M-Audio Wayoutware TimewARP 2600 Owner`s manual

M-Audio Wayoutware TimewARP 2600 Owner`s manual
Owner’s Manual
Copyright © 2004, by Way Out Ware, Inc.. All rights reserved.
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Table of Contents
1 The ARP 2600, 1970 – 1981 and onward... . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
2 System Requirements, Installation, Configuration, Setup and Usage . . . . . . . . . . . . . . . . . . 5
In this chapter you will find all of the platform-dependent information
you need in order to install and operate your TimewARP 2600 software
3 The Craft of audio Synthesis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
This chapter is about the facts – physical, mathematical, and auditory
– that make the TimewARP 2600, and the hardware that it emulates,
possible. We have to spend a few minutes here distinguishing between
physical signals, and the sounds that people hear in the presence of
certain kinds of signals.
4 Modular Components of the TimewARP 2600 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
In this chapter you’ll find detailed explanations of the TimewARP
2600’s features and functions.
5 Patching the TimewARP 2600. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
6 Appendices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
Table of Alternate keyboard tunings compiled by Robert Rich
7 Index
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
The ARP 2600,
1970 – 1981 and onward...
The ARP 2600 was the second product of ARP Instruments. It was released in 1970, and continued
until the manufacturer ceased operations in 1981.
Its design combined modularity (for studio flexibility, and for instructional use) and integration (for
realtime performance). Functionally, the ARP 2600 was completely modular: any signal output could
be routed to any signal input, with a patch cord. Operationally, the ARP 2600 was integrated, using
internally-wired default signal paths that made it possible to create a wide range of keyboard patches
by simply opening up slide attenuators, as though sitting in front of a mixing console.
The ARP 2600 earned an early reputation for stability, and for flexibility exceeding that of its
competitor the Minimoog. Used 2600’s in good condition command premium prices on eBay today
and businesses around the country can make a living reconditioning, rebuilding, and customizing 30year-old units.
Among rock musicians, the ARP 2600 was used by Stevie Wonder, Pete Townsend, Joe Zawinul,
Edgar Winter, Paul Bley, Roger Powell, Jean-Michel Jarre, Mike Oldfield, Herbie Hancock, and many,
many others.
Its modular design, and the popularity of its Owner’s Manual, made the ARP 2600 a widely
used teaching instrument in many schools and music conservatories around the country and
We are proud to bring you this software emulation of the 2600. Have fun with it, learn from it, but
above all, make noise with it.
CHAPTER 1 - The ARP 2600
Unfasten the seat belts of your mind. The TimewARP 2600 will be an astonishing, exhilarating, and
enlightening experience.
Creating this manual has been an astonishing, exhilarating, and enlightening experience for me. How
many are ever given the chance to revisit an earlier life, an earlier project, a project like the ARP 2600
Manual, decades later, and get it right? It’s time travel. I’m grateful to Way Out Ware for providing
me that opportunity.
When, at Alan R. Pearlman’s invitation, I began work on the original 2600 manual in September of
1970, the 2600 itself barely existed. For the first two months, I was writing “blind” - without a machine
in front of me. My first hands-on experience with a synthesizer had been only six months earlier (it
was a Putney VCS3).
I finished the text in March of 1971, Margaret Friend created the graphics, and the Owner’s Manual for
the ARP 2600 began what turned out to be a surprisingly long career. In spite of the many defects that
my inexperience contributed – the gaps in coverage, and outright errors - it became quite popular. To
this day, it still gets an occasional respectable mention in the analog-synthesis community.
When Way Out Ware’s Jim Heintz called, early in 2004, to tell me about the TimewARP 2600, a lot of
time had passed. Regarding software synthesizers, I had grown weary and cynical. Analog-modeling
software had been a decade-long disappointment; some products did interesting things but not the
things that real analog modules do. Jim, however, had already encountered, and thought about, and
solved, these problems. He owned a real 2600. He really aimed at getting it right and would not
be satisfied with anything less. It was a pleasure, finally, to accept his invitation to do an Owner’s
It’s clear, now, that Way Out Ware has set a new standard for software-based audio synthesis. The
behavior of the TimewARP 2600 software – both module-by-module and integrated into patches - is
effectively indistinguishable from that of the analog hardware that it emulates.
Soaring and swooping through the free air of analog synthesis – a world of nothing but sliders and
cords and continuously evolving patch configuration - was a capstone course at the Boston School of
Electronic Music in the 1970’s. That is the world that the TimewARP 2600, for a new generation of
musicians in a new millennium (that means you), provides access to: it is the first – and I believe only
- software synthesizer to support real-time performance by sliders and patchcords alone.
So here it is: your new Owner’s Manual, for the new TimewARP 2600.
Unfasten the seat belts of your mind. How else can you hope to experience time travel? How else
can you enjoy free flight?
Jim Michmerhuizen
Jim Michmerhuizen is the author of the original ARP 2600 Manual and Founder and Director of the Boston School of Electronic Music.
CHAPTER 1 - The ARP 2600
How this Manual is Organized
This is not a textbook; it’s a survival manual.
Chapter 2 is about installing and configuring the software so you can get up and running.
Chapter 3 is a brief introduction to the vocabulary and methods of classical analog synthesis, so that
we can understand each other throughout the rest of the book.
Chapter 4 is a module-by-module reference, including the digital extensions made possible by the
fact that the TimewARP 2600 is software – a piece of computer behavior – rather than a collection of
electronic hardware.
Chapter 5 is not in this book, but is a collection of patches, with accompanying documentation, located
in a separate document file called Patchman.pdf found on the distribution CD (or as a download from
www.wayoutware.com). The patches and commentary are keyed to the numbered chapters, sections,
and subsections of this manual. Some of the patches form a tutorial sequence, and some illustrate
vocabulary lessons and concepts.
How To Use This Manual
You’ll probably need to do a quick run through of Chapter 2 as you install the TimewARP 2600 and
learn some of the basic setup operations. After that, you can pretty much mix and match, according
to your experience:
If you’re new to audio synthesis, you’ll want to walk through the tutorial patch sequence in
Patchman.pdf, referring back to Chapter 3 for concepts and Chapter 4 for detailed module specifications
while you explore, and learn to control, the vast range and patch repertoire of the TimewARP 2600
software synthesizer.
If you already have some experience with audio synthesis, you might go directly to Chapter 4 for the
detailed module-by-module specifications of the TimewARP 2600. If you know and love the original
ARP 2600 itself, take particular note of section 4.1, where we describe what you can do with the
TimewARP 2600 that you could not do with its analog ancestor. These digital extensions include patch
load/save, additional VCO sine-wave outputs, dual-channel signal input, automatic Y-connections at
all signal outputs, sixteen keyboard micro-tuning options, MIDI Beat Clock synchronization, and MIDI
controller mapping (with subranges) for all the panel sliders.
System Requirements, Installation,
Configuration, Setup and Usage
In this chapter you will find all of the platform-dependent information you need in order to install and
operate your TimewARP 2600 software synthesizer.
The TimewARP 2600 provides an extended set of digitally-based features that no hardware-based
analog synthesizer can offer. We describe the most important of these in section 2.2.
Digidesign Pro Tools
This release of the TimewARP 2600 is an RTAS plug-in and runs only under Digidesign
Pro Tools 6.1 and later versions, on either Apple OSX or Windows XP. On either of these
platforms, the requirements for the TimewARP 2600 are essentially those of Pro Tools
System Requirements
These are the same as Pro Tools itself, including enough memory to satisfy the practical
needs of the TimewARP 2600 software.
