Propellerhead Reason 12.2 User Manual
Propellerhead Reason Reason 12.2 is a powerful and versatile music production software that allows you to create, record, edit, and mix your music with ease. It comes with a wide range of virtual instruments, effects, and tools, as well as a powerful sequencer and mixer. Reason is perfect for both beginners and experienced musicians alike.
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The information in this document is subject to change without notice and does not represent a commitment on the part of Reason Studios AB. The software described herein is subject to a
License Agreement and may not be copied to any other media except as specifically allowed in the License Agreement. No part of this publication may be copied, reproduced or otherwise transmitted or recorded, for any purpose, without prior written permission by Reason Studios AB.
©20 2 1 Reason Studios and its licensors. All specifications subject to change without notice.
Reason, Reason Intro, Reason Lite and Rack Extension are trademarks of Reason Studios AB.
All other commercial symbols are protected trademarks and trade names of their respective holders. All rights reserved.
Table of Contents
Introduction
Welcome!
Introducing Reason Rack Plugin
22
Conventions in the manual
22
The Authorization system
About Rack Extensions
25
The "Update Rack Extension Licenses" alert
25
The "Some licenses only available online" alert 25
Getting all your content
About automatic update checks
Overview
27
Adding Reason Rack Plugin in your project
28
The Reason Rack Plugin window
29
Editing parameters
30
Menus and alpha-numeric controls 32
Tool Tips
Context menus
33
33
33
33
Undo and Redo
Audio and MIDI Basics
General audio and MIDI handling
Typical input/output configurations 36
The I/O device
Audio settings
Render audio using host buffer size setting 40
About Plugin Delay Compensation
Using Reason Rack Plugin as an Instrument
Creating an instrument
42
Browsing for patches
43
Adding effects
Layering instruments
45
Using separate audio outputs
46
Adding Players
47
Using Mixer devices
48
Detailed control over MIDI note input
50
Using Reason Rack Plugin as an Effect
Creating an effect
Browsing for patches
53
Creating effect chains
55
Using sidechain inputs
55
4
TABLE OF CONTENTS
5
Working in the Rack
57
Creating devices
58
Selecting devices
Moving devices
60
Re-routing devices
60
Deleting devices
61
Replacing devices
61
Cut, Copy, Paste and Duplicate devices
61
Naming devices
61
Loading patches
Saving patches
73
Opening the Browser and setting Browse
Focus
About cross-browsing
75
Special instances of cross-browsing
75
Browser details
The Sound Banks and fixed Locations
78
User Locations and Favorite Lists 79
Browsing samples and loops
81
Searching in the Browser
Handling Missing Sounds
About ReFills
Routing Audio and CV
Working on the back of the rack
Signal types
66
66
66
Manual routing
Checking and following cable connections
68
Auto-routing
68
The I/O device
Introduction
86
The back panel
86
The front panel
87
87
87
Sounds, Patches and the
Browser
71
Kong Drum Designer
About patches
72
About the “Load Default Sound in New Devices” setting
72
Introduction
90
TABLE OF CONTENTS
6
Overview
90
90
91
About using custom backdrops 91
About file formats
Using patches
92
Checking the sounds in a Kit Patch 93
94
94
Pad Settings
95
95
Copying & Pasting Drums between Pads 96
The Drum and FX section
99
100
102
104
105
The Drum modules
Physical Bass Drum, Snare Drum and Tom Tom
114
Synth Bass Drum, Snare Drum and Tom Tom
116
117
The Support Generator modules
118
118
119
The FX modules
120
Using CV modulation of Bus FX and Master FX parameters
120
121
123
123
124
125
TABLE OF CONTENTS
Connections
127
127
128
128
Using Kong as an effect device
128
Using external effects with Kong
129
Redrum Drum Computer
131
Introduction
About file formats
133
Using patches
134
Checking the sounds in a patch
134
135
135
Programming patterns
136
137
Setting pattern resolution 139
140
140
140
The Enable Pattern Section switch
141
141
Copy MIDI Notes to a sequencer track 141
Redrum parameters
143
143
Using Redrum as a sound module
147
Connections
148
Dr. Octo Rex Loop Player
Introduction
150
About REX file formats
Slice handling
155
Editing Slices in the Waveform Display 156
157
Dr. Octo Rex panel parameters
158
158
Enable Loop Playback and Run 159
Dr. Octo Rex synth parameters
160
Copy MIDI Notes to a sequencer track 160
162
162
Setting number of voices - polyphony 167
167
160
149
Loading and saving Dr. Octo Rex patches
151
About the Dr. Octo Rex patch format 151
Playing Loops
Switching playback between Loop Slots
152
Adding Loops
153
153
154
Cut/Copy and Paste Loops between Loop Slots 154
Playing individual Loop Slices
Connections
168
168
168
168
Europa Shapeshifting
Synthesizer
Introduction
170
Panel overview
Signal flow
172
169
Playing and using Europa
173
173
Global performance and “play” controls
173
Panel reference
Sound Engines On/Off and Edit Focus section 175
175
180
The User Wave and Mixer section
185
190
194
The Modulation Bus section 198
Connections
202
CV Modulation inputs and outputs
202
202
Tips and Tricks
203
Creating an individual “pre amp envelope” for a Sound
Recording display movements in the sequencer
204
7
TABLE OF CONTENTS
Grain Sample Manipulator
205
Introduction
A few words about granular synthesis 207
Panel overview
208
Playing and using Grain
Loading and saving patches 209
Global performance and “play” controls 209
210
Panel reference
211
The Playback Algorithms section
213
219
220
225
225
229
Connections
233
CV Modulation inputs and outputs 233
Tips and Tricks
234
Automating sample playback parameters from the sequencer 234
Panel reference
244
244
252
The Filter Envelope and Amp Envelope sections
256
258
259
259
Connections
261
261
261
261
261
Tips and Tricks
Optimizing performance/DSP Load 262
Creating a “velocity layered” instrument
262
Extending the sample “tail” (without looping) 263
Automating the sample Start and End markers
264
Thor Polysonic Synthesizer
265
Mimic Creative Sampler
Introduction
Panel overview
237
Signal flow
Playing and using Mimic
Loading and saving patches 239
239
239
Performance and “play” controls
242
235
Introduction
266
Thor elements
The Controller panel
Using the Programmer
270
Basic connections - a tutorial
271
281
281
285
285
8
TABLE OF CONTENTS
288
Modulation bus routing section
290
Step Sequencer
299
Connections
303
327
Setting Number of Voices - Polyphony 328
About the Low Bandwidth button 328
External Modulation
328
Connections
329
329
330
Subtractor Synthesizer
Introduction
306
The Oscillator Section
307
307
Setting Oscillator 1 Frequency - Octave/Semitone/Cent
309
Oscillator Keyboard Tracking 309
310
314
The Filter Section
315
315
318
318
Envelopes - General
320
321
322
LFO Section
Play Parameters
325
Pitch Bend and Modulation Wheels 326
327
Malström Synthesizer
Introduction
332
332
333
Loading and Saving Patches 333
The Oscillator section
335
Controlling playback of the graintable 335
336
The Modulator section
337
The Filter section
339
340
342
Routing
345
346
The play controls
Polyphony - setting the number of voices
350
350
The Pitch Bend and Modulation wheels
351
The Modulation wheel controls 352
9
TABLE OF CONTENTS
Connections
353
353
354
354
Routing external audio to the filters
355
Using the ID8
373
373
373
About saving edited Sounds 374
Monotone Bass Synthesizer
357
Introduction
Panel overview
359
Signal flow
Playing and using Monotone
Loading and saving patches 361
361
Global performance and “play” controls 361
Panel reference
363
365
367
367
368
Connections
369
ID8 Instrument Device
Introduction
Rytmik Drum Machine
375
Introduction
Panel overview
Signal flow
378
Global controls
378
378
The Drum Channel sections
379
Muting and soloing Drum Channels 379
Setting the Drum Channel volumes 379
Setting the Send Effect levels 380
The Sample Playback section
Setting the Sample Start and End
381
381
Setting Fade In and Fade Out 382
The Insert Effects section
383
The Master FX section
385
386
388
388
Connections
390
390
10
TABLE OF CONTENTS
Radical Piano
Introduction
392
Using Radical Piano
394
394
394
395
398
399
399
Connections
400
Additional external control
Klang Tuned Percussion
403
Connections
417
417
417
Pangea World Instruments
419
Introduction
420
Panel overview
Using Pangea
421
Global performance and “play” controls
421
Panel controls
422
422
429
431
432
Connections
434
434
434
Introduction
Panel overview
404
Using Klang
405
Global performance and “play” controls
405
Panel controls
413
416
Humana Vocal Ensemble
Introduction
436
Panel overview
Using Humana
437
Global performance and “play” controls
437
11
TABLE OF CONTENTS
12
Panel controls
Introduction
444
Connections
451
NN-XT Sampler
Panel overview
455
Loading complete Patches and REX files
456
456
Loading complete REX files as Patches 457
Using the main panel
458
The Pitch and Modulation wheels
458
458
459
459
Overview of the Remote Editor panel
461
462
462
About Samples and Zones
Selections and Edit Focus
465
467
Adjusting parameters
467
Adjusting Synth parameters 467
Adjusting Group parameters 467
Managing Zones and Samples
469
469
About file formats and REX slices
470
Adding more samples to the Key Map
470
Quick browsing through samples
471
471
471
472
Working with Grouping
472
Moving a Group to another position in the List 473
Moving a Zone from one Group to another
473
Selecting a Group and/or Zones in a Group
474
474
Working with Key Ranges
474
474
About the Lock Root Keys function 478
About the Solo Sample function
479
Setting Root Notes and Tuning
481
Setting the Root Note manually 481
Setting the Root Note and Tuning using pitch detection
482
About changing the pitch of samples
482
Using Automap
482
Layered, crossfaded and velocity switched sounds
483
483
483
Setting velocity range for a Zone
485
About Crossfading Between Zones
485
Setting crossfading for a Zone 487
Using Alternate
487
About the Alternate function 487
TABLE OF CONTENTS
Sample parameters
488
488
489
489
489
Group parameters
491
Synth parameters
492
494
495
Connections
503
503
NN-19 Sampler
505
Introduction
General sampling principles
Multisampling vs. single samples 506
About audio file formats
507
Loading REX Files as Patches 507
About Key Zones and samples
Loading a Sample into an empty NN-19
508
509
Loading REX slices as samples 509
13
TABLE OF CONTENTS
510
510
Setting the Key Zone Range 510
About Key zones, assigned and unassigned samples
511
Adding sample(s) to a Key Map 511
512
Removing sample(s) from a Key Map
512
Removing all unassigned samples
512
Rearranging samples in a Key Map 512
512
About the Solo Sample function
513
Automap Samples
Mapping samples without Root Key or Tuning information
514
How Mapping Information is saved
514
NN-19 synth parameters
515
Play Parameters
Pitch Bend and Modulation Wheels
521
521
522
Setting Number of Voices - Polyphony 522
522
Connections
523
523
MIDI Out Device
Introduction
526
Using the MIDI Out Device
Setting up for MIDI controlling an external track/plugin
526
Modulating MIDI Controllers from CV signals 527
Connections
528
528
Quartet Chorus Ensemble
Introduction
Panel reference
530
536
Connections
538
538
538
529
Sweeper Modulation Effect
539
Alligator Triple Filtered Gate
559
Introduction
Overview and signal flow
Parameters
Common effect device parameters
562
562
564
566
568
Audio connections
CV connections
570
570
570
The built-in patterns
571
Methods and Tips
572
Playing the Alligator live 572
Playing the gates from Matrix patterns
572
Controlling other sounds and effects
572
Introduction
Panel reference
540
548
551
The Audio Follower Modulator 555
Connections
556
556
557
Pulveriser
573
Introduction
Parameters
Common effect device parameters
574
579
14
TABLE OF CONTENTS
580
Modulation inputs and outputs
581
Demolition tips and tricks
582
582
582
The Echo
583
Introduction
Parameters
Common effect device parameters
584
587
589
590
CV/Gate inputs
The Breakout Jacks
591
Tips and Tricks
592
Distorted external feedback 592
Scream 4
Sound Destruction Unit
593
Scream 4 Sound Destruction Unit
598
BV512 Vocoder
Introduction
602
602
Setting up for vocoding
Using the BV512 as an equalizer
604
BV512 parameters
605
Connections
Tips and tricks
608
Choosing a modulator sound 609
Using the modulator as carrier
609
610
Using the individual band level connections 610
“Playing” the vocoder from a MIDI keyboard
612
Using the BV512 as a reverb 613
RV7000 Mk II Advanced
Reverb
617
Overview
618
619
Reverb algorithms and parameters
620
Common effect device parameters
620
About the main panel parameters
620
620
621
622
624
15
TABLE OF CONTENTS
The EQ section
628
The Gate section
629
CV Inputs
Neptune Pitch Adjuster and
Voice Synth
Introduction
Overview and basic concepts
633
634
Setting up for pitch processing
634
Using pitch correction
Basic settings for pitch correction 635
Using automatic pitch correction
636
640
Using pitch shifting (Transpose)
Using Formant control
642
Using the Formant function 643
Using the Voice Synth
Panel parameters
644
Level Meter and Bypass/On/Off switch
644
645
645
646
647
Connections
648
648
649
16
TABLE OF CONTENTS
Softube Amps
Introduction
652
Using the Softube Amps
653
653
Selecting Amp and Cabinet model 653
About the Amp and Cabinet models 654
655
Audiomatic Retro Transformer
657
Introduction
Using Audiomatic Retro Transformer
658
658
659
660
660
Connections
661
661
Channel Dynamics
Compressor & Gate
663
Introduction
Panel reference
664
664
665
17
666
Connections
668
668
668
Channel EQ Equalizer
Introduction
Panel reference
670
670
Connections
674
Panel overview
Using Synchronous
684
Drawing and assigning modulation curves - a tutorial
684
Editing modulation curves - a tutorial 686
Panel reference
694
696
About automation of display section parameters
698
Connections
700
700
Master Bus Compressor
675
Introduction
Panel reference
676
676
677
Connections
679
679
679
The MClass Effects
701
The MClass effects
The MClass Equalizer
The MClass Stereo Imager
704
The MClass Compressor
705
The MClass Maximizer
Synchronous
Timed Effect Modulator
Introduction
TABLE OF CONTENTS
Half-Rack Effects
709
Common effect device features
DDL-1 Digital Delay Line
712
CF-101 Chorus/Flanger
Spider Audio Merger & Splitter
Spider CV Merger & Splitter
RV-7 Digital Reverb
D-11 Foldback Distortion
ECF-42 Envelope Controlled Filter
PH-90 Phaser
726
UN-16 Unison
COMP-01 Auto Make-up Gain
Compressor
729
PEQ-2 Two Band Parametric EQ
730
Moving devices out of a Combi 739
Deleting devices in a Combi 739
Configuring the Combinator panel
740
Opening the Configuration panel
740
741
Selecting front panel background color
741
Selecting a backdrop image 741
Selecting, positioning and editing front panel controls
743
Assigning panel controls to parameters in the Editor
Key Mapping instrument devices
747
Setting Velocity Ranges for instrument devices
749
Using Modulation Routing
About Rotary/Slider and Switch controls
751
751
Assigning panel controls to device parameters 751
CV Connections
756
756
756
Wheel CV In and Source CV In 757
The Combinator
731
18
Introduction
Pulsar Dual LFO
759
Combinator overview
Creating a Combinator device
734
Creating an empty Combinator device
734
Creating a Combinator by combining devices
734
Creating a Combinator by browsing patches 734
About internal and external audio connections
735
External audio connections 735
Internal audio connections 735
736
Adding devices to a Combi
737
Creating new devices in a Combi 737
Adding existing devices to the Combi from the rack 738
Combinator handling
739
Introduction
Panel parameters
762
762
LFO 2 to LFO 1 modulation parameters 762
765
Modulation inputs and outputs
767
767
767
Tips and Tricks
Patch between LFO 1 and LFO 2 on the back for more flexibility
768
Using Pulsar as a monophonic synth
768
TABLE OF CONTENTS
RPG-8 Arpeggiator
Introduction
Using the RPG-8
771
RPG-8 Parameters
MIDI-CV Converter parameters 774
777
CV connections
778
Tips and tricks
780
The Channel Strip
794
The Mixer signal flow
About the EQ modes
The Auxiliary Return Section
The Master Fader
Connections
Chaining several Mixer 14:2 devices
799
Matrix Pattern Sequencer
Introduction
About the three Output types 782
Programming patterns
783
787
788
789
789
789
Copy MIDI Notes to a sequencer track 790
Example usage
791
Using the Matrix for modulation
791
Programming “Acid Style” lead lines
792
Mixer 14:2
793
Introduction
The Line Mixer 6:2
Introduction
802
Channel parameters
802
The Auxiliary Return section
Master level
802
Connections
Working with Players
About this chapter
Overview
806
807
Using Players
807
808
808
About Players in Combinators 808
19
TABLE OF CONTENTS
Common Player device parameters
808
Getting the Player MIDI output onto a track in your DAW
809
Dual Arpeggio
810
Note Echo
818
Scales & Chords
819
819
Beat Map
824
825
826
827
829
Beat Map and the main sequencer
829
830
Tips & Tricks
831
Generating scale-correct arpeggios from single notes
831
Generating chord arpeggios 831
831
Using a Scales & Chords device as a “MIDI Note monitor”
832
Settings
Index
The Reason Rack Plugin Settings dialog
834
834
835
Load default sound in new devices
835
835
835
835
Render audio using host buffer size setting 836
837
20
TABLE OF CONTENTS
Chapter 1
Introduction
Welcome!
