Reason Rack Plugin 26 Compatible EQ (Mixer 14-2) 854
Automap Samples 408 Compressor (Kong) 264
Automap Zones 376 Computer (Location in Browser) 95
Auto-routing 78 Context Menus 39
For Devices 39
B For Parameters 39
Beat Map (Player) 900 For the Rack 39
Bipolar Curves (Matrix) 845 Convolution Algorithm (RV7000 Mk II) 697
Blend (Pulveriser) 658 Copy Loop to Track (Dr Octo Rex) 118
Browser Copy Patch 85
Browse Focus 92 Copy Pattern 842
Open and Close 86 Copy Pattern to Track (Redrum) 480, 848
Using 86 Copy Zones 366
Bus FX Parameter 1 & 2 262 Correction Speed (Neptune) 639
Bypass/On/Off Switch (on Effect Devices) 780 Create Instrument 91
C Cross-browsing
Cables Samples and REX files 91
Automatic Routing 78 Cross-Browsing Patches 89
Checking Connections 78 Curve (Matrix) 845
Manual Routing 77 About 840
Scroll to Connected Device 78 Cut Pattern 842
Catch Zone Size (Neptune) 638 CV
Categories (Editing in Browser) 99 About 76
Categories Buttons (in Browser) 97 Merging 787
CF-101 Chorus/Flanger 783 Splitting 790
Channel 8 & 9 Exclusive 484 vs. Gate 845
Channel Dynamics 605

914 INDEX
 D Effects 68
D-11 Foldback Distortion 793 Enable Loop Playback (Dr. Octo Rex) 117
DDL-1 Delay Line 782 Enable Pattern Section (Redrum) 479
Decay/Gate Switch (Redrum) 482 Envelope Controlled Filter 794
Delay 782 Envelopes 518
The Echo 770 EQ
Delete Unused Samples 406 Mixer 14-2 853
Device Palette 67 Parametric 800
Searching and Filtering 68 RV7000 Mk II Advanced Reverb 700
Devices Europa Shapeshifting Synthesizer 127
Creating 66 Ext Mod (Subtractor) 526
By Browsing Patches 88 External Effect (Kong) 271
Cut, Copy, Paste and Duplicate 71 F
Deleting 71
Favorite Lists (Location in Browser) 94
Folding/Unfolding 37
FFT (Vocoder) 590
Moving 70
Filter
Naming 71 Dr. OctoRex 121
Replacing 71
Effect Device 794
Re-routing 70
Malström 281
Routing 78
NN-19 410
Selecting 69
NN-XT 390
Diffusion (The Echo) 773 Pulveriser 655
Dirt (Pulveriser) 654 Subtractor 513
Distortion Filter (Kong) 265
D-11 793
Flam (Redrum) 478
Scream 4 704 Flanger 783
Do not index these folders 910 Flip Rack 74
Download Manuals 911 FM 511
Dr. Octo Rex Folding/Unfolding Device Panels 37
About 108
Follower (Pulveriser) 657
Adding REX Loops 111 Formant (Neptune) 643
Editing Slices 113
Panel Parameters 118 G
Playing REX Loop Slices 112 Gain Tool 829
Playing REX Loops 109 Gate
Drum Assignment (Kong) 237 About 76
Drum Output (Kong) 242 Programming in Matrix 844
Drum Room Reverb (Kong) 263 vs. CV 845
Dual Arpeggio (Player) 886 Gate (Alligator) 570
Ducking Gate (Matrix) 840
Alligator 578 Gate (RV7000 Mk II Advanced Reverb) 701
The Echo 776 Gate mode (Redrum) 482
Dynamics 477 Go to Shop 912
Grain Sample Manipulator 163
E Groups
ECF-42 Envelope Filter 794 NN-XT 366
Editing
Buttons 36 H
Buttons (Multi Mode) 37 Hide Cables 75
Display Values 38 Hide Cables Function 911
Faders and Sliders 36
Knobs 36