Disk Space and RAM
The executable file occupies 6.1MB on disc, and the RAM requirements when running are
the same as the Pro Tools minimums.
System Clock
The TimewARP 2600 performs all signal generation and processing in real time. In
complex polyphonic patches, this may put a considerable load on your CPU. You may
have difficulty with a clock speed less than 800MHz; higher clock rates will increase the
number of polyphonic voices you can use, and the complexity of the patches available.
Installation on either platform is accomplished by very simple standardized installation
sequences. You must, of course, have a current installation of Pro Tools on your target
CHAPTER 2 - System Requirements, Installation, Configuration, Setup and Usage
Apple OSX
Insert the distribution CD or download the .dmg installer file from www.wayoutware.com.
Double click the .dmg installer to initiate installation.
If you agreed to the license, a TimewARP 2600 installer icon appears. Double-click it to
initiate the installation process.
When finished, start Pro Tools and proceed to
Windows XP
Insert the distribution CD or download the installer file from www.wayoutware.com. Run
the .exe installer file.
If you agreed to the license, follow the installer instructions.
When finished, start Pro Tools and proceed to
Real-time Instrument for Performance and Recording
The TimewARP 2600 can transform your computer into a real-time instrument for live
performance or recording.
To set this up, create an audio track in Pro Tools. Select either mono or stereo in the
track-creation window; this will determine, in turn, the channel options offered by the
TimewARP 2600 for this track.
In the mix window display for this track, at the track-inserts block up at the top, click on
the first insert selector. Choose the TimewARP 2600 plug-in from the appropriate menu.
Now create another track, but in the track creation dialog, choose MIDI track instead of
audio track, and route the track output to the TimewARP 2600.
As input for the new track, select your MIDI keyboard device, and enable the track for
recording. Pro Tools will route all incoming MIDI events from your MIDI keyboard device
to the TimewARP 2600 plug-in.
If your MIDI keyboard has any additional MIDI control devices (sliders, knobs, buttons,
etc.), you can assign these to any TimewARP 2600 sliders, knobs and switches that you
choose. For details, see section 2.2.
Processing Audio Signals
The TimewARP 2600 can be used to process audio signals in a similar manner to a reverb
or other effects plug-in.
To set this up, create an audio track or select a track already recorded in Pro Tools. (If
this is a new track, select either mono or stereo in the track-creation window; this will
determine, in turn, the channel options offered by the TimewARP 2600 for this track.)
In the mix window display for this track, at the track-inserts block up at the top, click
on the first available insert selector. Choose the TimewARP 2600 plug-in from the
appropriate menu.
CHAPTER 2 - System Requirements, Installation, Configuration, Setup and Usage
If the current track is mono, the TimewARP 2600 for this track will be configured for one
channel of input, and you can select between mono and stereo output. If the current
track is stereo, the TimewARP 2600 insert will automatically be configured for stereo
throughout. Confirm this, when the synthesizer panel comes up, by observing that the
preamp module in the upper left corner has two channel outputs rather than just one.
If you are processing an existing track, press play on the transport control to hear
the audio being processed by the TimewARP 2600 in real time. You need to select an
appropriate TimewARP 2600 patch (such as those in the Voice or Guitar Effects categories
in the Factory group) to successfully process audio this way.
If you are processing a live track, enable the track’s record mode, and then audio from
your input will be fed through the TimewARP 2600. You need to select an appropriate
TimewARP 2600 patch (such as those in the Voice or Guitar Effects categories in the Factory
group) to successfully process audio this way.
MIDI Tracks
Running as a track plug-in, the TimewARP 2600 can process prerecorded MIDI tracks.
To do this, set up two tracks just as you did above in section an audio track that
is running the TimewARP 2600 as a track insert, and a MIDI track whose output is routed
to the TimewARP 2600.
When you play this track (use the Pro Tools transport window to play, rewind, pause the
MIDI sequence, etc.), the MIDI events stored in the track sequence will be routed to the
TimewARP 2600. MIDI note-events will become key depressions at the virtual keyboard;
MIDI pitch-bend will turn the virtual keyboard pitch-bend knob; and MIDI controllers
that are mapped to TimewARP 2600 sliders, knobs or switches (see section 2.2) will move
those controls.
Selecting Patches
The TimewARP 2600 gives you a three-level hierarchy for storing and organizing your
patches. All Patches are sorted into various Categories, which are in turn sorted into major
Groups. Each of the three patch selection buttons generates a drop-down list associated
with one layer in this hierarchy.
Groups, Categories, and Patches can also be selected by keyboard shortcuts. The up/down
arrow keys on the computer keyboard select Patches, the left/right arrow keys move
between Categories, and using the control key with the left/right arrow keys moves between
Making Patch Connections
Use the mouse to connect any output to any input jack. Position the cursor at any signal
output, click and hold down the mouse button, and drag the patch cord to any signal
input. To remove a patch cord, drag either end away from its signal jack.
The TimewARP 2600 will not connect two outputs or two inputs together. If you drag
from one jack to another but no patch cord appears, it may be that both jacks are inputs
or both are outputs.
CHAPTER 2 - System Requirements, Installation, Configuration, Setup and Usage
The TimewARP 2600 will allow Y-connections, distributing a single output signal to
multiple destinations by holding the control key down while clicking an output that
already has a cable connected to it. A second cable will appear attached to the cursor,
which can then be plugged into another input.
Using the Sliders and Knobs
To adjust a slider or knob, drag it with the mouse. For increased resolution, hold down the
command (Apple) or control (Windows) key during the drag operation.
If you pause for a moment over a slider, a display will pop up with the numerical value
and name of the parameter it controls.
If you have MIDI controller hardware, you can easily connect it to the sliders, knobs or
switches; see section 2.2.2.
Using the Virtual Keyboard Display
The graphic TimewARP 2600 keyboard display responds directly to mouse events; click
on any key to create a MIDI keydown signal. This is useful when you are creating and
tuning a new patch, if you don’t have a hardware MIDI keyboard handy.
The virtual keyboard also displays – as key depressions - incoming MIDI note-events. Use
this to monitor your external MIDI keyboard connection and activity.
Polyphonic Operation
The TimewARP 2600 can respond to as many as eight simultaneous keydown events. Use
the Voices dropdown (see section to set the number of voices.
Saving and Loading Patches
You may save patches using the Save or Save As buttons or load patches with the Patch
Manager (see section 4.1 below). There is no limit to the number of patches you can
create and save.To enable the Save button, the Lock button (padlock icon) must be in the
unlocked mode.
You may wish to create a bank of “template” patches, generic versions of often-used
The TimewARP 2600 responds to MIDI bank-select and patch-select commands.
MIDI Communication and Control
Slider, Knob and Switch Control
You may control any slider, knob or switch on the TimewARP 2600 plug-in with any MIDI
controller. If you have a hardware control surface, or if your MIDI keyboard has slider or
knob control devices, you can use them to control any combination of sliders, knobs or
switches in a patch.
Any connections you set up are global to the TimewARP 2600 – you can store them
independently of any specific patch (see section below).
To connect a slider, knob, or switch to an external MIDI controller, hold down the control
CHAPTER 2 - System Requirements, Installation, Configuration, Setup and Usage
(Mac) or shift (Windows) key and click on the slider, knob or switch image. The responding
dialog box offers you both global and patch-specific connections.
Global MIDI Map Settings
Select the controller number
number, either by directly
typing it in, or by twiddling the physical control
on the MIDI device you intend to use.
Set the response curve, polarity, and range. The
response curve may be linear, or exponential,
or logarithmic. The polarity is either direct or
inverted. To set the Control Range, drag either
end of the bar inward. Whenever the bar covers
less than the entire controller range, you may
also drag it to the right or left, to adjust the
range offset.