!
This is the Operation Manual for Reason Rack Plugin, part of the Reason Version 12 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 12 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 12 application, Reason Rack Plugin works on two platforms: Windows 10 (64-bit) and macOS 10.13 (High Sierra) (64-bit) or later. All functions are the same. If your DAW host supports cross-platform functionality (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.
22
INTRODUCTION
Frames and circles (call-outs)
Rack
Browser
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.
23
INTRODUCTION
The Authorization system
Reason Rack Plugin is authorized in the same way as the standalone Reason application, and uses the same license.
Here's how it works:
You need a user account on www.reasonstudios.com
and the Reason license must be registered on your account.
• If you purchased Reason directly from the Reason Studios shop, you already have an account, and the license was automatically registered when you purchased it.
• If you purchased the boxed version of Reason, you will find the license in the package, along with instructions on how to register.
Once the license is registered on your account, you can run Reason Rack Plugin with Internet Verification.
When you first open Reason Rack Plugin in your DAW host, a login dialog will appear:
D
Enter your User name or the e-mail address you used for your user account, and the password.
Reason Rack Plugin will contact the servers and verify your Reason license.
• If you turn on "Remember my password" you will only have to perform the log in procedure once. The next time you launch Reason Rack Plugin, license verification will happen silently in the background.
Refer to “Remove stored password”
for information on how to deactivate the “Remember my password” function.
If you need to use Reason Rack Plugin without Internet access you can instead authorize your computer. To do this, click the "More Options" link in the Login dialog and follow the instructions!
24
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 program or plugin, they behave just like built-in devices.
D
To browse, trial or purchase Rack Extensions, visit reasonstudios.com/shop
D
To download and install Rack Extensions that you own, visit your user account page .
D
To manage your installed Rack Extensions, use the Authorizer application that was installed with Reason.
.
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 with online verification (logging in), 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
Extensions.
The "Some licenses only available online" alert
This dialog may appear if you have authorized your computer or ignition key hardware to run Reason without logging in. It happens because some Rack Extension licenses (Trials, beta versions, rentals, etc) cannot be authorized offline this way - they require that you log in. Choose whether to log in to your account or to run this Reason Rack Plugin session without these particular Rack Extensions.
Getting all your content
The main sound banks are installed when you install Reason, as are a great many instrument and effect devices.
However, there is also additional, optional content that you should check out:
• Reason and Reason Suite include the Drum Supply and Loop Supply ReFills and four additional instrument devices (Radical Piano, Klang Tuned Percussion, Pangea World Instruments and Humana Vocal Ensemble).
To download these, you need to launch the stand-alone version of Reason and use the Manage Content function on the Window menu. After installation, these ReFills and devices will be available the next time you open Reason
Rack Plugin.
• If you have purchased Reason Suite, you automatically got licenses for a number of Reason Studios Rack Extensions.
To download these, go to your user account page and click the Sync All button.
25
INTRODUCTION
About automatic update checks
When you launch Reason Rack Plugin it automatically checks for new updates on the Reason Studios web site (provided that your computer has Internet connection). If a new update is found, an alert will be shown on the Global
Panel at the top of the Reason Rack:
• Clicking the alert will launch your web browser and download an installer for the new version.
Once download is complete, quit the DAW and run the installer to update Reason Rack Plugin.
Alternatively, you can download and install the update directly from within the standalone version of Reason. This will also update the Reason Rack Plugin.
26
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
Rack
The Reason Rack Plugin has three main areas:
• The Browser
The Browser is where you create devices, browse patches and load samples. This area is folded by default, but you can unfold it by clicking the circle button in its top left corner:
It will also automatically appear when you click the Browse button on a device. See the
Browser” chapter for more about browsing patches and samples.
• The Rack
The Rack is where you build your sound, by placing instrument and/or effect devices.
• The Tutorial area
This contains tutorials showing you how to use Reason Rack Plugin. You can fold and unfold this area by clicking the circle button in its top corner:
28
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 Undo/Redo, Flip Rack and a button for opening the Settings dialog. To the right is also a MIDI In indicator:
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-resolution panels in the background, which might take a little while.
About different Themes
In Reason Rack Plugin you can choose from a couple of different visual themes, i.e. how the user interface 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 appears. For the changes to take effect, you need to quit and relaunch your DAW host.
29
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 controls 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.
30
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 adjusting a slider).
31
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 arrow 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” ).
32
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
If you context-click on an automatable control (a mixer parameter, a device parameter, a fader, etc.), the context menu will contain the following items:
• Functions for mapping a MIDI control to the parameter, or clearing existing mapping.
• A Reset function for resetting the parameter to its default value.
This is the same as [Ctrl](win)/[Cmd](Mac)-clicking the parameter.
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 contain 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 Combinator 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.
33
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.
34
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.
•
.
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
MIDI Out from
Master Keyboard
1-2
MAIN OUT
3-4
TO MAIN
5-6
TO MAIN
7-8
TO MAIN
9-10
TO MAIN
11-12
TO MAIN
13-14
TO MAIN
15-16
TO MAIN
AUDIO IN
MAIN SIDECHAIN
AUDIO OUT
SHUFFLE
PITCH MOD
ID8 audio output signals
Piano
Grand
Upright
Dance
Vibes
C
D
A
B
Delay
Chorus
Reason Rack Plugin
ID8 1
VOLUME instrument device
Live and playback MIDI data from Instrument Track in sequencer
Stereo instrument example.
MIDI In
ID8 1
DAW Sequencer
FADER
MUTE
L R
SOLO
56
MIX CHAN...
DAW Mixer
36
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
MIDI Out from
Master Keyboard
1-2
MAIN OUT
3-4
TO MAIN
5-6
TO MAIN
7-8
TO MAIN
9-10
TO MAIN
11-12
TO MAIN
13-14
TO MAIN
15-16
TO MAIN
AUDIO IN
MAIN SIDECHAIN
AUDIO OUT
KICK
DRUM KIT 1
SNARE HHCL HHOP
DRM audio separate outputs
TOM1 TOM2 TOM3
Reverb
Compression
CYM
VOLUME
DRM 1
DRM
DRUM MACHINE
SHUFFLE
Reason Rack Plugin
Live and playback MIDI data from Instrument Track in sequencer
FADER FADER FADER FADER
MUTE
L R
SOLO MUTE
L R
SOLO MUTE
L R
SOLO MUTE
L R
SOLO
-
56
MIX CHAN...
-
56
MIX CHAN...
-
56
MIX CHAN...
-
56
MIX CHAN...
DAW Mixer
MIDI In
ID8 1
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
1-2
MAIN OUT
3-4
TO MAIN
5-6
TO MAIN
7-8
TO MAIN
9-10
TO MAIN
11-12
TO MAIN
13-14
TO MAIN
15-16
TO MAIN
AUDIO IN
MAIN SIDECHAIN
AUDIO OUT
SHUFFLE
FX audio out
FX audio in
FX 1
ECHO CHAMBER
FX 2 FX 3 FX 4
Reverb
Delay
VOLUME
ECO 1
E:C:O
MULTI FX
Reason Rack Plugin Effect
Stereo effect example.
FADER FADER FADER FADER
MUTE
L R
SOLO MUTE
L R
SOLO MUTE
L R
SOLO MUTE
L R
SOLO
MIX CHAN...
MIX CHAN...
MIX CHAN...
MIX CHAN...
DAW Mixer
37
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
1-2
MAIN OUT
3-4
TO MAIN
5-6
TO MAIN
7-8
TO MAIN
9-10
TO MAIN
11-12
TO MAIN
13-14
TO MAIN
15-16
TO MAIN
AUDIO IN
MAIN SIDECHAIN
AUDIO OUT
SHUFFLE
FX audio out
FX audio in
Threshold
Ratio
VOLUME
SC in
DUCK 1
Duck
MULTIBAND COMPRESSOR
FADER FADER FADER FADER
MUTE
L R
SOLO
12
MUTE
12
L R
SOLO MUTE
L R
SOLO
12
MUTE
L R
SOLO
12
MIX CHAN...
MIX CHAN...
MIX CHAN...
MIX CHAN...
DAW Mixer
Reason Rack Plugin Effect
Stereo effect with sidechain example.
38
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 devices 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.
39
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. However, 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.
40
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 the Instruments palette shown,
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.
42
USING REASON RACK PLUGIN AS AN INSTRUMENT
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.
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 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 the “Sounds, Patches and the Browser”
chapter.
43
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 context-clicking the instrument device and selecting an effect device from the
Effects sub-menu:
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 signals, for more complex effect setups!
44
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 Reason 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 receive notes and be heard at the same time. For more advanced layering techniques, you can use the Combinator de-
).
45
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 separate 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:
46
USING REASON RACK PLUGIN AS AN INSTRUMENT
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 separate output etc. See the documentation for the instrument device.
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 sequencer or drum machine.
Players are added from the Browser or from the context menu. They are listed on a separate palette or submenu.
47
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.
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:
48
USING REASON RACK PLUGIN AS AN INSTRUMENT
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
.
49
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 inside a Combinator. Devices inside a Combinator don't receive MIDI directly from the DAW host - instead, the Combinator 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 velocity splits. Read more about the Combinator in
!
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.
50
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 the Effects palette shown,
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!
52
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”
.
It's also possible to browse between different devices (rather than patches). For example, there are several different
Compressor devices available in Reason Rack Plugin, and many of them don't use patches.
53
USING REASON RACK PLUGIN AS AN EFFECT
1. Let's say you start by adding an M-Class Compressor to your rack:
2. Now, right-click the MClass Compressor device and select "Browse Effects" from the context menu.
The Browser opens with the Effect palette shown.
3. Scroll to find the Comp-01 Compressor/Limiter, then double click it:
It now automatically replaces the M-Class Compressor in the rack:
4. Continue until you've found the effect that suits you best.
This type of browsing, where you replace a device with another device type, is called cross-browsing. You can also use drag and drop from the Browser to do this.
54
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 order 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 parallel 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
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.
55
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.
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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.
Either:
D
Double click an instrument icon to add that device,
D click "Browse Instruments" or “Browse Effects” to open the Browser with the corresponding palette shown,
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:
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WORKING IN THE RACK
D
Select a device or patch in the Browser and click Create (or double clicking it the device).
• Drag and drop a device or 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 PX7 FM 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 dropping in the empty rack or below devices means creating a new device with that patch loaded.
Selecting devices
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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).
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To select several devices, hold down [Ctrl](Win) or [Cmd](Mac) and click on the desired devices.
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Hold down [Shift] and click to make a continuous (range) selection.
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To de-select all devices, click in the empty part of the rack.
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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
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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.
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WORKING IN THE RACK
Deleting devices
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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
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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.
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 especially useful if your rack contains several devices of the same type and you need to separate them.
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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.
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WORKING IN THE RACK
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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:
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Click Flip Rack to get there (or select Flip Rack from the context menu):
All audio and CV connections are represented by cables.
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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:
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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
Hides auto-routed cables.
Result
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.
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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 modulating 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).
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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 settings.
• 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 stereo pair, the right will automatically be connected as well.
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To change a connection, click and hold the jack to grab the cable. Then drag it to another jack.
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To disconnect a cable, click the jack at either end to grab the cable and drop it away from any jack.
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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.
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To disconnect a jack, right-click it and select "Disconnect" from the context menu:
Checking and following cable connections
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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:
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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 connected 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.
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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.
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ROUTING AUDIO AND CV
• Devices inside a Combinator will not auto-route outside the Combinator device.
• 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:
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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 embedded in a ReFill (see
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. This will also determine the default folder when you browse for patches for the device.
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
Loading patches can be done in the following ways:
• By using the patch selector directly on the device panel.
Either click and select from the menu that appears, or use the Previous/Next Patch buttons next to the display.
• By opening the browser, dragging a patch from there and dropping it on a device.