915 INDEX
 High Quality Interpolation Line Mixer 6-2
NN-XT 353 About 836
Redrum 485 AUX Send and Return 837
Hit Type (Kong) 238 Channel Parameters 836
Humana Vocal Ensemble 193 Connections 837
Live Mode (Neptune) 635
I Load Default Sound in New Devices 910
I/O Device 45, 103
Locations (in Browser) 93
ID8 Instrument Device 212
Loop Trig (Kong) 252
Controlling Sounds 213
Loops
Selecting Sounds 213 In Drum Samples 483
Improved EQ (Mixer 14-2) 854
In Samples 407
Impulse Response Samples
Low BW 416
Loading 698
Low BW (Dr. OctoRex) 125
Init Patch
Low Freq (Neptune) 635, 645
Kong 236
Init patch M
Redrum 473 Malström
Subtractor 504 About 274, 632
Input Meter (Effect Devices) 780 Filters 281
Instruments Graintables 277
Creating 91 Modulators 279
Interpolation Oscillators 276
Dr. OctoRex 125 Routing 287
Invert Routing external audio to 297
Pattern (RPG-8) 878 Shaper 285
Master Bus Compresssor 617
K Master FX Parameter 1 & 2 262
Keep Pitch (The Echo) 772 Matrix
Key Commands About 840
Syntax in Manual 24
Application Examples 849
Key Maps Programming 841
NN-19 405
MClass Compressor 627
NN-XT 363
MClass Effects
Key Zones About 624
NN-19 404
MClass Equalizer 625
NN-XT 349
MClass Maximizer 630
Keys (Matrix) 840
MClass Stereo Imager 626
Kit Patches 234
MIDI
Kit Patches (Kong) 234 Send All Notes Off 301
Klang Tuned Percussion 215
MIDI Out Device 300
Knobs 36
Setting up 300
Kong 234
Mimic Creative Sampler 303
Copying and Pasting Drums 238
Missing Sounds 101
MIDI Note Assignment 235
Mixer 14-2
L About 852
Lag (Pulveriser) 656 Auxiliary Return Section 855
LFO Sync Chaining 857
Dr. OctoRex 124 Channel Strip 852
NN-19 413 Channel Strip Controls 853
Subtractor 522 Connections 855
Signal Flow 854

916 INDEX
 Monotone Bass Synthesizer 333 Paste Pattern 842
Mouse Knob Range 36, 911 Paste Zones 366
Mute Patch Cables 77, 78
Mixer 836 Patches
Mixer 14-2 853 About 82
Redrum 481 Alligator 570
Browsing for 89
N Copy and Paste 85
Neptune
Cross-Browsing 89
Mixer Section 647
Initializing 85
Panel Parameters 644
Kong 234
Pitch Display 634
Loading 82
Setting up 634
Missing Sounds 84
NN-19
NN-XT 350
About 400
Pulveriser 652
Loading Samples 402
Redrum 472
Parameters 409
RV7000 Mk II 690
NN-Nano Sampler (Kong) 248
Saving 85
NN-XT
Scream 4 704
About 348
Searching for 96
Group Parameters 384
Subtractor 504
Groups 366
The Echo 770
Loading Samples 350
Pattern Shuffle 478
Main Panel 352
Patterns
Remote Editor Panel 355
Alligator 572
Sample Parameters 382
Cut, Copy and Paste 842
Synth Parameters 386
Muting in Matrix 842
Velocity Ranges 377
Muting in Redrum 478
Noise Generator (Kong) 260
Redrum 474
Note Echo (Player) 894
Running 474
Note To Slot (Dr. Octo Rex) 116
Selecting in Matrix 841
Nurse Rex Loop Player (Kong) 251
Selecting in Redrum 475
O PEQ-2 EQ 800
Offset R PH-90 Phaser 796
Delay (The Echo) 772 Phase Controls (Subtractor) 509
Feedback (The Echo) 773 Phaser 796
Online Help 911 Physical Bass Drum (Kong) 256
Open Companion 912 Physical Snare Drum (Kong) 256
Open Companion on Startup 912 Physical Tom Tom (Kong) 256
Open with Browser Shown 910 Ping-Pong Delay (The Echo) 773
Orkester Sound Bank 102 Pitch Adjust (Neptune) 634
Overdrive/Resonator (Kong) 268 Pitch Bend Range (Kong) 246
Pitch Correction
P Automatic 636
P Button (Mixer 14-2) 853 Manual via MIDI 640
Pad Group (Kong) 240 Players 882
Pad Settings (Kong) 237, 239 Polyphony
Pads (Kong) 235 Dr. OctoRex 125
Pangea World Instruments 419 Malström 292
Parametric EQ 800 NN-19 416
Parametric EQ (Kong) 265 NN-XT 384
Paste Patch 85 Subtractor 526