You may, if you wish, assign the same external
control to several TimewARP 2600 sliders or
knobs, and set a different curve, polarity, and
range for each one. In this way, you can create
patches and configurations of the TimewARP
2600 that are specifically adapted for expressive
performance either live or in the studio.
Patch-Specific Settings
Besides globally assigning a slider, knob or switch to a MIDI controller, you may also
assign the slider or knob to be controlled by the velocity and/or aftertouch parameters
of incoming keyboard events. (These assignments are not global; they are stored with
individual patches.)
In real time, these controller values are summed with the current global control values
to determine the real-time value of the slider. So, by careful tuning beforehand, you can
have both keyboard expression and hardware knob/slider control at the same time.
MIDI Patch Selection
In each group of patches, the first 128 are available to standard incoming MIDI patch selection
Patch categories don’t affect this numbering. For example, if the first category in a group
has 127 patches, then the second category will have just one patch available for MIDI
patch selection.
Groups themselves can be selected with MIDI bank-select commands
The Craft of Audio Synthesis
This chapter is about the facts – physical, mathematical, and auditory – that make the TimewARP
2600, and the hardware that it emulates, possible. We have to spend a few minutes here distinguishing
between physical signals, and the sounds that people hear in the presence of certain kinds of
This is important because synthesizer equipment can only deal with signals – physical commotion
of one sort and another. When you are fiddling with synthesizer equipment, you are generating and
modifying signals for the sake of the interesting (we hope) sounds you hear when those signals reach
your eardrums.
Signals and Sounds
A signal is something happening: a waving flashlight, ambulance siren, referee flag
dropping, winking at a friend. Tiny disturbances of the air around us are signals for our
ears; we hear them as noise, or singing, or sirens, shrieks, growls, whatever.
The signal is the physical disturbance in the air, the movement of the eardrum. The sound
is your perception of the signal: “Hello!”
Analog and Digital Representations of Signals
The signals we are concerned with in sound synthesis are audio signals: more or less
regular variations in air pressure, at our eardrums, repeating at rates of between 20 and
20,000 times in one second.
Such signals are straightforward physical processes which can be recorded and
reproduced. One way to do that is to look at the pattern of air-pressure variation, and
model it in some other medium. During the past century this has been done with grooves
in a phonograph record, magnetic fields along a length of tape or wire, and other media.
The usual scenario is: with one or more microphones, generate an electronic model
of the vibrating air, then use the electronic signal to drive a magnetic recording head,
or amplifier, or LP recording lathe. Throughout such processing, the signals we deal
with are directly analogous to each other; except for the change in medium from air to
voltage to magnetic field strength or stylus position, the signals are identical. Graphed
or charted, they even look the same. This is analog recording.
CHAPTER 3 - The Craft of audio Synthesis
Another way to record such a signal is, with high-speed digital
circuits, to measure the underlying medium many times
each second, and store the measured numbers. This is digital
Analog signals @ 250 Hz
(approximately middle C)
Air Pressure
Output Signal
LP groove
Field on Tape
Power amp
(Volt * current = Power)
o r
Since the TimewARP 2600 is not constructed of electronic circuits,
concepts such as “voltage” don’t apply here. The TimewARP
2600 is software, a complex piece of computer behavior. The
signal medium for the original ARP 2600 was electrical pressure,
measured in volts. The signal medium for the TimewARP 2600 is
simply number sequences. We use those numbers the way the
original machine used electrical pressure; where the original
Owner’s Manual used the word “voltage” we will just say “signal”
“signal level”, and in the module specifications we will refer to
“virtual Volts” or “vV”.
Your Eardrum
250 = 4 msec
Attributes of Signals
The simplest possible signal is a sine wave. It’s like the back-and-forth motion of a point
on a circle as the circle rotates. A lot of the mathematics of sine waves is based on that
rotating-circle idea; you don’t have to get involved in that unless you’re curious about
it, but it’s helpful to train your imagination by picturing the basic sine-wave graph as a
slightly stretched coil spring like a “slinky”.
The motion of a pendulum, or of a tuning fork, swinging back and forth as they slowly
come to rest, is a decaying sine wave.
A sine wave signal has exactly three attributes: its frequency, its amplitude, and its phase.
It has no other characteristics at all. (The decaying swing of the pendulum or the tuning
fork does have one other attribute: the amount of energy/amplitude that it loses on each
swing. It’s not a “pure” sine wave.)
Fundamental Attributes
Picture a point on that rotating
circle, leaving a trail behind as it
rotates, like an airplane propeller.
Picture the trail, stretched out
behind like a coil spring, and ask
frequency =
phase (important only
in comparing two signals)
CHAPTER 3 - The Craft of audio Synthesis
Amplitude: what’s the diameter of this imaginary circle?
Frequency: how fast is it rotating?
Phase: when does it start a new cycle?
Complex Attributes
Most of the activity that reaches our ears every day is far
more complex than just a sine wave. Banging on a garbage
can creates a much more complicated sort of vibration than
tapping a tuning fork.
But any kind of motion that isn’t a sine wave can still be
analyzed as a collection of sine waves. This is called Fourier
analysis, after the man who discovered that it was possible,
and proved – mathematically – that it always worked.
This goes both ways: any complex signal can also be
constructed from a collection of sine waves with the
appropriate attributes: amplitude, frequency, and phase
In fact, any kind of motion – any repeating pattern of the sort
that we are interested in here as “sound waves” - can be
examined and analyzed under two different headings:
A complex signal...
can be made of simple ones like this...
...and this...
...and a pinch of this
...and a dash of that.
Waveshape, and Time-Domain Attributes
Any repeating motion can be diagrammed in a graph where the horizontal axis is a period
of time and the vertical one is the motion itself, back and forth. This is the time-domain
view of a signal:
Spectrum, and Frequency-Domain Attributes
Instead of looking at a signal as something in motion through a period of time, we can look
at the collection of sine-wave components of the signal. In such graphs, the horizontal axis
is a frequency range (instead of a time period), and we indicate each spectrum component
with a single vertical line in the graph. The height of the line indicates the strength of the
component at that frequency. This is the frequency-domain , or spectral-domain, view of a
It is the distribution and relative strength of spectral components that we experience as
the tone-color of a sound or sounds. “Bright”, “dull”, “sharp”, “tinny”, “heavy”, and so on
– these are all descriptive words for the spectral attributes of sounds.
CHAPTER 3 - The Craft of audio Synthesis
Harmonic Series
The spectral view of any periodic signal has components at simple multiples of the signal
frequency. For example, suppose we examine a sawtooth wave at a frequency of 110Hz.
It will have components at 110, 220, 330, 440, and so on.
A simple number sequence like this is called a harmonic series. It is interesting to think
about musically; the smaller numbers in the series all form simple musical intervals:
octaves, fifths, fourths, and thirds.
Most musical instruments generate harmonic spectra. Some have more, some have
fewer harmonics, and there are wide spectral variations even within a single instrument
depending on how it is played. In general, whatever the actual spectral components are,
they will always form a harmonic series.
In audio synthesis, you will use oscillators to generate pitched tones that have a harmonic
A harmonic series is composed of
numerically equal frequency intervals,
and that means decreasing pitch intervals
Spectrum = the sine-wave component of a signal
4 5 6 7 8 9 101112
Increasing Power
Increasing Power
just before
dying out
Increasing Pitch
Increasing Frequency
middle C
The first couple of octaves’ worth of a harmonic
series make useful musical intervals: octaves, 5ths,
4ths, 3rds and so on. Beyond the 16th, they get too
close together to sound good.