• By dragging and dropping patch files from a Windows / Mac window and dropping it on a device.
• By clicking the Browse Patch button for a device or selecting Browse from the context menu.
This opens the browser and gives the device "browse focus" - read more about this in
“Opening the Browser and setting Browse Focus” below.
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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.
Opening the Browser and setting Browse Focus
You open and close the Browser by clicking the circle button in its top left corner:
It is also opened automatically when you set browse focus to a device. You set browse focus in one of the following ways:
• By clicking the Browse Patch button on a device:
• By selecting Browse from the context menu for a device.
• By selecting a device in the rack and clicking the browse button at the top of the Browser:
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SOUNDS, PATCHES AND THE BROWSER
Browse focus is indicated by an orange header in the Browser. The device is shown with orange side bars and an orange-colored patch section:
In this mode, the Browser is "locked" to the device and only displays patches compatible with this particular device type. For example, if you set browse focus to an instrument device such as Europa, only instrument patches for this device will be shown.
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To load a patch from the Browser in Browse Focus mode, double click the patch in the Browse list or select it and click the Load button at the bottom of the Browse list.
You can also step up or down in the list of patches shown with the arrow buttons below the list - this will automatically load the selected patch
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To go back to the original patch for a device in this mode, click the Revert button at the bottom of the Browse list.
Note that the Revert button appears only after you have loaded at least one other patch in the device:
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When you're happy with the loaded patch, you clear browse focus by clicking the X button at the top of the browser (or by clicking elsewhere in the rack):
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SOUNDS, PATCHES AND THE BROWSER
Why can’t I see all files?
Sometimes, some patches or samples may seem to be missing from a folder, or a folder might look completely empty.
Check if the Browser is in Browse Focus mode - is the header orange? Remember, in this mode, the Browser will not show all items, only those that match the selected device. This may mean that only instrument patches or effect patches are shown, or only loops or samples!
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Click the X button to leave Browse Focus mode and see all items again.
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Browser settings
In the Settings dialog are two options for the Browser:
Option
Open with Browser Shown
New devices get browse focus
Result
By default, the Browser area is folded when you open Reason Rack Plugin. Tick this option if you want it to be shown right away.
If this is on when you create a device (and the Browser area is shown), the device will get browse focus, as if you had clicked its browse button. This means that the
Browser will automatically go to the default folder for the device and show the right device type (Instrument, Effect, Utility or Player).
About cross-browsing
When you are browsing patches for an instrument device, you are free to load any kind of instrument patch. For example, if you're browsing patches for a Europa synthesizer, you can load a Radical Piano patch. This will replace the
Europa device with a Radical Piano device and load the patch.
Cross-browsing is useful in several ways:
• When you're looking for a sound of a certain type (e.g. a synth bass), you don't have to care about which device to select - you can just search in the Browser and step through the list of patches in the search result.
!
• You can make Favorite lists with different patches and step through these freely (see
!
• By using the Instruments and Effects palettes at the top of the browser, you can even cross-browse between devices that don't use patches.
You can cross-browse Instruments, Effect devices or Players, but you can only do it within category (it isn't possible to replace an effect with an instrument, etc).
Cross-browsing isn't available for Utility devices.
Special instances of cross-browsing
There are a few instances when cross-browsing between device types might lead to lost cable connections in the rack:
• Non-standard audio connections may be lost.
An example is replacing an NN-XT, which can use up to 16 audio outputs, with a Subtractor which only has one audio output.
!
• CV connections on the back panel may be lost.
The only connections that are retained between device types are Sequencer Control CV/Gate in.
If you encounter such situations and you want to restore the original connections, use the "Undo" function.
Browsing back to the original device patch will not restore lost connections.
SOUNDS, PATCHES AND THE BROWSER
Browser details
1
2
3
4
5
6
7
The Browser when using the Browse Patch button/function on a Subtractor instrument device.
The Browser has the following sections:
• 1. Browse focus field
Shown in orange when a device has browse focus. Click the X button to the right to set or clear browse focus.
• 2. Navigation field
Shows the name of the current folder. To the left are back and forward buttons, and to the right is an Up button for going to the parent folder.
• 3. Search field
Type something and click Search to find patches or samples of that name in the current folder.
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• 4. Locations and favorites
The column to the left contains shortcuts for going directly to various locations. At the top are the four palettes (Instruments, Effects, Utilities and Players), showing all included devices. In the middle are fixed locations such as the sound banks, and below these you can add your own locations and favorites (see
).
• 5. Browse list
Shows the contents of the current folder, with hierarchical subfolders. If you have made a search or selected a favorite list, this will be a flat file list instead. This is where you select the file to load. Scroll to the right to see more columns, for sorting the list.
• 6. Controls
The area directly below the browser list holds buttons for Load, Previous/Next and Revert. If you are browsing
samples, there are also audition controls here (see “Browsing samples and loops” ).
• 7. Info area
Shows information about the selected item. This area can be folded with the arrow button in the left corner.
q
When you are browsing patches for an instrument device, you are free to load any kind of instrument patch.
For example, if you're browsing patches for a Europa synthesizer, you can load a Radical Piano patch. This will replace the Europa with a Radical Piano device.
The Device Palettes
Click the shortcuts at the top of the area to the left in the browser to show one of the device palettes. These show icons of all devices in Reason Rack Plugin. At the top you'll find the built-in devices (under "Built-in Devices"); below you'll find all installed Rack Extensions sorted by manufacturer.
• You can fold or unfold a section in the palette by clicking the arrow button next to the manufacturer name.
• Searching when a device palette is shown will find both device names and manufacturer names.
Since most devices have longer names that describe their functionality, you can also search for something like
"compressor" or "drum" to quickly get a list of devices of that kind!
!
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To add a device from the palette, use drag and drop, double click it or select it and click the Create button at the bottom of the Browse list.
Note that Players can only be added directly above instrument devices in the rack.
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The Sound Banks and fixed Locations
Below the shortcuts to the device palettes you will find the sound banks installed with the program. The sound banks are all ReFills, a kind of component package for Reason which can contain sounds and effect patches, samples, REX files and more:
• Reason Sounds
This is a huge selection of instrument patches, sorted by category. If you for example need an Electric Piano sound, go to the Keys & Chords folder and then into the Electric Piano subfolder for a wide variety of patches for different devices.
• Orkester Sounds
A separate sound bank containing sampled orchestral instruments: strings, woodwinds, brass and more.
• Factory Sounds
Contains patches for all included devices (sorted by device type), samples and REX loops. This is also where you find effect patches.
• Drum Supply
A great collection of modern drum samples and drum kit patches for the Kong Drum Designer. This is an optional download, which can be downloaded from Manage Content on the Window menu in the stand-alone Reason program.
• Loop Supply
Drum and percussion loops in a number of contemporary styles, for use with the Dr Octo REX loop player device or the Nurse REX module in Kong. This is an optional download, which can be downloaded from Manage Content on the Window menu in the stand-alone Reason program.
• Rack Extensions
Most Rack Extensions come with patches that are contained within the Rack Extensions themselves. This virtual location lists all installed Rack Extensions and allows you to browse into them as if they were regular file folders.
A Rack Extension may also contain patches for other devices as well! For example, they often come Combi patches that combine the Rack Extension with stock effect devices, etc.
• Fixed folder locations
To navigate in the file system, there are shortcuts to your user folder and the desktop. These cannot be removed.
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User Locations and Favorite Lists
User Locations
User Locations are shortcuts to folders, either on your hard disk or within a ReFill or Rack Extension. It might for example be the folder where you save your patches, or a particularly useful subfolder in the Factory sound bank.
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Add a Location by dragging a folder from the Browse list to the lower part of the left browser section:
Adding a new Location.
You can then click it to quickly navigate into that folder.
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To remove a Location, right-click it and select Delete.
This will only remove the shortcut, not affect the actual folder or its contents.
If a User Location folder has been removed or renamed on disk, Reason Rack Plugin won't be able to find it. It will be shown with a yellow warning triangle in the list.
Favorites and Favorite Lists
In the same section as User Locations, you can add Favorites:
D
Drag a patch, loop or sample to the lower part of the left browser section to create an alias to it, for instant access.
The Favorite can be used like any other file in the browser.
D
To remove a Favorite, right-click it and select Delete.
This will only remove the alias, not the actual file.
It's also possible to create Favorite Lists. These are virtual folders where you can collect aliases to files you like or need to use often:
1. Click the + button below the Favorite section:
The “Create New Favorites Lists” button
A new Favorite List is created.
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SOUNDS, PATCHES AND THE BROWSER
D
If you like, type in a name for the list.
You can do this later by double clicking the list and typing.
2. Click the Favorite List to select it and show its contents.
At this point, it's empty.
3. Navigate elsewhere in the Browser, find the desired files and drag them onto the Favorite List.
Note that you can also add devices from the device palettes:
4. When done, click the Favorite List to see the items you have added:
D
If you like, you can drag them to reorder them.
q
Note that devices can also be included in Favorite Lists, by dragging them from any of the Palettes!
D
To delete a Favorite List, right-click it and select Delete. This will only remove the list and the shortcuts in it, not the actual items.
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Browsing samples and loops
Some devices support loading samples or REX loops (a special audio file format for playing back loops in any tempo).
As with patches, you typically can load samples and loops directly from the device panel or from the browser, but there are a couple of things to note:
• A device may have multiple sample browsers.
For example, the Redrum drum machine has ten drum channels, each of which has its own sample browser. When you use drag and drop to load samples, it matters on which drum channel you drop the sample:
Dragging and dropping a sample in Redrum channel 3.
When you click a Browse Samples button, that sets Browse Focus to that particular drum channel on the Redrum device:
Browsing for a sample in Redrum channel 2.
When a sample or REX file is selected in the Browser, audition controls appear below the browse list:
D
Click the small triangular Play button to audition the sound - or activate Auto to automatically play back the sounds when you select them.
The slider to the right sets the audition playback level.
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Searching in the Browser
The Search function allows you to search for patches, samples and loops:
1. Navigate to the folder, or click a Location in the Browser, in which you want to search.
The search will happen in that folder/Location and all its subfolders, so if you e.g. click the Factory Sounds shortcut, you will search the whole factory sound bank.
2. Click in the Search field, type in a text and click the Search button.
The search result is shown in alphabetical order, in separate groups. If you search in the Factory Sound Bank the result is divided into “Instrument Patches”, “Effect Patches”, “Other Patches”, “Samples” and “Folders”, with files and folders whose names match the search string.
If there are no search hits, the Browser list is empty.
D
To redo the search, type something else in the Search field and click Search again.
You don't have to go back to where you started to do this.
If you scroll to the right in the search result list, you'll see a Parent column, showing the parent folders for each item.
To go to the folder for an item, select it, click "Search Result" at the top and select "Go to Parent Folder" from the menu that appears:
D
To go back to the search result list, click the Back button:
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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:
This lists all samples and loops that couldn't be found.
2. Do one of the following:
D
If you know where the missing sounds are, go to that folder in the Browser and then click "Search Folder".
D
To search all your Browser locations for the missing sounds, click "Search Locations".
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To replace a single sample, select it in the list and click "Replace". Then find the replacement sample in the
Browser and double click it.
3. When you're done, close the Missing Sounds dialog.
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SOUNDS, PATCHES AND THE BROWSER
Missing sounds that haven't been replaced will be indicated with an asterisk (*) before the file names in the sample browsers on the device panels.
About ReFills
A ReFill is a kind of component package for Reason, which can contain sounds and effects patches, samples, REX files, SoundFonts and demo songs. If you like, you can compare ReFills to ROM cards for a hardware synthesizer. On your computer, ReFills appear as large files with the extension “.rfl”.
All sounds included with Reason Rack Plugin are embedded in three ReFills named “Reason Sounds”, “Orkester
Sounds” and “Factory Sounds”, which were (downloaded and) copied to the hard disk during installation.
Additional ReFills are available for purchase. You can also download ReFills from other Reason users on the Internet, purchase them from other sample manufacturers, 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 required.
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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 devices 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.
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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 device 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 (output). Note that if you have connected an instrument to Output 3-4 and turned on sum To Main, the audio out indicators 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
) and the Redrum drum computer (see
).
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.
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THE I/O DEVICE
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THE I/O DEVICE
Chapter 10
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 processing 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
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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 selected pad in the pad section. From the Drum Control Panel you can also load and save Drum Patches. See
.
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 references to any used drum samples (but not the actual samples themselves). Switching patches is the same as selecting a new drum kit.
Drum Patches
A Kong Drum Patch (Windows extension “.drum”) contains all settings for the selected Drum sound channel, including 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.
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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.
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 contains 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.
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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.
C1 C2 C3 C4 C5 C6
In the C1-D#2 range, each MIDI note will trig 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.
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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
!
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 Support Generator modules” and
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 removes 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”
.
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.
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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
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.
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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.
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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.
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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 assigned to one or more of these 9 Pad Groups independently. Pad Groups are useful if you, for example, want to trig several pads from a single pad, have one pad mute another, or randomly trig 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 example, if you assign an open hi-hat and a closed hi-hat sound to the same Mute Group, playing on one pad will automatically mute the sound assigned to the other pad.
Link Groups
Pads assigned to the same Link Group will play together when you trig 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.
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KONG DRUM DESIGNER
The Drum and FX section
Drum Control Panel
Drum Module Slot
FX1 Slot
FX2 Slot
Bus FX 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.
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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 external 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”.
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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”.
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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 parameters 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.
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• 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.
• The Aux 1 and Aux 2 Send knobs controls the level to any devices connected to the Aux 1 and Aux 2 Send Outputs on the back of the panel, see
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.
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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
,
and
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 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
).
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 channels.
The Pitch Bend Range knob for each of the 16 Drum channels
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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 different 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”
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.
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!
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
. Each Hit Type can contain one or several samples which can be layered and/or altered and controlled 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:
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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
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).
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Sample parameters
There are a number of parameters that are unique to each individual sample and Hit in the NN-Nano. These parameters 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 parameter:
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 surface, 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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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 depending on selected Hit Type for the assigned pad.
• Loop Trig
In Loop Trig mode, you trig 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
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
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 display.
Doing so will automatically move the start position of the subsequent chunk so that the chunks will always be adjacent 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 alternatingly. 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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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 stopping 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 playback 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).
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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.
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• 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 calculations 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
). 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
).
• 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.
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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
). 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.
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• 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
). 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.
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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 indicates the current parameter value.
By connecting cables to the CV modulation inputs on the back of the Kong panel, you can modulate the corresponding 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 preserved - but the CV signals will now control the first two parameters of the replacement module(s).
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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 transients 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.
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• 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 Compressor 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 compression 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.