917 INDEX
 Polytone Dual-Layer Synthsizer 435 REX files
Post-fader Sends (Mixer 14-2) 853 Loading in Dr.Rex 109
Preserve Expression (Neptune) 640 Loading in NN-19 401
Preview (Dr. Octo Rex) 111 Loading in NN-Nano 248
Programmer CV In (Combinator) 821 Loading in NN-XT 351
Pulsar Dual LFO 860 Loading in Nurse Rex Loop Player 251
Pulveriser 652 Loading in Redrum 473
Dirt 654 REX Loops
Filter 655 Editing Slices 113
Follower 657 Editing Sound 118
Squash 654 Loading 111
Tremor (LFO) 656 Playing in Dr. Octo Rex 109, 112
Ring Modulation (Subtractor) 512
Q Ring Modulator (Kong) 266
Quartet Chorus Ensemble 661 Ripley Space Delay 671
R Roll (The Echo) 771
Rack Extensions and Content 912 Root and Scale (Neptune) 636
Radical Piano 457 Routing (Cables)
Randomize Automatic 78
Pattern (Redrum) 479 Manual 77
Pattern (RPG-8) 878 RPG-8
Randomize Pattern (Matrix) 847 About 870
Rattler (Kong) 267 Arpeggiator parameters 875
RCY files 109 MIDI-CV Converter Parameters 874
Reason Factory Sound Bank 102 Setting up 871
Reason Library (Location in Browser) 94 Run button 476
Reason Rack Plugin Window 34 Run Button (Redrum) 474
ReCycle 108 RV-7 Digital Reverb 791
Redo 40 RV7000 Mk II Advanced Reverb 690
Redrum Rytmik Drum Machine 487
Individual Outputs 486 S
MIDI Notes for 485
S1/S2 controls (Redrum) 481
Parameters 481
Samples
Patches 472 Browsing for 90
Programming Patterns 474
Cross-browsing 91
ReFills
Kong 233
About 102
Missing 101
Registering Reason 26
NN-19 402
Remove Long-Term Authorization 29
NN-XT 357
Render Audio Using Host Buffer Size Setting 46, 912 Redrum 471
Reset Band Levels (Vocoder) 594
Searching for 96
Reset Device 85
Scale Memory (Neptune) 636
Resolution (Redrum Pattern) 477
Scales & Chords (Player) 895
Reverb
Scream 4 704
RV-7 791
Scroll to Connected Device 78
RV7000 Mk II 690
Search Field (in Browser) 96
Select Slice Via MIDI (Dr. OctoRex) 113
Send Out (Redrum) 481
Sends
Mixer 14-2 853
Redrum 481
Settings dialog 910

918 INDEX
 Shift The Echo 770
Pattern (Matrix) 847 Color Section 774
Pattern (Redrum) 479 Delay Section 772
Pattern (RPG-8) 878 Diffusion 773
Shortcuts (Location in Browser) 94 Ducking 776
Show Parameter Value Tool Tip 911 Feedback Section 773
Shuffle 105 Mode 771
Shuffle (Redrum) 478 Modulation Section 775
Sidechain Tool 711 Theme 911
Signal Flow Graphs (on Effect Devices) 781 Thor
Slice Edit Mode (Dr. Octo Rex) 115 Assignable Controls 533
Slice Trig (Kong) 253 Oscillator section 538
Slices Using the Programmer 534
About 108 Tie Switch (Matrix) 846
Making Settings for (Dr. Octo Rex) 114 To Main 105
Selecting 113 Tone Generator (Kong) 261
Softube Amps 720 Tool Tips 38, 911
Solo Transient Shaper (Kong) 263
Mixer 836 Transpose (Neptune) 641
Mixer 14-2 853 Tremor (Pulveriser) 656
Redrum 481 Trig Next Loop (Dr. Octo Rex) 110
Sound Packs (Location in Browser) 94 Trigger Buttons (Redrum) 472
SoundFonts Type Buttons (in Browser) 95
NN-19 403
NN-XT 351
U
UN-16 Unison 798
Sounds
Undo 40
Missing 101
Unipolar Curves (Matrix) 845
Spider
Unison 798
Audio Merger and Splitter 785
User Library (Location in Browser) 93
CV Merger and Splitter 787
Utilities 68
Spread (Pulveriser) 656
Squash (Pulveriser) 654 V
Stereo Tool 435, 671, 711, 725, 829 Vocoder
Stop Hit Type (Kong) 254 About 590
Subtractor How it Works 590
About 504 Parameters 593
External Modulation 526 Setting up 591
Filter 513 Using as EQ 592
Oscillators 505 Voice Synth (Neptune) 643
Waveforms 506
Support Generators (Kong) 260 W
Sweeper Modulation Effect 729 Wide Vibrato (Neptune) 635, 645
Synchronous Timed Effect Modulator 749 Z
Synth Bass Drum (Kong) 258 Zoom (Application) 35
Synth Hi-Hat (Kong) 259
Synth Snare Drum (Kong) 258
Synth Tom Tom (Kong) 258
T
Tags (Editing in Browser) 99
Tags Buttons (in Browser) 97
Tape Echo (Kong) 267
Text Field (in Browser) 96

919 INDEX
920 INDEX