Pick any frequency; the multiples of that are a harmonic
series. A, B, and C are all harmonic series.
500 750 1K
Increasing Power
Increasing Power
after pitch
maj min
3rd 3rd
Increasing Pitch
200 300 400
Increasing Pitch
4K 6K
CHAPTER 3 - The Craft of audio Synthesis
Enharmonic Series
Enharmonic, in this context, means
“not forming a harmonic series.”
Some waveforms do not repeat
themselves at a regular interval;
the spectra of such waves will have
components at strange non-integral
frequency ratios, or they may have
shifting spectral components that die
out and reappear. Some percussion
instruments, such as kettledrums,
or bells, have enharmonic spectral
Not all spectra are harmonic. An Arbitrary collection of
sines at unrelated frequencies is "unharmonic".
In audio synthesis, you can use various
modulation techniques – such as AM
and FM - to generate enharmonic spectra.
Increasing Power
Increasing Pitch
Attributes of Random Signals
A completely random signal – noise – actually has a continuous spectrum: it does not
consist of isolated sines in a harmonic series, nor even in an enharmonic series. A noise
spectrum is a continuum of frequency components; in order to describe it, we have to
talk about how much energy is in each band in this continuum. One kind of noise may
have high-frequency energy, and another has low-frequency energy.
In audio synthesis, noise is an extraordinarily useful signal. Filters can be used to shape a
noise spectrum into almost anything – even pitched sounds.
Spectral Balance (Color)
The physical processes (such as molecular motion or random popping of electrons from
one atom to another) that we typically depend on for random electronic noise have a
frequency distribution of equal probability at any frequency, or through any frequency
interval. Curiously enough, this produces noise signals that, for audio purposes, don’t
sound properly balanced across our range of hearing. They sound too bright.
CHAPTER 3 - The Craft of audio Synthesis
Attributes of Auditory Events
“white” noise has equal energy in any
two equal frequency bands…
10 12 14 16 18 20K
so it’s NOT equal in equal pitch intervals…
20 40 80 160 320 640 1.25 2.5
5 10K 20K
octave intervals
…even when you use dB for the vertical axis…
-9dB -6
THIS is why white noise sounds so shrill
Of all the thousands of sounds you hear in a single day, only some
have a definite beginning or ending point. Many just come and go
without a clear start/stop. Those that do have a reasonably clear
start/stop we will call auditory events. The other sounds, the less
clearly defined ones, we can call auditory textures or backgrounds or
even structures.
Think how some sounds are more complicated than others. The
hum of a refrigerator is sort of simple, the hum of a tuning fork is
(as we heard above) really simple. A single bass note from a piano
is quite complicated, the way it keeps changing and evolving as
it dies away. Human speech is a very complicated kind of sound,
even when it’s your native tongue (when it’s a language you don’t
understand, it sounds even more complicated).
There are many, many more kinds of sounds to hear than you or
I have words to describe them with. Here are just a couple of the
most general distinctions people make about different kinds of
Steady-State Attributes: some sounds, once they get started, remain pretty constant.
They don’t change much while they continue, and when they stop, they just stop. We say
such sounds reach a steady state, in which we can pick out a definite Pitch, Loudness, and
Tone Color. Notes from organ pipes – or from electronic drawbar organs – are steady-state
Time-varying Attributes (i.e. “Envelopes”): some sounds have a pretty clear start
and end, but while they’re happening they change. Think of a bird song, or ordinary
human speech.
Such sounds can sometimes be analyzed as more or less rapidly changing in one or
more of those fundamental auditory attributes: pitch, loudness, or tone-color. Think, for
example, of how the sound of a plucked guitar string evolves from the moment you pick
it to the moment you can’t hear it anymore. It starts out loud and fades away; and it
starts out “bright” - with lots of harmonics – and is slowly “muffled” as it fades out.
One way to conveniently diagram what happens in such events is to chart each changing
attribute separately, in its own time-domain graph.
Such a graph is often referred to as an envelope. Over the years, some standard vocabulary
has developed for talking about envelopes: Attack (for the first part of an event), Decay (for
some later parts), and Release (for the last part, when the event is coming to an end).
CHAPTER 3 - The Craft of audio Synthesis
How Signals and Sounds Go Together... Sort Of
Signal activities, being entirely physical kinds of things, are easily nameable, measurable,
catalogable, countable. Sounds, as we pointed out above, are not quite so easily
domesticated. The music and other sounds that we listen to correlate in several wellestablished ways with the signals around us; but the correspondence is not simple. There
are always surprises.
Signal Frequency and Audible Pitch
This is probably the best-established and most reliable correspondence. It dates all the
way back to Pythagoras, the Greek philosopher who 2500 years ago worked out that
halving the length of a vibrating string made the pitch rise by one octave.
How Adding Pitches Means Multiplying Frequencies
The range of audio frequencies
– of human hearing, in other words – is
conventionally stated to be 20Hz to
20KHz. And when you think linearly, it
sounds tragic to learn of someone, say,
who can only hear up to 10KHz. But in
the realm of human pitch perception,
such a person has kept 90% of his
hearing range: from 10KHz to 20KHz is
just one octave out of the 10 we hear.
A harmonic series is composed of numerically
equal frequency intervals, and that means
decreasing pitch intervals
Frequency in cycles / seconds
This is a really important fact in
audio synthesis; you will encounter
it over and over again, in every
patch you create. It governs much
of the arithmetic of generating and
controlling spectral distributions and patches:
Each division
is one octave.
To add an octave, you have
to multiply F x 2.
middle C
Pitch by octaves
4 Equal pitch intervals require exponentially increasing frequency increments;
4 Equal frequency increments make a harmonic series. In a harmonic series, as we go
up the series we get smaller and smaller pitch intervals.
CHAPTER 3 - The Craft of audio Synthesis
Signal Amplitude and Audible Volume
This is a very dicey relationship. It is true that, for any given signal, increasing its
amplitude will increase the volume of the associated sound. But a lot of other signal
characteristics have a greater impact on our perception of volume than amplitude does.
For example, the difference between talking and shouting at someone is far more a
matter of pitch and spectrum – tone-color – than of mere amplitude. When I yell, I “raise
my voice”; that is, I raise the pitch of my voice, and I put more energy into it, which
generates more harmonics, which is a matter of spectral content, not amplitude. Stage
actors have to learn to overcome this tendency; in order to be heard onstage without
shouting, they have to learn to increase their speaking amplitude without yelling.
If you examine – visually - the recorded signal from a pipe organ, or even an orchestra,
you may be surprised to find the apparent amplitude of the softer signals almost equal
to that of the loudest. This has to do with the spectral distribution of the sound. Of two
different signals of approximately equal amplitude, the one with the broadest spectral
distribution – the most harmonic content - will sound louder.
How Adding Volumes Means Multiplying Amplitudes
A 3dB rise or fall in sound level is noticeable, but it's not
much; it represents a factor of 2 change in signal power.
Power level
Power level
-15dB -12dB
-6dB Ref-3dB Reference
Sound level
sound level
Nonetheless, for a given signal and its
spectrum, it is always true that a bare increase
in amplitude will be heard as an increase in
loudness. But how much of an increase?
It turns out that, just as with frequency and
pitch, the relationship here is exponential.
A century of research has established that
equal steps in loudness are represented by
multiplicative ratios in signal amplitude.
In other words, if you double the amplitude
of a signal, you will hear an increase in
loudness. But then to get another increase
equal to the first one, you have to double the
amplitude again.