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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 negative 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 (Qvalue). The parameters are as follows:
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• 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 synthetic metallic sound. The Ring Modulator also features a MIDI controlled envelope generator for sweeping the internal 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 Frequency value. The Amount is not affected by velocity.
• MIDI Trig EG Decay
This sets the MIDI controlled envelope decay time.
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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 electromechanical 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 patterns 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:
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• 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 according 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 section 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 described in the chapter
.
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”
.
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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
.
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 bottom 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 processing 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.
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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
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
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
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Chapter 11
Redrum Drum
Computer
Introduction
At first glance, Redrum looks styled after pattern-based drum machines, like the legendary Roland 808/909 units. Indeed, 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. Complete drum kits can be saved as Redrum Patches, allowing you to mix and match drum sounds and make up custom kits with ease.
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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 references to the used drum samples (but not the actual drum samples themselves). Switching patches is the same as selecting 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.
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Using patches
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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:
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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 device context menu, or click the folder button in the patch section on the device panel.
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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.
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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.
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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:
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By clicking the Trigger (arrow) button at the top of each drum sound channel.
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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.
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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.
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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
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.
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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 samples for all drum sound channels, and sets all parameters to their default values.
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Programming patterns
Pattern basics
Redrum contains a built-in pattern sequencer. Unlike the main sequencer in Reason, the Redrum sequencer repeatedly 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:
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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.
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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).
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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.
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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.
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If you are running Redrum separately and start playback of the main sequencer, the pattern device will automatically restart in sync with the sequencer.
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Pattern changes can be controlled by pattern automation in the main sequencer.
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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.
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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.
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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.
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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.
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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.
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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 previously 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:
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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.
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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, although 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.
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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.
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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.
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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 pattern sequencer.
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Pattern Shuffle
Shuffle is a rhythmic feature, that gives the music a more or less pronounced swing feel. It works by delaying all sixteenth 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
.
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.
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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.
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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.
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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):
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Function
Shift Pattern Left/Right
Shift Drum Left/Right
Randomize Pattern
Randomize Drum
Alter Pattern
Alter Drum
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Description
These functions move all notes in the pattern one step to the left or 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.
Creates a random pattern. Random patterns can be great starting points and help you get new ideas.
Creates a random pattern for the selected drum sound only - the notes for the other drum sound channels are unaffected.
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.
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.
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:
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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:
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Be sure to disable the Enable Pattern Section function on the Redrum panel afterwards, to avoid “doubled notes” during playback.
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Redrum parameters
Drum sound settings
Redrum features ten drum sound channels that can each be loaded with a sample. Although they are basically similar, 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.
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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.
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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 (provided that these inputs aren’t already in use).
This feature allows you to add effects to independent drum sounds in the Redrum.
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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.
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Note that there must be send effects connected to the AUX Sends and Returns in the mixer for this to work.
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Also note that if Redrum is soloed in the Mixer the effect sends will be muted.
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Another way to add independent effects to drum sounds is to use the independent drum outputs.
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Pan
Sets the Pan (stereo position) for the channel.
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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.
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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.
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A negative value inverts this relationship, so that the volume decreases with higher velocity values.
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If the Vel knob is set to zero (middle position), the sound will play at a constant volume, regardless of the velocity.
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:
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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.
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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.
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• 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 repeats 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.
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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.
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The Rate knob determines the bend time - the higher the value, the slower the bend.
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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.
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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.
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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.
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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.
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When Start Velocity is set to any other value than zero, the LED above the knob lights up.
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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.
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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.
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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
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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.
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Connections
On the back of the Redrum you will find the following connections:
For each drum sound channel:
|
Connection
Audio Outputs
Gate Out
Gate In
Pitch CV In
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Description
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.
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.
Allows you to trigger the sound from another CV/Gate device. All settings apply, just as when playing the drum sound conventionally.
Lets you control the pitch of the drum sound from another CV device.
Other
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Connection
Send Out 1-2
Stereo Out
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Description
Outputs for the send signals controlled with the S1 and S2 knobs.
This is the master stereo output, outputting a mix of all drum sounds (except those for which you use individual outputs).
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Chapter 12
Dr. Octo Rex
Loop Player
Introduction
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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 separate 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.
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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).
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• 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 format (.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
.
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).
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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.
DR. OCTO REX LOOP PLAYER
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You can also play the loop once via MIDI, by using the D0 key.
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To check out the loop(s) together with other device sequencer data and patterns already recorded, click the sequencer 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 example.
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.
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DR. OCTO REX LOOP PLAYER
The picture shows what keys should be pressed to select and play Loop Slot and to stop loop playback:
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
).
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 section, 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 consecutive 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:
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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
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 individual 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.
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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.
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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 below the waveform display. If you want to edit a single parameter for several slices at once, a more convenient way
|
Parameter
Pitch
Pan
Level
Decay
Rev
F.Freq
Alt
Output
|
Description
Allows you to transpose each individual slice in semitone steps, over a range of more than eight octaves.
The stereo position of each slice.
The volume of each slice. The default level is 100.
Allows you to shorten individual slices.
Allows you to play back individual slices reversed (backwards)
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”
.
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
.
Allows you to assign individual slices to separate audio outputs (1-8). If the REX loop is in stereo, there is also an option to select individual output pairs (1+2, 3+4, 5+6 or 7+8) for individual slices.
!
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 assign 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.
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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 Edit Slice 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 Output.
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 values 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.
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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.
and
.
Trig Next Loop
.
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 recorded 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.
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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
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
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 recorded 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).
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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 parameters 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 Automation 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.
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 sequencer:
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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
.
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:
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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
). 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
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
F. Freq
F. Res
F. Decay
|
Description
This sets modulation wheel control of the filter frequency parameter. A positive value will raise the frequency if the wheel is pushed forward. Negative values invert this relationship.
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.
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:
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DR. OCTO REX LOOP PLAYER
• After having recorded MIDI notes in the main sequencer, you can edit the velocity values in the in the sequencer.
• 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 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:
|
Parameter
F. Env
F. Decay
Amp
|
Description
This sets velocity control for the Filter Envelope Amount parameter. A positive value will increase the envelope amount with higher velocity values. Negative values invert this relationship.
This sets velocity control for the Filter Envelope Decay parameter. A positive value will increase the Decay time with higher velocity values. Negative values invert this relationship.
This let’s you velocity control the overall volume of the file. If a positive value is set, the volume will increase 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 multimode 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 filter.
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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 rolloff 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 maximum, all frequencies in the waveform are heard. Gradually changing the Filter Frequency produces the classic synthesizer filter “sweep” sound.
) 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 becomes 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 frequencies.
• 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 envelopes 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
in the Subtractor chapter for a description of the basic envelope parameters.
Amplitude Envelope
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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 frequency. 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 produce 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
Triangle
Inverted
Sawtooth
|
Description
This is a smooth waveform, suitable for normal vibrato.
This produces a “ramp up” cycle. If set to control pitch (frequency), the pitch would sweep up to a set point (governed by the Amount setting), after which the cycle immediately starts over.
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DR. OCTO REX LOOP PLAYER
|
Waveform
Sawtooth
Square
Random
Soft Random
|
Description
This produces a “ramp down” cycle, the same as above but inverted.
This produces cycles that abruptly changes between two values, usable for trills etc.
Produces random stepped modulation to the destination. Some vintage analog synths called this feature “sample & hold”.
The same as above, but with smooth modulation.
Destination
The available LFO Destinations are as follows:
|
Destination
Osc
Filter
Pan
|
Description
Selecting this makes LFO control the pitch (frequency) of the REX file.
Selecting this makes the LFO control the filter frequency.
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 maximum range is 24 semitones (=up/down 2 Octaves).
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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 polyphony 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 polyphony 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 algorithm. 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
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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
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 envelope 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 individual 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 autorouted to the first available outputs in the I/O device.
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DR. OCTO REX LOOP PLAYER
Chapter 13
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 control. 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.
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EUROPA SHAPESHIFTING SYNTHESIZER
Panel overview
The Europa front panel contains the following sections:
1 2
3
4
8 9
11
5
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.
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EUROPA SHAPESHIFTING SYNTHESIZER
6
12
10
7
Signal flow
The picture below shows the basic signal flow in Europa:
Engine III
Engine II
Engine I
Amt Mod Amt Mod
Modifier 1 Modifier 2
KBD Mod Vel Count Blend
Pan
Oscillator Spectral Filter
Pitch KBD Shape Freq Reso
Harmonics Unison
Level
Modify Amt Detune Width
KBD Mod Vel Pan
Filter
Amp
Envelope
Multi FX
Out
Volume
Per Voice
Drive Freq Reso Gain Vel
: audio signal
: 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 oscillators 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 effects.
• 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 destination parameters via the Modulation Bus section.
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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
and
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
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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
.
• 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. Portamento 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 commence. 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
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
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 parameter in any of the Modulation Source boxes in the different sections. Or use it as Source parameter in the Modulation
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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).
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EUROPA SHAPESHIFTING SYNTHESIZER
176
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
•
.
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 sawtooth 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 towards a 100% duty cycle pule wave (silence) at Shape=100%.
q
Modulate the Shape parameter from an LFO to achieve PWM, see
• 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
.
• 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
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 > 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
above.
• Noise > Freeze Noise
This signal produces a range of noises, from tonal noise up to almost white noise, by amplitude modulating the signal’s partials with noise.
• Wave Tables >
The Wave Tables sub-menu contains a selection of very useful wave tables. Each wave table features eight waveforms that you could crossfade between with the Shape knob.
• User Wave/User Wave Smooth
). 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 example 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 settings.
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.
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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
178
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.
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 headroom. 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 headroom. 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 frequency with the Amount knob.
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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
• 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
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
180
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.
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
.
•
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.
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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 (multipeak+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 control the cutoff/center frequency with the Freq knob and the resonance with the Reso knob. See
• 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
). Create a positive ramp (inverted 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
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EUROPA SHAPESHIFTING SYNTHESIZER
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% attenuate 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
!
• 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 algorithm 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
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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EUROPA SHAPESHIFTING SYNTHESIZER
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 below) now controls the phases of the signal copies, instead of the detuning. This is great for creating wide stereo sounds without lots of detuning.
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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EUROPA SHAPESHIFTING SYNTHESIZER
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 duplicates 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
Wave/User Wave Smooth” ) and/or in the Spectral Filter (see
). In the Mixer you can mix and pan the signals 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 buttons 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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EUROPA SHAPESHIFTING SYNTHESIZER
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
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
) 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 envelope” for a Sound Engine”
.
The picture below shows the various stages of the ADSR envelope:
Level
Gain
(level)
Sustain
(level)
Time
Attack
(time)
Key Down
The ADSR envelope stages.
Decay
(time)
Release
(time)
Key Up
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 parameter 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 gradually 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.
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, turning 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 exact same time positions. Otherwise, only the closest point will be changed. Also, any inclining/declining segment 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 “resonance”.
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 parameter(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
At the top of the Effects section are six Effect buttons. Click any of these to bring up the control panel for the corresponding 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 sideways 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
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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.
EUROPA SHAPESHIFTING SYNTHESIZER
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 values 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 affected, signals below it will not. In practice, this means that the lower the Threshold setting, the more the compression 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-assigned 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
Velocity
|
Description
This applies modulation according to the Keyboard Velocity values (how hard or soft you strike the MIDI keyboard keys).
This allows you to modulate parameters from LFO 1, LFO 2 and LFO 3 respectively.
LFOs (LFO 1, LFO 2 and LFO 3)
Envelopes (Amp Envelope, Envelope 1,
Envelope 2, Envelope 3, Envelope 4,
Envelope 3 * Envelope 4, Envelope 3 *
LFO 3)
This allows you to modulate parameters from any of the Envelopes.
As a special feature you can also modulate parameters from the multiplied signal of Envelope 3 and
Envelope 4, as well as from the multiplied signal of Envelope 3 and LFO 3.
Mod Wheel This allows you to modulate parameters from the Mod Wheel.
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EUROPA SHAPESHIFTING SYNTHESIZER
|
Parameter
MW Latched
Pitch Wheel
Breath
Expression
Aftertouch
Sustain
Key
Random
Key In Octave
Noise
Polyphony
Last Velocity
CV Input 1/2/3/4
CV Input 1/2/3/4 Latched
|
Description
This allows you to modulate parameters based on the current Mod Wheel value at a given Note On.
This allows you to modulate parameters from the Pitch Bend control.
This allows you to modulate parameters from the Breath performance controller
This allows you to modulate parameters from the Expression performance controller
This allows you to modulate parameters from Keyboard Aftertouch (channel aftertouch)
This allows you to modulate parameters from a connected sustain pedal. Note that continuous sustain data (0-127) is supported - not just on/off.
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.
This sends out a random value each time a new note is played.
This allows you to modulate parameters based on 12 separate note values (within each octave).
This allows you to modulate parameters from white noise.
This allows you to modulate parameters based on the number of playing voices at a given time.
This applies modulation according to the latest Keyboard Velocity value (how hard or soft you hit the latest MIDI keyboard key).
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.
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 dragging vertically in the corresponding Amount box.
Note that the Amount range is +/-100. This means that the Amount value can exceed the modulated parameter’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.
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EUROPA SHAPESHIFTING SYNTHESIZER
200
The following parameters can be used as modulation Destinations:
|
Parameter
Engine: Pitch
Engine: Shape
Engine: Mod 1 Amount
Engine: Mod 2 Amount
Engine: Filter Freq
Engine: Filter Res
Engine: Harmonics Pos
Engine: Harmonics Amount
Engine: Unison Count
Engine: Unison Detune
Engine: Unison Blend
Engine: Unison Spread
Mixer: Level
Mixer: Pan
Filter: Drive
Filter: Freq
Filter: Reso
Amplifier: Gain
Amplifier: Pan
Amp Envelope: Attack
Amp Envelope: Decay
Amp Envelope: Sustain
Amp Envelope: Release
LFOs: Delay
LFOs: LFO Rate
Envelopes: Env Rate
Portamento
CV Outputs: CV1/2/3/4 Out
Reverb: Decay
Reverb: Amount
Delay: Time
Delay: Feedback
Delay: Amount
Delay: Pan
Dist: Drive
Dist: Tone
Dist: Amount
Compressor: Release
Compressor: Ratio
Chorus/Flanger/Phaser: Frequency
Chorus/Flanger/Phaser: Amount
Par EQ: Frequency
Par EQ: Gain
|
Description
This affects the (full range) pitch of the Oscillator.
This affects the Shape parameter in the Oscillator section.
This affects the Modifier 1 Amount parameter in the Sound Engine.
This affects the Modifier 2 Amount parameter in the Sound Engine.
This affects the Spectral Filter Frequency parameter in the Sound Engine.
This affects the Spectral Filter Resonance parameter in the Sound Engine.
This affects the Harmonics Pos parameter in the Sound Engine.