Signal Spectrum and Audible Tone-Color
This is a quite solid relationship. In fact, the word “spectrum” has achieved a meaning in
both worlds: depending on the context, it can refer to a measurable attribute of physical
signals, or a character of perceived sounds.
Any change you hear in the character of a sound – in its tone-color or “spectrum” - must
have an associated variation in the character of the physical signal that is arriving at
your eardrums. Tone change = waveform change.
The reverse is not quite so certain. It’s fairly easy to find waveform changes that listeners
can’t hear. For example, we humans simply aren’t sensitive to phase relationships within
a complex spectrum. But, in the time domain, two identical spectra with shifted phase
relationships among their components can look unrecognizably different.
CHAPTER 3 - The Craft of audio Synthesis
Signal Envelopes and Audible Event-Contours
One of the most fascinating areas of audio synthesis is listening for the envelopes of
time-varying events. Here there are all sorts of mysteries, in which signal attributes get
regularly “misperceived”: frequency variations get heard as volume, spectral evolutions
are heard as pitch, and amplitude envelopes generate an elusive spectral twitter.
Modules and Methods for Generating Signals
How can you get access to the signals and processes we’ve just described, so that you can
play with them on your own?
Throughout the past century people have been noodling around with electronic ways of
generating audio signals. In particular, beginning in the 1960’s, people such as Bob Moog,
Don Buchla, and Alan R. Pearlman began to settle on some ideas that have become
almost standard for audio synthesis: independent, modular functions for signal generation
and signal processing, capable of being controlled not only by hand but also by signals of
the same kind as they generate.
This was the idea of voltage-controlled operation, and it was completely revolutionary.
Using independent, modular functions made it possible to change one attribute of a
signal without necessarily affecting any other attribute; so the craft of synthesis became
the craft of constructing, and tuning, integrated configurations of modules.
The cables that connected modules were called patchcords, and so connecting modules
together came to referred to as patching them, and so, finally, any working configuration
came to be called a patch.
Let’s take a look at some modules that generate signals.
A device that repeats the same
motion over and over is an
oscillator. In audio synthesis,
oscillators typically produce very
simple geometrical-pattern signals
such as sine, triangle, pulse, and
sawtooth waves, named simply
for what their time-domain graphs
look like.
Oscillator Output
TIME - domain
FREQUENCY - domain
-3dB / Octave
4f 5 6 7 8f 10 13 16f
12f 16f
In an analog synthesizer, the
underlying medium in motion
is usually electrical pressure, or
voltage. In a digital synthesizer,
the signal is actually generated as a sequence of calculated numbers. (Conventionally,
these are generated at the standard sampling rate for music CDs, 44.1KHz.) This does not
become motion until, at the output of the synthesizer, the number stream is converted
to variations in voltage, and then amplified, and then used to drive a loudspeaker. (A
loudspeaker is a motor that moves back and forth instead of around and around.)
Because oscillator-generated signals are periodic, their spectral components always form
a harmonic series.
CHAPTER 3 - The Craft of audio Synthesis
Noise Generators
A device that jiggles at random without ever repeating itself
is a noise generator. Waterfalls, steam, wind, fans, and such
things are all noise generators.
The spectrum of a noise signal is a statistical distribution
of frequency components. (This is the opposite of a sine
wave, which is exactly one frequency.) A noise spectrum
that is perfectly balanced throughout the musical range is
called pink noise. Pink noise is very useful in listening tests
of loudspeakers, because a trained human listener can hear
even tiny differences between two different noise spectra.
Red (LF)
Envelope Generators
envelope generators are controlled by gate
and trigger signals...
A device whose output is intended to control some time-varying
attribute of an event is called an envelope generator. These are
sometimes referred to as transient generators, to call attention to
the fact that their output is not constant but transient.
An envelope generator produces an output signal only “on
demand”. The demand is made by means of timing signals called
gates, and triggers.
Filtering and equalization can shape a noise spectrum into
almost any sound.
Sample & Hold Processors
The idea of “sampling” a signal does not directly relate to any particular characteristic of
audio events; instead, it is an idea from electronics that has turned out to be useful for
creating patterned control signals.
Modules and Methods for Processing/Modifying Signals
A signal inverter works exactly like a seesaw. When the input goes high, the output goes
low, and vice versa. An analog inverter would output negative voltages on positive input;
a digital inverter simply multiplies its input number-stream by (-1).
CHAPTER 3 - The Craft of audio Synthesis
Signal Mixing
A signal mixer adds two or more signals together and outputs the result of the addition.
This is a more complex signal, usually, than any of the inputs. But not necessarily; if
signal B, for example, is the exact inversion of signal A, then mixing the two will produce
a signal of exactly zero.
these four
signals added
together make....
two sines
added together
An attenuator cuts the strength of a signal passing through. Digitally, this is accomplished
by multiplying the input by some value ranging from 0.0 (which passes no signal) to 1.0
(which passes the signal at its full input amplitude).
An amplifier may increase the amplitude of a signal. But not necessarily; it is usual for a
voltage-controlled amplifier to have a maximum gain factor of 1.0. In fact, the purpose of
VCA’s is actually to “chop” the signals passing through them. It would make more sense
to think of them as “voltage controlled attenuators”.
Gain Factor
To describe the behavior of an amplifier or attenuator, we may use the expression “gain
factor” to mean the ratio of output signal amplitude to input amplitude.
A filter is a device that works better at some frequencies than at others. (The inverters,
mixers, and attenuators we have been describing work the same at all frequencies, so
they are not filters.)
Because of this frequency-dependent characteristic of filters, they change the shape of
any complex waveform passing through. And so it will be important for you to get to
know what filters do to signals in both the time domain and in the frequency domain.
CHAPTER 3 - The Craft of audio Synthesis
Low-Pass Filters
Any device or mechanism that passes along slower motions better
than faster ones can act as a lowpass filter.
16f 32f
A lowpass filter in
the time domain . . .
Picture yourself stirring a cup of tea with one of those little
wooden paddles they hand out in the coffee shops. Stir it back and
forth, fast. Now slow down. Now imagine the tea has turned to
syrup. You can still stir it slowly, but if you try to go fast the stick
will simply not move.
That’s a lowpass filter. You can see the effect of this on a signal
quite easily.
For audio signals, you will usually be more interested in the
frequency-domain effects of filtering. For subaudio signals, it is
usually the time-domain effects – changes in waveshape – that
we care about. In the time domain, a low-pass filter rounds off
any sharp transitions in the signal. A good example of this is the lag processor described
in section below.
. . . makes it difficult
for the medium to
change value.
. . . and in the frequency
domain, this cuts off
In the frequency domain, it weakens spectral components that are higher than the filter
cutoff frequency – the frequency at which the filter begins to have an effect on the signal.
High-Pass Filters
Any device or mechanism that passes along faster motions better
than slower ones is a highpass filter.
Take the drinking straw from your water glass. Seal the end of it
with your thumb and dip it back into the water. Notice that you
can pump it up and down in the glass, fast or slow, but the water
never leaks into the straw as long as you hold your thumb over the
end. Think of the up and down motion of the water at the bottom
end of the straw as “the signal”.
A highpass filter makes
it difficult for the medium
to keep a new level - it
"leaks" away . . .
. . . and so lowerfrequency components
are weakened.
Now start letting a little air leak into the straw as you move it
up and down. The water level at the bottom end of the straw no
longer stays down when the straw “signal” goes down – it starts
to come up again. And then when you draw the straw back up,
the water leaks back down more or less rapidly depending on the
position of your thumb at the top of the straw.
This is a high pass filter. In the time domain, it constantly “leaks” its output signal level
back to zero, at a rate related to the cutoff frequency and slope. In the frequency domain,
it passes all spectral components higher than the cutoff frequency, and attenuates those
below the cutoff frequency by an amount proportional to the cutoff slope.
CHAPTER 3 - The Craft of audio Synthesis
Cutoff Slope
This is the rate at which a filter attenuates spectral components, as a function of their
frequency. It is usually a multiple of 6dB/octave.
Feedback and Resonance
It is usually possible, with any signal-processing device or system of devices, to mix some
of its output signal back into the input signal. This may be intentional, or it can happen
by accident; everybody who has ever worked with a PA system has experienced the
terrible screech of a system “in feedback”.
In filter modules for audio synthesis, it is common to provide controllable feedback. With
just a little, the filter response begins to peak around its cutoff frequency; as the feedback
level increases, the peak gets stronger. Eventually – just as happens with a PA system
– the filter falls into oscillation. Regardless of the input signal, it “screams” a sine-wave at
its cutoff frequency. In this state it is no longer behaving as a filter at all; it has become
an oscillator.
Systems in feedback have been very well studied in physics. Their behavior can be
described mathematically. Whereas feedback in a PA system can be unpredictable
and uncontrollable, feedback in a filter module for audio synthesis can be – and is – a
controllable and useful feature.
Lag Processors
A lag processor is a low-pass filter intended specifically for processing subaudio signals. It
introduces a “lag” in the output signal wherever the input shows a sharp change in value.
How much of a lag depends on the “cutoff frequency” of the processor.
Modulation Methods
The simplest possible signal, we said above, has exactly three attributes and no more:
frequency, amplitude, and phase. If, starting from a steady-state signal, we systematically
modify any of these characteristics, we are said to be modulating the signal. And so,
based on these three signal attributes, there are three possible forms of modulation:
Amplitude Modulation (AM), Frequency Modulation (FM), and Phase Modulation (PM).
The first two are more commonly used in audio synthesis than the third; we won’t say
anything here about phase modulation.
Using AM and FM methods, it is possible to generate waveforms and spectra that are far
more complicated – and interesting to the ear – than anything that can be produced by
merely mixing and filtering signals. Mainly that is because of sidebands.
CHAPTER 3 - The Craft of audio Synthesis
Sidebands and Sideband Spectra
What happens to the spectrum of a sine wave when we modulate its amplitude? What
happens when we modulate its frequency?
Clearly, since the signal that results from AM or FM methods is no longer a plain vanilla
sine wave, then, in the frequency domain, it must have some additional components.
These additional components are called sidebands. They have been studied for at least a
century, and are pretty well understood physically and mathematically.
Audio synthesis is probably the only application of AM or FM modulation where we
are interested in sidebands for their own sake, as something to listen to directly; in the
past, this stuff was only interesting to radio engineers, radar, sonar, television broadcast
engineering. In those disciplines, modulation sidebands are products of broadcasting
methods, in the electromagnetic spectrum. Because of that historical background, some
of the conventional language for talking about modulation processes is a little weird:
the signal being modulated is often referred to as the carrier signal
signal, and the signal that
provides the modulation pattern is called the program. (Guess why.)
Given this for a carrier:
. . . and this for a program
here's the AM result:
here's the ring modulation:
the carrier
is gone
and here's the FM result:
number of
and varying
on the
Any form of modulation generates, for each component of the
original signal, at least one lower sideband – at a frequency equal
to the component minus the modulating frequency – and at least
one upper sideband, at a frequency equal to the component plus
the modulating frequency. If the carrier – the original signal - is
itself complex, with multiple spectral components, then each of
its components will produce its own sidebands independently of
all the others. Likewise, if the modulating signal – the program
- is complex, the arithmetic applies separately to each of its
So, just for example, if you modulate a 10-component carrier signal
with a 10-component program signal, the signal resulting from the
modulation will have not less than 100 spectral components. This
can get very messy; the most useful thing you can do with such a
signal, before you do anything else, is filter it to get rid of some of
the fuzz.
Amplitude Modulation
Suppose we modulate the amplitude of a 1000Hz sine wave with a 5Hz sine wave. The
result is indistinguishable from what we would get if we mixed three sine waves, at
995Hz, 1KHz, and 1005Hz. They are the same signal.
The 995Hz component of the output is the lower sideband resulting from the modulation,
and the 1005Hz component is the upper sideband.
CHAPTER 3 - The Craft of audio Synthesis
Ring Modulation
Whereas a VCA responds only to a positive-going signal at its amplitude-control input,
a ring modulator responds to both positive and negative levels at both of its inputs. Its
output is simply the product, arithmetically, of the two inputs. If you are new to audio
synthesis, draw a couple of signal graphs – it doesn’t matter what they are – on the
same timebase and vertical scale, and use a pocket calculator to work out the result of
multiplying the two signals together. That’s what a ring modulator does. (The expression
“ring modulator” describes the appearance of the analogue circuit design that’s required
for the multiplication.)
In the frequency domain, the difference between this and ordinary AM is only that the
carrier signal components are suppressed. Once again, if you work out the arithmetic, a
single-frequency carrier, modulated by a single-component program, generates a threecomponent AM spectrum but only a two-component ring-modulation spectrum.
What’s useful about this? Well, since the carrier is almost always periodic (it comes from
an oscillator, right?), it has a harmonic spectrum. Suppressing this spectrum lets you hear
just the sidebands, which can be completely enharmonic if you’re careful about the ratio
of the two input signal frequencies.
Frequency Shifting
It’s theoretically possible not only to suppress the original carrier, as in ring modulation,
but to isolate the lower and upper sidebands and make them available separately.
The arithmetic here is fascinating, because the end result (for once) is in one-to-one
correspondence with the input: for each component of the program signal, there is a
component in the output at C-p (or at C+p). In other words, the final spectrum has only as
many components as the original program did. This is called frequency shifting. Picture
the entire program signal spectrum shifted up or down by some fixed frequency.
The important thing to remember about this is that it’s not pitch shifting – which would
have to be accomplished by frequency multiplication – but frequency shifting. It’s an
addition or subtraction process, and it really messes up any harmonic relationships that
might have existed in the original spectrum.
Frequency Modulation
The spectrum resulting from amplitude modulation always has three components for
every one component of the program signal: the carrier itself, and two sidebands. In
Frequency Modulation, however, the number of sidebands depends on the modulation
depth. It is possible from only two sine waves to generate a spectrum with dozens or
even hundreds of components. Modulating one sawtooth with another can produce a
spectrum so complex that it sounds almost like a noise generator. In such a patch, you
will usually reach for a filter to take the edge off the resulting spectrum.
What happens is this: as the depth of modulation increases, the number of sidebands
does too, without limit. The additional sidebands come in at – guess what – integral
multiples of the program frequency.
For this reason, the most useful FM techniques involve only sine-wave carrier and
program signals.
CHAPTER 3 - The Craft of audio Synthesis
Controlling One Module by Means of Another
The modulation methods we’ve just described can be accomplished with voltagecontrolled or digitally-controlled equipment. For example, to set up an AM effect,
feed the carrier signal into a VCA audio input, and the program signal into one of the
amplitude-control inputs.
Likewise, to set up an FM method, route the program signal into one of the frequencycontrol inputs of a VCO. (See section 4.3 for news of some TimewARP 2600 extensions
relating to this.)
Linear and Exponential Sensitivity To Control Signals
In designing and constructing a voltage-controlled device, or a digitally-controlled
algorithm, we have to consider how we intend the controlling parameter to relate to the
controlled parameter.