This affects the Harmonics Amount parameter in the Sound Engine.
This affects the Unison Count parameter in the Sound Engine.
This affects the Unison Detune parameter in the Sound Engine.
This affects the Unison Blend parameter in the Sound Engine.
This affects the Unison Spread parameter in the Sound Engine.
This affects the Sound Engine Level in the Mixer section.
This affects the Sound Engine Pan in the Mixer section
This affects the Drive parameter in the Filter section.
This affects the Frequency parameter in the Filter section.
This affects the Resonance parameter in the Filter section.
This affects the Gain parameter of the Amplifier section.
This affects the Pan parameter of the Amplifier section.
This affects the Attack time of the Envelope in the Amplifier section.
This affects the Decay time of the Envelope in the Amplifier section.
This affects the Sustain level of the Envelope in the Amplifier section.
This affects the Release time of the Envelope in the Amplifier section.
This affects the LFO Delay parameters.
This affects the LFO Rate parameters.
This affects the Envelope Rate parameters.
This affects the Portamento Time parameter.
This sends out the source modulation value(s) on the CV1/2/3/4 Output on the rear panel.
This affects the Decay parameter in the Reverb effect.
This affects the Amount parameter in the Reverb effect.
This affects the Time parameter in the Delay effect.
This affects the FB parameter in the Delay effect.
This affects the Amount parameter in the Delay effect.
This affects the Pan parameter in the Delay effect.
This affects the Drive parameter in the Dist effect.
This affects the Tone parameter in the Dist effect.
This affects the Amount parameter in the Dist effect.
This affects the Release parameter in the Compressor effect.
This affects the Ratio parameter in the Compressor effect.
This affects the center frequency of the Chorus/Flanger/Phaser effects.
This affects the Amount parameter of the Chorus/Flanger/Phaser effects.
This affects the Freq parameter in the EQ effect.
This affects the Gain parameter in the EQ effect.
Modulation Bus Destination parameters.
EUROPA SHAPESHIFTING SYNTHESIZER
6. Set the Amount for the second Destination (if desired) by turning the corresponding Amount knob, or by clicking 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 parameters.”
.
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 Destination 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 corresponding Scale box.
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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.
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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.
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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.
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Chapter 14
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 various 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 features 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.
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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 snippets of audio data - grains - “extracted” from an audio sample. The grains could be of a selectable length and spacing, 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.
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Panel overview
The Grain front panel contains the following sections:
1 2
3
4 5
6 7
9
8
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
and
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 continue at the current sample playback position (and not restart playback).
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Porta
Portamento makes note pitches glide from previous notes to new ones, at the time set with the Time knob. Portamento 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 commence. 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
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
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 section” .
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
modes instead of lowering the Voices parameter to 1.
Master Volume
This is the main stereo output volume control.
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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 marker
Sample End marker
Waveform display
Loading
!
D
Load a sample using drag & and drop, or by clicking the Browse sample button, or by using the Up/Down buttons 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, regardless 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
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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 entirety 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-
), 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 according 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 Reason sequencer. Use a straight ramp (up) in Envelope 1, activate Beat Sync and set the sync to a suitable bar length, see
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
above.
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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 reanalyzed.
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 Algorithms 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 selected algorithm.
Playback Algorithm selector Pitch controls
D
Click the Playback Algorithm selector and choose one of the following four algorithms:
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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. “4” is the most accurate detection. However, it’s also slower since it also detects lowfrequency 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
flat.
• Formant
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 instead, 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
).
• 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.
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Grain Oscillator
The Grain Oscillator plays back a mix of two very short grains of the original sample. The grain playback rate corresponds 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 unaffected 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.
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Long Grains
The Long Grains playback algorithm plays back fairly long grains of the original sample. This means that it’s the original pitch of the sound (Root Key) that affects the pitch, along with the Pitch settings (see
).
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 effects!
• 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
) have no effect. The sample will play back at the same pitch regardless of which note you play.
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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%.
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The Oscillator section
The Oscillator can be used in addition to the sample playback. The oscillator features a number of selectable waveforms 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 buttons.
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.
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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 keyboard 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 Playback Algorithms and Oscillator sections equally. The picture below shows the various stages of the ADSR envelope:
Level
Gain
(level)
Sustain
(level)
Time
Attack
(time)
Key Down
The ADSR envelope stages.
Decay
(time)
Release
(time)
Key Up
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 parameter 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 gradually 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.
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.
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.
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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:
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
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, turning 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 exact same time positions. Otherwise, only the closest point will be changed. Also, any inclining/declining segment 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 parameter(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
At the top of the Effects section are six Effect buttons. Click any of these to bring up the control panel for the corresponding 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:
225
Moving the Chorus effect to another position in the effects chain.
You can reorder the effects at any time.
GRAIN SAMPLE MANIPULATOR
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.
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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 values 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 affected, signals below it will not. In practice, this means that the lower the Threshold setting, the more the compression 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.
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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-assigned 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
Velocity
|
Description
This applies modulation according to the Keyboard Velocity values (how hard or soft you strike the MIDI keyboard keys).
This allows you to modulate parameters from LFO 1, LFO 2 and LFO 3 respectively.
LFOs (LFO 1, LFO 2 and LFO 3)
Envelopes (Amp Envelope, Envelope 1,
Envelope 2, Envelope 3, Envelope 4,
Envelope 3 * Envelope 4, Envelope 3 *
LFO 3)
This allows you to modulate parameters from any of the Envelopes.
As a special feature you can also modulate parameters from the multiplied signal of Envelope 3 and
Envelope 4, as well as from the multiplied signal of Envelope 3 and LFO 3.
Mod Wheel
MW Latched
Pitch Wheel
Breath
This allows you to modulate parameters from the Mod Wheel.
This allows you to modulate parameters based on the current Mod Wheel value at a given Note On.
This allows you to modulate parameters from the Pitch Bend control.
This allows you to modulate parameters from the Breath performance controller
Expression
Aftertouch
Sustain
Key
This allows you to modulate parameters from the Expression performance controller
This allows you to modulate parameters from Keyboard Aftertouch (channel aftertouch)
This allows you to modulate parameters from a connected sustain pedal. Note that continuous sustain data (0-127) is supported - not just on/off.
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
Key In Octave
Noise
Polyphony
Last Velocity
Sample Pitch Curve
This sends out a random value each time a new note is played.
This allows you to modulate parameters based on 12 separate note values (within each octave).
This allows you to modulate parameters from white noise.
This allows you to modulate parameters based on the number of playing voices at a given time.
This applies modulation according to the latest Keyboard Velocity value (how hard or soft you hit the latest MIDI keyboard key).
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.
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|
Parameter
Display Y Position
Display Mouse Gate
CV Input 1/2/3/4
CV Input 1/2/3/4 Latched
|
Description
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.
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.
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.
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 dragging vertically in the corresponding Amount box.
Note that the Amount range is +/-100. This means that the Amount value can exceed the modulated parameter’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
Position
Speed
Jitter
Start Position
End Position
Pitch
Octave
Formant
|
Description
This affects the sample “playhead” position in the original sample.
This affects the Speed control in the sample window.
This affects the Jitter control in the sample window.
This affects the Sample Start marker position in the original sample.
This affects the Sample End marker position in the original sample.
This affects the pitch of the original sample.
This affects the Oct control in the Playback Algorithm section.
This affects the Formant Form control in the Playback Algorithm section (if applicable).
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231
|
Parameter
Sample Level
Grain: Length
Grain: Rate/Spacing
Grain: X-Fade
Grain: Pan Spread
Grain: Pitch Jitter
Spectral Grain: Harm Snap
Spectral Grain: Harm Filter
Spectral Grain: Curve Amount
Oscillator: Modulation
Oscillator: Octave
Oscillator: Pitch
Oscillator: Level
Filter: Freq
Filter: Reso
Amplifier: Gain
Amplifier: Pan
Amp Envelope: Attack
Amp Envelope: Decay
Amp Envelope: Sustain
Amp Envelope: Release
LFO’s: LFO1/2/3 Delay
LFO’s: LFO1/2/3 Rate
Envelopes: Env1/2/3/4 Rate
Portamento
CV Outputs: CV1/2/3/4 Out
Reverb: Decay
Reverb: Amount
Delay: Time
Delay: Feedback
Delay: Amount
Delay: Pan
Dist: Drive
Dist: Tone
Dist: Amount
Compressor: Release
Compressor: Ratio
Chorus/Flanger/Phaser: Frequency
Chorus/Flanger/Phaser: Amount
Par EQ: Frequency
Par EQ: Gain
Note Trig
|
Description
This affects the Sample Level control of the Playback Algorithm section.
This affects the Grain Length parameter on the Grain Oscillator and Long Grains playback algorithms.
This affects the Grain Spacing parameter on the Grain Oscillator and the Rate parameter on the Long
Grains playback algorithms.
This affects the X-Fade parameter on the Long Grains playback algorithm.
This affects the Pan Spread parameter on the Grain Oscillator and Long Grains playback algorithms
This affects the Pitch Jitter parameter on the Grain Oscillator and Long Grains playback algorithms
This affects the Snap parameter on the Spectral Grains playback algorithm.
This affects the Filter parameter on the Spectral Grains playback algorithm.
This affects the Amount parameter on the Spectral Grains playback algorithm.
This affects the Mod parameter on the Oscillator.
This affects the Oct parameter on the Oscillator.
This affects the (full range) pitch of the Oscillator.
This affects the Osc Level parameter of the Oscillator section.
This affects the Frequency parameter in the Filter section.
This affects the Resonance parameter in the Filter section.
This affects the Gain parameter of the Amplifier section.
This affects the Pan parameter of the Amplifier section.
This affects the Attack time of the Envelope in the Amplifier section.
This affects the Decay time of the Envelope in the Amplifier section.
This affects the Sustain level of the Envelope in the Amplifier section.
This affects the Release time of the Envelope in the Amplifier section.
This affects the LFO1/2/3 Delay parameter.
This affects the LFO1/2/3 Rate parameter.
This affects the Envelope 1/2/3/4 Rate parameter.
This affects the Portamento Time parameter.
This sends out the source modulation value(s) on the CV1/2/3/4 Output on the rear panel.
This affects the Decay parameter in the Reverb effect.
This affects the Amount parameter in the Reverb effect.
This affects the Time parameter in the Delay effect.
This affects the FB parameter in the Delay effect.
This affects the Amount parameter in the Delay effect.
This affects the Pan parameter in the Delay effect.
This affects the Drive parameter in the Dist effect.
This affects the Tone parameter in the Dist effect.
This affects the Amount parameter in the Dist effect.
This affects the Release parameter in the Compressor effect.
This affects the Ratio parameter in the Compressor effect.
This affects the center frequency of the Chorus/Flanger/Phaser effects.
This affects the Amount parameter of the Chorus/Flanger/Phaser effects.
This affects the Freq parameter in the EQ effect.
This affects the Gain parameter in the EQ effect.
This is mainly intended for use with the Display > Mouse Gate or Display > Y-Position source parameters, 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.
GRAIN SAMPLE MANIPULATOR
6. Set the Amount for the second Destination (if desired) by turning the corresponding Amount knob, or by clicking 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 parameters.”
.
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 Destination 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 corresponding 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.
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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.
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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 basic 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 automation 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 Modulation 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 appear:
• 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
.
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Chapter 15
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 !
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Panel overview
The Mimic front panel contains the following sections:
1 2
4
6
5
3
7
8 9
12
10
13
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.
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11
14
Signal flow
Slots (x8)
Pitch or
Slice
(x8)
Multi Slot or
Multi Pitch
(x8)
(x1)
Master Volume
(x8)
Separate Out (x8)
Master Out
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” .
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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
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 locators). The sample can then be played chromatically pitched over the entire keyboard range.
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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 automatically. 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), drummachine 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 4
Slot 2
Slot 4
Slot 2
Slot 4
Slot 2
Slot 4
Slot 2
Slot 4
Slot 2
Slot 4
Slot 3
Slot 1
Slot 3
Slot 1
Slot 3
Slot 1
Multi Slot Mode with samples loaded in Slots 1-4.
q
Multi Slot Mode works great with pad controllers.
Slot 3
Slot 1
Slot 3
Slot 1
Slot 3
Slot 1 Slot 1
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Multi Pitch Mode
C1
C1
C2
C2
C3
C3
C4
C4
C5
C5
C6
C6
C1 C2 C3 C4 C5 C6
C1 C2 C3 C4 C5
One MIDI keyboard playing four Slots simultaneously (layered)
C6
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 playback (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 keyboard 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 settings.
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. Portamento 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 commence. 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
, 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
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
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.
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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, depending on which Slot Mode you have selected (see
Start marker indicator
Sample range markers
End marker indicator
Root Key section
Sample Overview
Slice markers
End marker
Start marker Waveform display
Position marker
Sample Load/Sampling section
A sample loaded in a Slot with Slice Mode selected.
Note/range indicator
Loading samples
D
Load a sample using drag & and drop, or by clicking the Browse sample button, or by using the Up/Down buttons 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.
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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 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.
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.
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MIMIC CREATIVE SAMPLER
Previewing samples
D
Preview the sample by holding down [Option]/[Alt] and clicking at the desired position in the Waveform Display.
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 markers” .
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/dragging sideways in the waveform display:
!
If you are playing back the sample reversed (see
) the looped region originates from the Start Marker instead 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 between Slice Markers).
Note that this function is only useful in when modulating the Start Position (see above) in Pitch Mode, Multi
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
) is currently selected, the sonic result will vary. If you have se-
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 gradually 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
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
), 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 crossfaded.
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.
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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 overlap 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
).
q
Even though slices are the core of Slice mode, they are available in the other modes too. They can be used for
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 example, 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).
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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
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
Amplitude
Q
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 difference between “Comb +” (see below) 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.
• 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 attenuation 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 keyboard 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
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
(time)
Key Down
The ADSR envelope stages.
Decay
(time)
Release
(time)
Key Up
A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the maximum frequency value (Filter Envelope) or Gain level (Amp Envelope). How long this should take, depends on the Attack 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 parameter 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 Frequency/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
).
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.
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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
). You could then route the signals to external effect devices and then further to a separate 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.
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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
).
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-
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 piano, 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 recording, 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 Automation 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 16
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 designs 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 modulation 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
.
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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.
.
• 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
.
• 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.
.
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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
Polyphony
Release
Polyphony
Mono Legato
|
Description
This setting determines the number of voices that you can play simultaneously when Polyphonic mode is selected. The maximum number of voices is 32.
This governs the number of voices that are allowed to naturally decay/ring out (in the release phase of 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 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
Polyphonic
Mono Retrig is also monophonic and this mode means that when you press a key the envelopes are always retriggered.
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
).
• Or both.
The section also has a standard Note On indicator.
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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.
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Note that the key note function can only switch from off to on, not the other way around, so make sure the button 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 keyboard.
• 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.
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Using the Programmer
The Programmer contains the main synth parameters.
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To show the Programmer panel, click the “Show Programmer” button on the Controller panel.
The Programmer appears below the Controller panel.
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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.
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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.
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The Global section contains a second LFO, a Global Envelope, a third open Filter slot and Chorus and Delay effects.
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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 section”
). 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 basics 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 oscillator 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
Selecting oscillator type.
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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 currently nothing is activated in the Global section the signal passes through unprocessed.
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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
.
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 select 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 perfectly “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”
.
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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 assignments:
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The Amp Envelope and the Filter Envelope control the volume level and frequency of the Filters (1 & 2), respectively.
The amount of filter envelope control is controllable by using the “Env” parameter in each Filter section.
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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 parameters. 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.
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The Arrow button in the top left corner of each slot opens a pop-up menu where an oscillator type can be selected for the corresponding slot.
There are six Oscillator types available:
• Analog
• Wavetable
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• 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 parameters that apply to all oscillator types. These are:
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Octave (OCT) knob - this changes the pitch of the oscillator in octave steps.
The range is ten octaves.
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The Semi knob changes the pitch of the oscillator in semi-tone steps.
The range is 12 semitone steps (1 octave).
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The Tune knob fine tunes the pitch of the oscillator in cent steps.
The range is +/- 50 cents (down or up half a semitone).
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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.
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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.
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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 parameters are left unchanged.
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Oscillators can be synced - see
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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 Modulation) which is a standard feature in most vintage analog synths. q
For a perfect square wave, set pulse width (PW) to 64.
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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 timbre variations.
The parameters are as follows:
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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.
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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.
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There are 32 wavetables that can be selected using the up/down buttons or by clicking in the Wavetable display.
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.
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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.
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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
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• 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 altogether. 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.
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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.
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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.
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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.
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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.
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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 produce 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.
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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 detune waveforms as the names imply between zero to full Amount settings. “Linear” will change the amount of detune 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:
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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.
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The Waveform selector button in the bottom left corner is used to set the Oscillator mode.
The following modes are available:’
|
Mode
Band
S/H
Static
Color
White
|
Description
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 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.
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.
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.
This produces pure white noise, where all frequencies have equal energy. There is no associated Oscillator parameter 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 associated 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.
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Switch Oscillator Sync on or off by activating the Sync buttons to the left of Oscillator slots 2 and 3.
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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.
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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 difference between the two signals. This can be used for creating complex, enharmonic sounds.
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The amount of AM is set using the slider to the left of the Oscillator 1 slot.
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Mix section
The Mix section allows you to adjust the levels and the relative balance of the three oscillators.
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The two sliders controls the output levels of oscillators 1-2 and oscillator 3, respectively.
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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 oscillators with e.g. an LFO. Note that the oscillators have to be connected to the filter(s) via the numbered routing buttons 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” ).
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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:
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Filters are pre-wired to the Filter Envelope (see “Filter Envelope” ).
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• 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).
passage.
Common parameters
As with the open oscillator slots, there are certain parameters which are common for all filter types.
These are as follows:
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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
.
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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.
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The “ENV” parameter sets how much the filter frequency responds to the Filter Envelope.
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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.
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The “INV” button inverts how the filter frequency responds to Envelope settings.
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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.
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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 renowned 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.
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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.
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Note that “Self Osc” (see below) must be activated for the shaper to operate.
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This filter can self-oscillate and will produce a playable note 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 frequency 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 filter.
The filter modes are as follows:
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LP 12 (12 dB lowpass)
Lowpass filters let low frequencies through and cut off high frequencies. This filter type has a 12dB/Octave slope.
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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.
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HP 12 (12 dB highpass)
Highpass filters let high frequencies pass and cut off low frequencies. This filter type has a 12dB/Octave slope.
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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 highpass. 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.
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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 frequency tracks the keyboard, turned fully clockwise will produce 12 semitones/octave tracking.
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Comb filter
The Comb filter can add subtle pitch variations and phasing-like effects to sounds.
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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.
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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.
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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.
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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.
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The ENV-VEL-KBD knobs affect the “X” parameter.
The parameter can be CV controlled.
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The “Gender” parameter changes the basic timbre of the vowel generation between male (low Gender settings) and female (high Gender settings) voice characteristics.
Gender can also be CV controlled.
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Shaper
Waveshaping is a synthesis method for transforming sounds by altering the waveform shape, thereby introducing various types of distortion. The Shaper can radically transform the sound or just add a little warmth, depending on the mode and other settings.
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The Shaper input is taken from the Filter 1 output.
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The Shaper is activated with the button in the top left corner of the section.
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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.
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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 distortion 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).
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The Gain knob controls the level and the Velocity knob controls the Gain modulation, i.e. how much velocity affects the level - positive values means that you get higher level the faster you strike a key.
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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.
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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.
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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.
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You select a LFO waveform by using the spin controls beside the waveform display, or by clicking in the display and moving the mouse up or down.
The following parameters are available for LFO 1:
|
Parameter
Rate
Waveform
Delay
KBD Follow
Key Sync
Tempo Sync
|
Description
This sets the frequency or rate of the LFO.
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.
This introduces a delay before the LFO modulation onset after a note is played. Turn clockwise for longer delay.
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.
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 modulation may start anywhere in the LFO waveform cycle. If Key Sync is on, the LFO cycles are reset for each note played.
If this is on, the Rate will be synced to the sequencer tempo.
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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 described separately - see
.
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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.
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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 envelope) there will be no output from oscillators or any external audio source routed to the Voice section.
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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.
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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.
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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.
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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).
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Amp Envelope
The Amp Envelope is also a standard ADSR envelope.
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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.
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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.
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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
Delay
Loop
Tempo Sync
Gate Trigger
|
Description
This can set a delay before the onset of the envelope.
If this is activated, the envelope phase from Delay to Decay will continuously loop.
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).
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.
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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.
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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.
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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 sequencer tempo.
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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.
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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
Delay
Loop
Hold
Tempo Sync
Gate Trigger
|
Description
This can set a delay before the onset of the envelope.
If this is activated, the envelope phase from Delay to Decay will continuously loop.
This allows you to set a “hold” phase before the Decay.
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).
The “Gate Trig” button can be used to switch off the envelope triggering from notes and allow the envelope to be triggered by some other parameter. This button is normally activated.
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LFO 2
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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 parameters for legato notes.
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You select a LFO waveform by using the spin controls beside the waveform display, or by clicking in the display and moving the mouse up or down.
The following parameters are available for LFO 2:
|
Parameter
Rate
Waveform
Delay
Key Sync
Tempo Sync
|
Description
This sets the frequency or rate of the LFO.
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.
This introduces a delay before the LFO modulation onset after a note is played. Turn clockwise for longer delay.
If Key Sync is off, the LFO cycle is free running, meaning that when you play a note the modulation may start anywhere in the LFO waveform cycle. If Key Sync is on, the LFO cycle is reset for each note played.
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 Destination modulation assigned.
2. Click in the top row of the leftmost Source column.
A pop-up menu appears listing all available Source modulation parameters.
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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 section destinations, and the lower half contains Global section destinations, as well as the Step Sequencer, CV and Audio 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%).
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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.
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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 parameters, 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.
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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:
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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.
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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 parameter that affects the relative modulation Amount for both Destinations.
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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.
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Modulation Sources - Voice section
The following parameters can be used as Voice section modulation Sources:
|
Parameter
Voice Key
Osc 1/2/3
Filter 1/2
Shaper
Amp
LFO 1
Filter Envelope
Amp Envelope
Mod Envelope
|
Description
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.
This allows you to route the audio output from the oscillators to a destination.
This is the audio output of the filters. All filter parameters affect the destination.
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.
This is the audio output of the Amp Gain section.
This allows you to modulate parameters with LFO 1.
This allows you to modulate parameters with the Filter Envelope.
This allows you to modulate parameters with the Amp Envelope.
This allows you to modulate parameters with the Mod Envelope.
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Modulation Sources - Global
The following parameters can be used as Global section modulation Sources:
|
Parameter
Global Envelope
Voice Mixer
Last Key
MIDI Key
LFO 2
Performance parameters
Modifiers
Sustain Pedal
Polyphony
Step Sequencer
CV Inputs 1-4
Audio Inputs 1-4
|
Description
This allows you to modulate parameters using the Global Envelope.
This allows you to modulate parameters using the Left and Right Mixer inputs.
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.
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.
This allows you to modulate parameters with LFO 2.
On this sub-menu you can assign the one of the standard Performance controllers to modulate/scale parameters; Mod Wheel/Pitch Bend/Breath/AfterTouch/Expression.
This is where you assign parameters and functions to be controlled with the virtual 2 Rotary and 2 Button controls on the Controller panel.
This allows you to assign the Sustain Pedal as a modulation source.
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.
This allows you to apply modulation according to the settings for each step in the Step Sequencer.
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.
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.
These are Audio inputs on the back panel which allows you to connect external audio signals
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Modulation Destinations - Voice section
The following parameters can be used as Voice section modulation Destinations:
|
Parameter
Osc 1
Osc 2/ Osc 3
Filter 1/ Filter 2
Shaper Drive
Amp
Mix
|
Description
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)” .
Oscillator slots 2 and 3 have the same Destination parameters as Osc 1, except that there is no AM.
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 oscillator 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 oscillator 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).
This will control the Shaper Drive parameter.
The Amp section has three destinations on the sub-menu:
• Input - this allows you to connect a source (e.g. an oscillator or an external 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 differently for each voice you play.
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 oscillator 1 and 2, e.g. to sweep from one oscillator to the other.
• Osc 3 Level - this controls the level of oscillator 3.
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|
Parameter
Filter Envelope
Amp Envelope
Mod Envelope
LFO 1 Rate
|
Description
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.
This has the same destination parameters as the Filter Envelope.
This has the same destination parameters as the Filter Envelope.
This allows you to control the LFO 1 Rate parameter.
Modulation Destinations - Global
The following Global modulation destinations are available:
|
Parameter
Portamento
LFO 2 Rate
Global Envelope
Filter 3
Chorus
|
Description
This allows you to control the Portamento time parameter.
This allows you to control the LFO 2 Rate parameter.
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.
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).
The Chorus effect has the following destinations:
• DryWet balance
• Delay (time)
• ModRate
• ModAmount
• Feedback
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|
Parameter
Delay
Step Sequencer
CV Output 1-4
Audio Output 1-4
|
Description
The Delay effect has the following destinations:
• DryWet balance
• Time
• ModRate
• ModAmount
• Feedback
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 response.
• Gate Length - this enables control over the Step Sequencer Gate
Length response.
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.
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.
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Routing audio to Global destinations does not require a gate trigger and stereo inputs are provided.
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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.
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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:
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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).
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Each step button has a knob above it, which is used to set values for the corresponding step.
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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.
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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:
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Repeat mode - this will repeat the sequence continuously.
Click the Run button again or use the Transport to stop.
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1 Shot mode - this will play the sequence once then stop.
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Step mode - the Run button steps the sequencer forward one step at a time.
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Off - the step sequencer is inactive.
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Setting the direction
The Direction parameter is used to set the direction of the step sequence. The following options are available:
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Forward - plays the sequence from the first step to the last.
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Reverse - plays the sequence from the last step to the first.
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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.
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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.
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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.
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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 counterclockwise to lower the pitch.
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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.
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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.
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This is simply done by pressing one or several step buttons so they go dark.
Dark steps will be rests.
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Note that the Step Duration value still “counts” for rests.
Setting the number of steps
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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 selected step.
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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:
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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.
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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.
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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.
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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 selected 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.
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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
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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.
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The Filter 1x allows for CV control of the Filter 1 frequency.
If the Formant filter is used this is the “X” parameter - see
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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.
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Chapter 17
Subtractor Synthesizer
Introduction
!
Subtractor is an analog-type polyphonic synthesizer based on subtractive synthesis, the method used in analog synthesizers. This chapter will go through all parameters of each section of Subtractor. In addition to the parameter descriptions, 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:
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Up to 99 Voice Polyphony.
You can set the number of voices for each Patch.
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Dual Filters.
A combination of a multimode filter and a second, linkable, lowpass filter allows for complex filtering effects. See
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Two Oscillators, each with 32 waveforms.
.
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Frequency Modulation (FM).
See “Frequency Modulation (FM)”
.
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Oscillator Phase Offset Modulation.
This is an unique Subtractor feature that generates waveform variations. See
.
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Two Low Frequency Oscillators (LFO’s)
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Three Envelope Generators.
.
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Extensive Velocity Control.
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Extensive CV/Gate Modulation possibilities.
.
Loading and Saving Patches
.
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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.
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It is worth noting here that all waveforms can be radically transformed using Phase offset modulation (see
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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.
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5
6
19
20
21
22
15
16
17
18
23
24
25
26
27
11
12
13
14
7
8
9
10
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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 guideline, 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
Sawtooth
|
Description
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
Sine
The Triangle waveform generates only a few harmonics, spaced at odd harmonic numbers. This produces a flute-like sound, with a slightly hollow character.
The sine wave is the simplest possible waveform, with no harmonics (overtones). The sine wave produces a neutral, soft timbre.
This waveform emphasizes the higher harmonics, a bit like a sawtooth wave, only slightly less bright-sounding.
This waveform features a rich, complex harmonic structure, suitable for emulating the sound of an acoustic piano.
This waveform generates a glassy, smooth timbre. Good for electric piano-type sounds.
This waveform is suitable for keyboard-type sounds such as harpsichord or clavinet.
This waveform is suitable for electric bass-type sounds.
This is a good waveform for deep, sub-bass sounds.
This produces a waveform with strong formants, suitable for voice-like sounds.
This waveform produces a metallic timbre, suitable for a variety of sounds.
This produces a waveform suitable for organ-type sounds.
This waveform is also good for organ-type sounds. Has a brighter sound compared to waveform 13.
This waveform is suitable for bowed string sounds, like violin or cello.
Similar to 15, but with a slightly different character.
Another waveform suitable for string-type sounds.
This waveform is rich in harmonics and suitable for steel string guitar-type sounds.
This waveform is suitable for brass-type sounds.
This waveform is suitable for muted brass-type sounds.
This waveform is suitable for saxophone-like sounds.
A waveform suitable for brass and trumpet-type sounds.
This waveform is good for emulating mallet instruments such as marimba.
Similar to 23, but with a slightly different character.
This waveform is suitable for guitar-type sounds.
This is a good waveform for plucked string sounds, like harp.
Another waveform suitable for mallet-type sounds (see 23-24), but has a brighter quality, good for vibraphone-type sounds.
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|
Waveform
28
29
30
31
32
|
Description
Similar to 27, but with a slightly different character.