Suppose, for example, we build an oscillator that changes its output frequency by exactly
1000Hz for each rise of 1 volt in electrical pressure at a designated point (a “control input”)
in the oscillator circuitry. This is a linear relationship between the control value and the
frequency. In order to march such an oscillator through a musical scale, we would have
to provide exponentially increasing voltage steps for each rising musical interval. That’s
pretty awkward, and it gets worse very rapidly. The pitch interval produced by, say, a 0.5volt control step will depend – completely – on the initial frequency of the oscillator.
Suppose we tune this imaginary linear-responding VCO to 500Hz, and apply a sequence
of 0.5V control steps. What frequency do we get at each step? What musical interval?
Major third
Minor third
And so, instead of linear sensitivity, VCO’s for audio synthesis are designed, almost
universally, with exponential sensitivity to control signals. Such a design sets up an
exponential relationship between control signal and frequency in order to maintain a
simple linear relation between control signal values and audible pitch changes.
Similar arguments apply in the design of voltage-controlled filters: it’s more practical
to work with a linear relationship between control-signal and “cutoff timbre”, and
therefore the relationship between control-signal and cutoff frequency really has to be
Modular Components
of the TimewARP 2600
Top Row Control Panel Buttons and Indicators
Just above the panel graphics, outside the “case” of the TimewARP 2600, is a horizontal
row of buttons and indicators for patch storage, import/export, voice-cloning, and other
These powerful features of the TimewARP 2600 have no equivalent in the world of analog
synthesis; they are unique digital extensions of the original ARP 2600 synthesizer.
Patch Lock Button (Padlock Icon)
This padlock button helps you to avoid accidentally overwriting your favorite patch.
Basically it disables the Save button, while leaving the Save As button enabled. Under
these conditions, you can save your current patch only by assigning it a new name.
Group, Category, and Patch Drop-Down Lists
The TimewARP 2600 gives you a three-level hierarchy for storing and organizing your
patches. All Patches are sorted into various Categories, which are in turn sorted into major
Groups. Each of the three patch selection buttons generates a drop-down list associated
with one layer in this hierarchy.
Groups, Categories, and Patches can also be selected by keyboard shortcuts. The up/down
arrow keys on the computer keyboard select Patches, the left/right arrow keys move
between Categories, and using the control key with the left/right arrow keys moves
between Groups.
Save Button.
The Save button saves the current patch configuration and settings under the name of
the most recently loaded patch. This button is disabled if the patch is locked (see 4.1.1
CHAPTER 4 - Modular Components of the TimewARP 2600
Save As Button.
The Save As button saves the current patch configuration and settings under a group,
category, and patch name of your choice.
Within the Save As dialog, you may create new groups and categories at will.
There is no limit to the number of groups and categories you may create.
Patch Manager Button
Use the Patch Manager to organize
and reorganize your patches; to move
patches from one category to another,
and to move whole categories from one
group to another. Use it to export and
import patch collections – dozens or
hundreds of patches at a time.
The Patch Manager window displays
all three levels of the patch hierarchy,
and supplies a number of tools for
managing the entire collection.
These tools are listed in a column on
the left of the window.
To use them, select one or more items from the hierarchy, and then click on the operation
you want to perform. Any operation that cannot be applied to the current selection of
items will be disabled in the list.
CHAPTER 4 - Modular Components of the TimewARP 2600
Import / Export
Use the Import / Export commands to write or read entire Groups, Categories, and Patches
to/from external files.
Export works on whatever items are currently selected; by selecting all of the current
groups, you can use Export to make a backup of all of your patches.
When you export a single Patch, the names of the Group and Category for the patch are
exported with it.
Import asks you to select the file you want to read in. Import will never overwrite any
existing Group, Category, or Patch; if any Group or Category or Patch in the file to be
imported has the same name as a Group or Category or Patch that already exists, Import
will append a number to the loaded name. So when a friend sends you his collection
of a thousand patches, you can import them without worrying about possible name
Cut / Copy / Paste
The Cut / Copy / Paste buttons appear within the Patch Manager dialog to move items from
one place in the hierarchy to another. You can, for example, Cut a patch, or a group of
patches, from one category and Paste it into another.
Up / Down
The Up / Down buttons appear within the Patch Manager dialog to move selected items Up
or Down in their list. (This is useful for arranging your MIDI patch lists.)
Rename / Delete
The Rename / Delete buttons appear within the Patch Manager dialog to Rename or Delete
selected items.
Voice Button
Clicking on the Voice button activates a drop-down list from which you may select the
number of simultaneously sounding voices you want to use.
Because the TimewARP 2600 is a true analog synthesizer emulator, its modules are
running even when no audio signals appear at the synthesizer’s output. Each voice added
to the multi-voice capability of the TimewARP 2600 is a clone of the entire patch and
module set. This will have an immediate and obvious effect on the CPU load meter.
So: how many voices you can generate without overtaxing your CPU will depend on your
machine’s clock speed.
CHAPTER 4 - Modular Components of the TimewARP 2600
Reset Button
The Reset button removes all patch cords and returns all sliders to a standard position.
MIDI Indicator
This virtual LED glows when there is any MIDI input to the TimewARP 2600 – not just
keystrokes, but also controller input and sysex dumps.
Output-Level Meter
This shows the output signal level. If it reaches into the red segment, your signal will
CPU Load Meter
This meter shows, roughly, how much of the time between samples (the sample period) is
being devoted to the TimewARP 2600 emulation process. In a complex patch, or a manyvoice polyphonic performance, the meter may indicate overload; when this happens, it
is likely that the TimewARP 2600 output signal will be interrupted, so your audio feed
will develop a glitch. To avoid this, you will have to simplify your patch, or decrease the
number of voices, or acquire a faster, more capable computer.
The Magic Logo
At the lower right of the main panel is the TimewARP 2600 Logo.
Clicking on this brings up a menu:
About TimewARP 2600 identifies the team; the people who worked together to bring
you this software.
Load/Save MIDI Maps
Use the Load/Save MIDI Maps commands to save
– and reload – the MIDI-controller to slider assignments
that you set up. In the TimewARP 2600, these are
global assignments, independent of any particular
patch settings; saving a patch does not save these
assignments, and loading a patch does not change the
current assignments. You can, if you want, set your
mappings once, and they will be there throughout all of
your personal patch changes.
CHAPTER 4 - Modular Components of the TimewARP 2600
Load Microtuning
You may also load alternate tunings for the keyboard.
These are described in Appendix 6.1. The TimewARP
2600 does not allow you to modify these tunings or to
save or create new ones.
Microtunings are a global attribute of the keyboard;
once loaded, the tuning will govern anything you play
until you load a different one, regardless of your patch
MIDI Beat Synchronization
You may synchronize the Internal
Clock (IC) (see section 4.13), to
the MIDI Beat Clock (MBC) by
specifying the number of MBC
pulses per IC transition. As a
reference, there are 24 MBC
pulses per quarter note.
The keyboard LFO (see section
4.14.1) may also be synchronized
to incoming the MBC, independently of the Internal Clock.
Setting different sync counts for these is a fun way to program complex rhythms that are locked to
the tempo of your MIDI tracks.
In order for the MBC messages to be sent to the TimewARP 2600, you must enable MIDI Beat Clock in
the Pro Tools MIDI Menu, and select the TimewARP 2600 as a recipient of these messages. Also, MBC
messages are only sent when the Pro Tools transport control is running.
MBC synchronization is a patch attribute, not a global one; the sync counts you set here will be stored
with the current patch when you save it.