This waveform has complex, enharmonic overtones, suitable for metallic bell-type sounds.
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.
Similar to 30, but with a slightly different character.
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, regardless of any incoming note pitch messages, although the oscillator still reacts to note on/off messages. This can be useful for certain applications:
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When Frequency Modulation (FM - see “Frequency Modulation (FM)” ) or Ring Modulation (see
) is used.
This produces enharmonic sounds with very varying timbre across the keyboard.
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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.
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Adding a second oscillator enables many new modulation possibilities which can produce richer timbres. A basic example 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 oscillators, 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 position.
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 Subtractor, and could be regarded as an “extra” waveform for Oscillator 2, as it is internally routed to the Oscillator 2 output. 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.
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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.
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There are three Noise Generator parameters. These are as follows:
|
Parameter
Noise Decay
Noise Color
Level
|
Description
This controls how long it takes for the noise to fade out when you play a note. Note that this is independent 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.
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.
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 (seemingly) cannot provide the commonly used pulse waveform and the associated pulse width modulation (PWM). Or oscillator 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).
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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 multiplies the two waveforms with each other. The resulting waveforms can be seen in the illustration below.
Ampl.
t.
Ampl.
2. The result of subtraction: t.
3. The result of multiplication:
Ampl.
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 Envelopes to modulate the phase offset.
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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 example of PWM.
Note that if you activate waveform subtraction with a Phase offset set to “0” for an oscillator, the second waveform 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.
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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 harmonic 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.
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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 (lowering) 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.
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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 Modulator, 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.
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The Filter Section
In subtractive synthesis, a filter is the most important tool for shaping the overall timbre of the sound. The filter section 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 effects.
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 illustrated 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 rolloff 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.
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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.
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• 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 really 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.
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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 maximum, all frequencies in the waveform are heard. Gradually changing the Filter Frequency produces the classic synthesizer filter “sweep” sound.
) 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 narrower. 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 lowpass 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.
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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.
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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 filters 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.
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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 frequency for Filter 2, but keeping the relative offset.
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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!
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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 modulation envelope which has selectable modulation destinations.
Level
Sustain
(level)
Time
Attack
(time)
Key Down
The ADSR envelope parameters.
Decay
(time)
Release
(time)
Key Up
Attack
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the maximum 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 silence, 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.
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If you wanted to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain parameter 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
leve
l, 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 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.
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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
Osc 1
Osc 2
Osc Mix
FM
Phase
Freq 2
|
Description
Selecting this makes the Mod Envelope control the pitch (frequency) of Osc 1.
Same as above, but for Osc 2.
Selecting this makes the Mod Envelope control the oscillator Mix parameter. Both oscillators must be activated for this to have any effect.
Selecting this makes the Mod Envelope control the FM Amount parameter. Both oscillators must be activated for this to have any effect.
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.
Selecting this makes the Mod Envelope control the Frequency parameter for Filter 2.
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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 waveform 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 between 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
Triangle
Inverted Sawtooth
Sawtooth
Square
Random
Soft Random
|
Description
This is a smooth waveform, suitable for normal vibrato.
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.
This produces a “ramp down” cycle, the same as above but inverted.
This produces cycles that abruptly changes between two values, usable for trills etc.
Produces random stepped modulation to the destination. On some vintage synths, this is called “sample & hold”.
The same as above, but with smooth modulation.
Destination
The available LFO 1 Destinations are as follows:
|
Destination
Osc 1&2
Osc 2
Filter Freq
FM
Phase
Osc Mix
|
Description
Selecting this makes LFO 1 control the pitch (frequency) of Osc 1 and Osc 2.
Same as above, but for Osc 2.
Selecting this makes the LFO 1 control the filter frequency for Filter 1 (and Filter 2 if linked).
Selecting this makes the LFO 1 control the FM Amount parameter. Both oscillators must be activated for this to have any effect.
Selecting this makes the LFO 1 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.
Selecting this makes the LFO 1 control the oscillator Mix parameter.
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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 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 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 crossmodulation effects, with several LFO cycles that “beat” against each other. This also enables LFO 2 to produce modulation rates that vary across the keyboard (see the “Keyboard Tracking” parameter below).
Destination
The available LFO 2 Destinations are as follows:
|
Destination
Osc 1&2
Phase
Filter Freq 2
Amp
|
Description
Selecting this makes LFO 2 modulate the pitch (frequency) of Osc 1 and Osc 2.
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.
Selecting this makes the LFO 2 modulate the filter frequency for Filter 2.
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 example, 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.
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SUBTRACTOR SYNTHESIZER
q
If the LFO is set to modulate the phase offset, LFO keyboard tracking can produce good results. For example,
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 common 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.
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The following parameters can be velocity controlled:
|
Destination
Amp
FM
M. Env
Phase
Freq 2
F. Env
F. Dec
Osc Mix
A. Attack
|
Description
This let’s you velocity control the overall volume of the sound. If a positive value is set, the volume will increase 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.
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.
This sets velocity control for the Mod Envelope Amount parameter. A positive value will increase the envelope amount the harder you play. Negative values invert this relationship.
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 values invert this relationship.
This sets velocity control for the Filter 2 Frequency parameter. A positive value will increase the filter frequency the harder you play. Negative values invert this relationship.
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.
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.
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.
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 messages should affect the sound. Subtractor 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 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).
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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
F. Freq
F. Res
LFO 1
Phase
FM
|
Description
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.
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.
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.
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.
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 previous, 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 Portamento 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 Portamento at all, set this knob to zero.
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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
F. Freq
LFO 1
|
Description
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.
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.
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SUBTRACTOR SYNTHESIZER
|
Destination
Amp
FM
|
Description
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 relationship.
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. Using 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 available 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.
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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 following 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 device. 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 selected:
• Amp Envelope
• Filter Envelope
• Mod Envelope
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Chapter 18
Malström Synthesizer
Introduction
!
The Malström is a polyphonic synthesizer with a great number of different routing possibilities. It is based on the concept 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 produce 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.
for details.
• Two Modulators, featuring tempo sync and one-shot options.
• 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 Filter Envelope” for details.
• Polyphony of up to 16 voices.
• Velocity and Modulation control.
.
• A number of CV/Gate Modulation possibilities.
.
• 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.
for more details.
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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 controlled, 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 processing 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
.
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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 generate 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.
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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 receives 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 graintable, 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.
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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 motion 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.
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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 actually work like envelopes.
Modulator parameters
The two Modulators have a few controls in common, but there are also some differences. Both the common parameters 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).
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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 applied. 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
).
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
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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
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 filters, a filter envelope and a waveshaper.
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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.
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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 signals. 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 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 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.
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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 filter for example, the cutoff frequency determines the limit above which high frequencies will be cut off. Frequencies 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 normally 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 example; 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.
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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.
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MALSTRÖM SYNTHESIZER
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.
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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!
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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 respective 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 outputs. This is essentially a mono configuration and hence Spread should probably be set to “0”.
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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).
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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 series (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.
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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.
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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 maximum 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 system 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 separately (polyphonic).
This will be most apparent with longer release times.
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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 settings for how incoming MIDI Pitch Bend and Modulation wheel messages should affect the sound, but also two functional 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 clicking 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 negative 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.
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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 increase 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
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.
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Connections
Flipping the Malström around reveals a wide array of connection possibilities. Most of these are CV/Gate related. Using 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.
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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 various Malström parameters from other devices, or from the modulation outputs of the same Malström device. These inputs 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 device.
The Modulation Outputs are:
• Mod:A
• Mod:B
• Filter Envelope
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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.
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Chapter 19
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.
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MONOTONE BASS SYNTHESIZER
Panel overview
The Monotone front panel contains the following sections:
1
4 5 6
2
7
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.
8 9
3
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MONOTONE BASS SYNTHESIZER
Signal flow
The picture below shows the basic signal flow in Monotone:
OCT Wave
Oscillator 1 Noise
Drive Freq Res A D S R Amount Amount
FM Env
Oscillator 2
Detune
OSC
MIX
Filter Amplifier Chorus Delay
Master
Volume
ENV
(Freq) KEY LFO
(Freq)
VEL Rate Spread Time Feedb.
LFO
(pitch)
OCT Wave Semi
Envelope LFO
Shape
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.
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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
and
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.
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MONOTONE BASS SYNTHESIZER
Range
D
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
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
) 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 even 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.
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MONOTONE BASS SYNTHESIZER
• Triangle
The Triangle wave only contains odd harmonics, and at lower intensities than the square wave overtones. 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
FM Env
D
Turn the knob to have the oscillator 2 signal frequency modulate the oscillator 1 signal according to the current Envelope settings (see
).
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.
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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
FREQ
(Cutoff Frequency)
Frequency
Amplitude
Amplitude
Amplitude
Time
RESONANCE
FREQ
(Cutoff Frequency)
Frequency
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Amplitude
Amplitude
Amplitude
Time
RESONANCE
FREQ
(Cutoff Frequency)
Frequency
!
Be careful when using high Resonance values as this could generate quite loud audio levels!
MONOTONE BASS SYNTHESIZER
Time
Time
Env
D
).
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 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
(time)
Key Down
The ADSR envelope stages.
Decay
(time)
Release
(time)
Key Up
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 instantly. 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.
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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 parameter 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.
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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 features an LFO which can be set to control Oscillator pitch (see
) 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.
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MONOTONE BASS SYNTHESIZER
The Envelope section
The Envelope section features a standard ADSR envelope, which can be used for controlling Oscillator Frequency
Modulation (see
) and/or Filter Frequency (see “Env” ).
The various envelope stages work exactly like those of the Amplifier, see
Level
Volume
(level)
Sustain
(level)
Time
Attack
(time)
Key Down
The ADSR envelope stages.
Decay
(time)
Release
(time)
Key Up
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.
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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 modulations 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 .
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Chapter 20
ID8 Instrument Device
Introduction
The ID8 Instrument device is a synth module packed with great sounds - ideal for quickly creating nice complete arrangements. 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.
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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.
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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 semitones 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”, “Expression” and “Breath Control”. The parameter assignments can be a little different depending on selected sound. However, “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. However, 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
for more details.
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Chapter 21
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.
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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).
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Signal flow
The picture below shows the basic signal flow in Rytmik:
Sample Playback (x8) Distortion (x8)
Send (x8)
Filter (x8)
Pan (x8)
Send (x8)
Vol (x8)
Reverb
Delay
Clean
Lvl (x1)
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 Channel).
• 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
and
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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:
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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:
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
).
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 always included (as opposed to other sampler devices in Reason).
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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.
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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.
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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 Compressor. The Send Effects can be used by all Drum Channels simultaneously, and the effects are active simultaneously.
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Delay
This is a delay with two different modes - Stereo and Ping Pong. The delay repeats are tempo synced to the main sequencer 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.
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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.
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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 effect.
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 affected, 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”.
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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 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.
D
Click and drag the Release box up/down to set the compressor release time.
Range: 0-200 ms
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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 positive 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 frequencies 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%.
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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
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Connections
!
Remember that CV connections are NOT stored in the Rytmik patches! If you want to store CV connections between 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 processing.
Note that Drum Channels routed via separate outputs are automatically disconnected from the Master FX section. 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.
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Chapter 22
Radical Piano
Introduction
!
Radical Piano is designed to be a straightforward, awesome sounding and very flexible piano. Radical Piano combines sample playback technology with physical modeling to give you great sound quality and seamless dynamic response 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.
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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.
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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
and
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 selected 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 pronounced tone.
Changing the Character value temporarily mutes the audio outputs.
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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 experimental 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-
.
• 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 respond 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 volume.
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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 controller 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.
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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 sustain pedal down, all other strings in the piano will also vibrate at various intensities. Similarly, if you hold down a number 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.
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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 release 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 pronounced 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 characteristics 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.
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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
.
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!
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-
to Natural (see
).
• 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.
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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
) as well as any Pitch CV modulation (see
) 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 master 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
for more details.
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Chapter 23
Klang
Tuned Percussion
Introduction
!
The Klang Tuned Percussion instrument features an assortment of high-quality multi-sampled tuned percussion instruments - 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:
3 4
1
5 6
2
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.
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Using Klang
Loading and saving patches
Loading and saving patches is done in the same way as with any other internal Reason device, see
and
Global performance and “play” controls
Note
The Note LED lights up each time Klang receives a MIDI Note On.
Range
D
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
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.
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.
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KLANG TUNED PERCUSSION
!
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 desired 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:
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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 recorded 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.
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• 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.
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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
409
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.
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).
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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.
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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
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).
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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
413
Volume
(level)
Sustain
(level)
Time
Attack
(time)
Decay
(time)
Release
(time)
Key Up Key Down
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.
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 parameter 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.
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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” ).
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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.
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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” control. 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 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 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.
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KLANG TUNED PERCUSSION
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Chapter 24
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 highquality 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:
3 4
1
5 6
2
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.
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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
and
Global performance and “play” controls
Note
The Note LED lights up each time Pangea receives a MIDI Note On.
Range
D
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
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.
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.
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PANGEA WORLD INSTRUMENTS
!
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 desired 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:
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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.
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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.
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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.
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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.
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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.
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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
Vel
D
Turn the Vel knob to set how much the cutoff/center frequency should be modulated by Keyboard Velocity.
Range: 0% to 100%.
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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
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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
Key Down
Attack
(time)
Decay
(time)
Release
(time)
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 parameter 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.
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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.
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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.
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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” control. 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” ).
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PANGEA WORLD INSTRUMENTS
Connections
!
Remember that CV connections are NOT stored in the Pangea 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 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.
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PANGEA WORLD INSTRUMENTS
Chapter 25
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 reverb 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:
3 4
1
5 6
2
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.
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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
and
Global performance and “play” controls
Note
The Note LED lights up each time Humana receives a MIDI Note On.
Range
D
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
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.
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.
437
HUMANA VOCAL ENSEMBLE
!
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 desired 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:
438
HUMANA VOCAL ENSEMBLE
• Mars ah
A male 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.
• 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. Conducted 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. Conducted by Robert Geary. The samples are from Soundiron's Olympus Symphonic Choir.
• Venus ah
439
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.
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.
440
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.
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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.
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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.
443
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
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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%.
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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
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%.
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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
Key Down
Attack
(time)
Decay
(time)
Release
(time)
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 parameter 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.
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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.
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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.
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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” control. 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” ).
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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 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 Humana device, these outputs are auto-routed to the first available outputs in the I/O device.
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HUMANA VOCAL ENSEMBLE
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HUMANA VOCAL ENSEMBLE
Chapter 26
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
). Just like the NN-19, NN-XT lets you load samples and create multi-sample patches by mapping samples across the keyboard. The sound can then be modified by a comprehensive set of synth-type parameters. 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
D
8 stereo output pairs.
This makes it possible to route different samples to different mixer channels for individual effect processing (see
D
The possibility to create layered sounds.
D
The possibility to create sounds that only play over certain velocity ranges, velocity switched key maps and velocity crossfading.
See “Setting velocity range for a Zone” .
D
Key maps with individual synth parameter settings for each sample.
.
!
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.
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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.
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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
and
• 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 control.
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.
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NN-XT SAMPLER
Loading SoundFonts
The SoundFont format was developed by E-mu systems in collaboration with Creative Technologies. It 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. 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 parameters 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 assigned 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.
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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 remote 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 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)
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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! Listen 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 automation recording.
The controls are, from left to right:
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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
) and the key that triggered 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.
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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
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.
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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 information 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.
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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 completely different.
• A zone can be empty, playing no sample at all.
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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 focus. 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.
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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.
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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 selecting 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 select all samples that belong to the same group:
Clicking in the
Group column...
...selects all zones in the group
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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 modifier 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
).
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 parameter 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 filtering 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:
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NN-XT SAMPLER
Whenever two or more
selected
zones have conflicting parameter settings, NN-XT will notify you about this by showing 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 settings.
!
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.
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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
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.
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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.
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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:
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Click the Browse Samples button.
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Select Browse Samples from the Edit menu or the NN-XT context menu.
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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 samples for the zone.
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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.
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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
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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:
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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
). Furthermore, the sample will play
back in its unprocessed state. That is, without any synth-parameters applied (see “Synth parameters”
).
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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. However, 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.
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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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NN-XT SAMPLER
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
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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 adjustment 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.
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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
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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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NN-XT SAMPLER
Selecting a Group and/or Zones in a Group
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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 performed by making several recordings of different notes at close intervals, and then mapping these samples to separate, 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 unnatural when played too far from its original pitch, since the amount you can transpose a sound without negatively affecting 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.
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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 parameters 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 extending 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:
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NN-XT SAMPLER
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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 turning the Hi Key knob, all other selected zones will also be extended to D4 as the High Key.
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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!
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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.
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NN-XT SAMPLER
• Dragging the boundary handles for that zone will also simultaneously change the key range for a number of surrounding zones if:
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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.
!
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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 differently 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 before 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).
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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.
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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
.
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
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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 automatically 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.
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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.
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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.
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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:
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Make sure the zone has edit focus (for example by clicking on it), and do one of the following:
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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).
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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 instruments and/or each other:
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Make sure the zone has edit focus (for example by clicking on it).
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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).
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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 adjustments 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:
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Trust that the root note information in the files is already correct.
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Manually adjust the root notes (and tuning) for all the samples.
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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:
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The zones will be sorted in the display (from top to bottom - lowest key first) according to the root keys.
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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
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This Edit menu item will give each zone a key range of one semitone (i.e. one key), starting from C2 and upwards.
The function does not take root key into account. It simply places each selected sample on successive keys according the position in the sample list (from the top down).
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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.
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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
8 0
60
40
20
Velocity 0
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.
Zone 3
Zone 2
Zone 1
127
100
8 0
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.
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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 automatically sort the selected zones in the display in descending order according to their set low or high velocity values.
When you invoke this option, the selected zones will be sorted from top to bottom starting with the one with the highest “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 crossfades you adjust the fade out and fade in values for the overlapping zones.
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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 gradually 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
8 0
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
8 0
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.
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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.
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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” ).
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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 repetition.
• 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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NN-XT SAMPLER
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 selected zones. Adjusting a parameter with multiple zones selected, will set the parameter to the same value for all selected 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.
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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 decayed to silence.
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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
Lo Vel and Hi Vel
These parameters are described in
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.
!
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To select which output a selected zone should be directed to, use the knob marked “Out” in the sample parameter 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.
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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 parameter 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 regardless 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 assigned 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 previous, 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
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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 External 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
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
).
Mod Dec
).
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
Sample Start
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 semitone).
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, frequencies 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
• Res
Technically, this knob controls feedback of the output signal from the filter, back to its input. Acoustically it emphasizes 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 constant 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 parameter 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
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
). 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-
:
• 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 determines 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
Key
Key 2
Jump
|
Description
This will make the pan position shift gradually from left to right, the higher up on the keyboard you play.
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.
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 appears 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 frequency, 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 deactivated, 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
Triangle
|
Description
This is a smooth waveform, suitable for normal vibrato.
Inverted Sawtooth
Sawtooth
Square
Random
Soft Random
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.
This produces a “ramp down” cycle, the same as above but inverted.
This produces cycles that abruptly change between two values, usable for trills etc.
Produces random stepped modulation to the destination. Some vintage analog synths called this feature “sample & hold”.
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 described 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
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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 27
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 directly into the NN-19.
Also, there are thousands of high quality sample libraries available, covering every conceivable musical style or direction 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” (analog to digital). This “samples” the signal at very short time intervals and converts it to a digital representation of the analog 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 sampled 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 rootkey) 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 sample 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 piano, for example, sound good across the whole keyboard, you need to first have made many samples at close intervals 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 samples 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.
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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 parameters 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 assigned 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.
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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.
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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 displayed, 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 clicking 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 standardized data format containing wavetable synthesized audio and information on how it should be played back in wavetable 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.
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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 settings for it as with any other sample.
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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:
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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.
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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:
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By clicking on a non-selected key zone in the display.
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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
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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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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.
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By using the “Lowkey” and “Highkey” knobs to set a lower and upper range, respectively.
Deleting a Key Zone
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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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If the zone contained a sample prior to loading, this will be replaced , both in the zone and in the sample memory , unless the sample was also used by anothe r key zone, in which case it will be kept.
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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.
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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 guitar 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
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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.
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To remove a sample from a key zone/map, without removing it from memory, you can either select “No Sample” with the Sample knob for that zone, or simply replace it with another sample in the same way.
Removing all unassigned samples
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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.
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Select the key zone(s) that contains the out of tune sample(s), and use the Tune knob below the keyboard display. q
If all samples originate from different sources, and all or most of them are pitched slightly different (a not uncommon sampling scenario), you could first tune them so that they all match each other, and then, if necessary, use the Sample Pitch controls in the Osc section to tune them globally to the “song” you wish to use the samples in.
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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 external 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:
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OFF
No looping is applied to the sample.
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FWD
The part between the loop points plays from start to end, then the cycle is repeated. This is the most common loop mode.
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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
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The Solo Sample button will allow you to listen to a selected sample over the entire keyboard range.
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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 “Automap 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:
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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.
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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
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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 important 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 following 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 samples’ start position, so that they will play back from a position further “into” the samples’ waveform. This is useful mainly for two things:
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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 beginning which was not intended to be part of the sample. By adjusting the sample start position, this can be removed.
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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:
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Octave steps
The range is 0 - 8. The default setting is 4.
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Semitone steps
Allows you to raise the frequency in 12 semitone steps (1 octave).
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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
Keyboard Tracking
The Osc section has a button named “Kbd. Track”. If this is switched off, the sample’s pitch will remain constant, regardless 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
). You can set negative or positive values here, which determines whether an envelope parameter 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 rolloff 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 gentler 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 frequencies 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
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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 maximum, all frequencies in the waveform are heard. Gradually changing the Filter Frequency produces the classic synthesizer 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 result.
Resonance
The filter resonance parameter (sometimes called Q) 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 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 narrower. Generally, the Notch filter produces more musical results using low resonance settings.
Envelope Section
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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 Envelope 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 Filter 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 frequency. 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 produce 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
Triangle
Inverted
Sawtooth
Sawtooth
Square
Random
Soft Random
|
Description
This is a smooth waveform, suitable for normal vibrato.
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.
This produces a “ramp down” cycle, the same as above but inverted.
This produces cycles that abruptly changes between two values, usable for trills etc.
Produces random stepped modulation to the destination. Some vintage analog synths called this feature “sample & hold”.
The same as above, but with smooth modulation.
Destination
The available LFO Destinations are as follows:
|
Destination
Osc
|
Description
Selecting this makes LFO control the pitch (frequency) of the sample patch.
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 setting 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 common 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
Amp
F. Env
F. Dec
S.Start
A. Attack
|
Description
This let’s you velocity control the overall volume of the sound. If a positive value is set, the volume will increase 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.
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.
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.
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.
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.
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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
F. Freq
F. Res
F. Dec
LFO
Amp
|
Description
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.
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.
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.
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.
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:
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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”.
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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 previous, 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 Portamento 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 Portamento at all, set this knob to zero.
Setting Number of Voices - Polyphony
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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 number 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
Key
Key 2
Jump
|
Description
This will shift the pan position gradually from left to right the higher up on the keyboard you play.
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.
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
LFO 1
Amp
This sets external modulation control of the filter frequency parameter. A positive value will increase the frequency with higher external modulation values. Negative values invert this relationship.
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.
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 relationship.
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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 example, 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 parameters:
• 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
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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
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NN-19 SAMPLER
Chapter 28
MIDI Out Device
Introduction
The MIDI Out Device is designed for routing MIDI out of the Reason Rack Plugin instance to other tracks/destinations in your main DAW. A typical scenario would be to route MIDI from a Player device in Reason Rack Plugin to another 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 instance 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).
526
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 “throughput” 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:
527
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” .
528
MIDI OUT DEVICE
Chapter 29
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
and
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.
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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. Therefore, 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:
Input signal
Feedback
Delay Line
Delay Mod
Depth
Mod
Rate
+
Output signal
: audio signal
: control signal
Amplitude
Delay
Frequency
(log)
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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.
532
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 parallel, and therefore provides a much richer and denser effect than the Chorus algorithm.
The picture below shows the basic principle of the BBD algorithm:
Input signal
Delay Line
Delay Line
Delay Line
+
Output signal
: audio signal
: control signal
Delay Mod
Depth
Mod
Rate
Noise
Mod
Amplitude
Delay
Delay
Frequency
(log)
Here you set the delay time between the dry and processed signals. The delay is preset scaled between the three delay lines. In practice, this determines where the notches/peaks will appear in the frequency spectrum.
Range: 1.00-30.00 ms
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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
).
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 effect.
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).
534
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
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 “handles” sideways.
535
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
(log)
Frequency
(log)
Frequency Range Frequency Range
The first example shows the modulation of the partials at full bandwidth. The second example shows the partial modulation 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
Time
Input signal
5 “extracted” grains
Level
An example of a signal generated from 5 grains of the input signal.
536
QUARTET CHORUS ENSEMBLE
The resulting signal is generated by appending and crossfading the grains.
Time
Phase
The Random Phase function randomly alters the phase of the grains to create a “bubbly” kind of effect, caused by
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 produce a really fat and dense chorus/ensemble effect, whereas low values generate a “thinner” effect.
537
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”
).
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QUARTET CHORUS ENSEMBLE
Chapter 30
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
and
Volume
This is the master volume control.
540
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
) to the Frequency control of the Phaser (see “Frequency”
), Flanger (see “Frequency” ) and Filter (see
The control is bipolar, which means that negative values will invert the modulation.
Mod -> Freq
This controls the modulation amount from the Modulator (see
) to the Frequency control of the Phaser (see “Frequency”
), Flanger (see “Frequency” ) and Filter (see
) section.
The control is bipolar, which means that negative values will invert the modulation.
541
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
Modulator” ) to a separate built-in amplifier.
The control is bipolar, which means that negative values will invert the modulation.
542
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 principle 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:
Input signal
Amplitude
Feedback
Bandwidth
Allpass filters
Stage 1 Stage 2 Stage 3 Stage 40
+
Output signal
Frequency
: audio signal
: control signal
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
543
SWEEPER MODULATION EFFECT
Bandwidth
This controls the bandwidth of the all-pass filter(s) in the phaser.
Amplitude 3-stage phaser
Low Bandwidth
Amplitude 3-stage phaser
High Bandwidth
Frequency
(log)
Feedback
This controls the level/intensity of the phaser peaks and notches.
Amplitude 3-stage phaser
Low Feedback
Amplitude 3-stage phaser
High Feedback
Frequency
(log)
Frequency
(log)
Frequency
(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 filters you want to use. Each all-pass filter contributes with one notch/peak in the frequency spectrum.
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SWEEPER MODULATION EFFECT
Range: 1-40 stages (notches)
Amplitude
1-stage phaser
Amplitude
2-stage phaser
Amplitude
3-stage phaser
Polarity
Frequency Frequency Frequency
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
Mute Dry
Frequency
Frequency
(log) Frequency
Frequency
(log)
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
Stage 1 Stage 2 Stage 3 Stage 40
Input signal
+
Output signal
Frequency
This will give more of a “tremolo” effect rather than phasing.
: audio signal
: control signal
545
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 chorus 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:
Input signal
Feedback
Comb Filter Delay
+
Output signal
: audio signal
: control signal
LFO Frequency
Amplitude
Frequency
Frequency
Frequency
(log)
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
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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
(log)
Mute Dry
Frequency
Frequency
(log) Frequency
Pressing this button mutes most of the dry signal in the Flanger section:
X
Feedback
Comb Filter Delay
+
Output signal
Input signal
: audio signal
: control signal
LFO Frequency
547
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:
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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
A state variable (SVF) notch filter.
Frequency
(log)
549
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, additional resonance peaks appear.
Be careful when using this filter type as high Resonance values could generate quite extreme audio levels!
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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.
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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
) or a CV trig signal on the rear panel (see
stead.
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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:
553
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, turning 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 exact same time positions. Otherwise, only the closest point will be changed. Also, any inclining/declining segment 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.
554
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 received. 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 Frequency 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.
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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.
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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
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”
).
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Chapter 31
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 processing 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, totally 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 procedures.
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
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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
.
• 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
Audio Input
Band Pass
Filter
FX
(Drive, Phaser,
Delay Send)
Pan and
Volume
Amp Env
To Low Pass
Gate
(pattern, CV,
MIDI or Trig button)
Filter Env
From Low Pass
In this, more detailed diagram, we see that the gate isn't a simple on/off switch - there is actually an amplitude envelope 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 interesting, spatial effects - especially since you can pan the three channels, the dry signal and the delay independently!
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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 second 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 setting 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 pattern 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).
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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 patterns.
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.
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ALLIGATOR TRIPLE FILTERED GATE
Filters and Modulation
The three channels have identical settings, even though their filters are of different types. Below, all descriptions apply 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, allowing for positive or negative modulation of the filter frequency.
Frequency
q
• 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, allowing for positive or negative modulation of the filter frequency.
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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.
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ALLIGATOR TRIPLE FILTERED GATE
Effects
The three channels have identical effect parameters. Distortion and phaser effects are separate for the three channels (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.
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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 effect.
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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 outputs for the three gate/filter channels. If you connect these outputs, the corresponding channel signals will be removed 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.
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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 output 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 Ma