CHAPTER 4 - Modular Components of the TimewARP 2600
Jacks, Patchcords, and Default Connections
The panel has eighty-one mini-jacks. Forty-five are inputs, twenty-nine are outputs, and
7 operate as both input and output.
Of the 45 inputs, 32 are in a row running across the center of the panel. (There are
actually 34 jacks in the row, but the two labeled “gate” and “trig” are outputs.) This row
of input jacks divides the control surface almost evenly in half.
Above this row, in the upper half of the control surface, there are only three input jacks.
They are at the upper right corner, labeled Left Input, Pan, and Right Input respectively. All
of the other jacks in the upper half of the control surface are outputs.
In the lower half of the control surface are inputs to the voltage processors, and of the
column of four jacks in the section labeled Sample & Hold, the upper and lower jacks are
The seven jacks that are both input and output, belong to the Electronic Switch and the
Multiple outlet. Because the switch works in either direction, it has either two inputs and
one output or one input and two outputs. The Multiple output distributes at least one
input to 1, 2, or 3 outputs.
All the remaining jacks are outputs. Most of them are labeled as such; a few are not, but
have arrows pointing to them. For example, in the Voltage Processor (VP), the three jacks
furthest to the right are outputs; and in the Envelope Generator section of the upper half
of the panel, the two jacks labeled Gate and Trigger are outputs.
Sliders and Slider Operations
There are fifty-eight sliders. Thirty-six of these are plain old signal attenuators. And of
those 36, 29 are all in a row across the middle of the panel. (There are actually 31 sliders in
the row, but the two on either side of the box labeled Attack Release are not plain old signal
Most of the attenuators are directly associated with either an input to something or
an output from something. Each vertical attenuator across the middle of the panel, for
example, adjusts the strength of the signal coming in from the input directly below it.
Normalled Jacks
The most commonly used signal connections are “normalled” (i.e. defaulted). A default
signal is identified by a small icon at an input jack.
To see a simple example of this, note that the first input jack on the left, in the row
of jacks running across the middle of the TimewARP 2600, is an input to the Envelope
Follower. The symbol underneath this jack indicates that the default signal to this input
comes from the Preamp. That means that the Preamp output is prewired to the Envelope
Follower input, except when a plug is inserted into the jack.
For another simple example, note that in the same row of jacks, the third one from the
right is a mixer input, and that the symbol just beneath it indicates that the default signal
to this input comes from the Voltage-Controlled Filter (VCF):
Note again that the fifth of the five audio inputs to the VCF is similarly defaulted from the
Noise Generator (counting across from left to right this is the 21st jack):
CHAPTER 4 - Modular Components of the TimewARP 2600
So you can listen to this input by opening the VCF input to the mixer, and the Noise
Generator input to the VCF. Now experimenting with the two horizontal control sliders at
the top of the VCF panel will give you a wide range of filtered sounds.
It will be worth your while to experiment thoroughly and systematically with the default
signal connections at this point, particularly if you are planning to use the TimewARP
2600 in live performance. In section 4 we will document the behavior of each separate
module, and in section 5 we give sample patches for further experimentation; here we
will only mention a few general principles to keep you from going out of your skull with
Experiment with one signal at a time. With the VCF, for example, when you have listened
to everything the filter can do with a noise input, close that input and open the default
VCO-3 sawtooth immediately to its left. Now you can experiment not only with the VCF
controls, but also with the manual frequency controls of VCO-3; and when you have done
that, experiment one by one with the control input signals to VCO-3.
Preamp/Gain Control
The Preamp section controls the gain of the audio signal(s) from the track in which
the TimewARP 2600 is running. A rotary knob labeled Gain adjusts the signal
If the TimewARP 2600 is running in full stereo configuration – as a plug-in to
a stereo track – the preamp will display two output jacks, one for each stereo
channel. Use these signals for any purpose for which you might ordinarily use an
internally-generated signal. You can filter them, run them through the Ring Modulator, or
use one as an AM or FM program signal. The default input to the Envelope Follower, under
these conditions, is taken from the left channel.
Voltage-Controlled Oscillators (VCO)
These generate three or more of the following basic waveforms. The output amplitude
and phase relationships are the same for all oscillators. The oscillator sensitivity under
virtual voltage control is 1vV/octave.
For convenience in fine-tuning control depth, the three attenuator-governed FM Control
inputs at each oscillator provide three different sensitivity ranges. The leftmost slider is
full-range; wide open, it passes its signal unchanged. The second slider is 50%; wide open,
it passes its signal at half strength. The third slider, wide open, passes its signal at 25% of
its original amplitude.
CHAPTER 4 - Modular Components of the TimewARP 2600
VCO 1 generates saw, square, and sine outputs. The sine output is a TimewARP
2600 extension; the original ARP 2600 VCO1 provided just sawtooth and
squarewave outputs.
The default signal to the first (unattenuated) FM Control input is from the
keyboard. The Audio/LF switch above this input switches the mode of the VCO
from Audio (10Hz - 20,000Hz) to LFO Mode (0.03Hz – 30Hz). When the VCO is in LFO
Mode, the default connection to the keyboard is removed. This can be overridden
in this mode by patching a cable to the Keyboard CV output on the left side of the
front panel.
The default signals to the next three FM Control inputs are from a) the Sample &
Hold, b) the ADSR Envelope Generator, and c) VCO2 sine.
VCO 2 generates sine, triangle, sawtooth, and pulse outputs. A pulse-width slide
control can adjust the duty cycle from 10% to 90%; at the middle of its travel, the
pulse width is 50%, that is, a square wave.
The default signal to the first (unattenuated) FM Control input is from the
keyboard. The Audio/LF switch above this input switches the mode of the VCO
from Audio (10Hz - 20,000Hz) to LFO Mode (0.03Hz – 30Hz). When the VCO is in LFO
Mode, the default connection to the keyboard is removed. This can be overridden
in this mode by patching a cable to the Keyboard CV output on the left side of the
front panel.
The default signals to the next three FM Control inputs are from a) the Sample &
Hold, b) the ADSR Envelope Generator, and c) VCO1 square.
There is a fourth attenuator-governed input, for digital control of the pulse width.
The default signal at this PWM input is from the Noise Generator.
CHAPTER 4 - Modular Components of the TimewARP 2600
VCO 3 generates sawtooth, pulse, and sine outputs; the pulse width is manually
variable. The sine output is a TimewARP 2600 extension; the original ARP 2600 VCO3
provided just sawtooth and pulse outputs.
The default signal to the first (unattenuated) FM Control input is from the keyboard.
The Audio/LF switch above this input switches the mode of the VCO from Audio (10Hz
- 20,000Hz) to LFO Mode (0.03Hz – 30Hz). When the VCO is in LFO Mode, the default
connection to the keyboard is removed. This can be overridden in this mode by
patching a cable to the Keyboard CV output on the left side of the front panel.
The default signals to the next three FM Control inputs are from a) the Noise Generator,
b) the ADSR Envelope Generator, and c) VCO2 sine.
Voltage Controlled Filter (VCF)
The Voltage Controlled Filter has variable cutoff frequency (Fc) and
resonance (Q). The response below Fc is flat down to DC; above Fc the
response falls off at 24Db per octave. Fc range is from 10Hz to 10KHz
without control voltages; under voltage control, Fc can be driven as far
down as 1 Hz and as high as 20KHz.
Fc is controlled manually by a coarse tuning slider (labeled initial filter
frequency) and a fine tune slider. Fc may also be controlled by external
voltages; the sensitivity under voltage control is 1.0vV/oct.
The Q, or resonance, of the filter circuit is controlled by a single
manual slider. As the Q is increased by moving this slider from left to
right, the response below Fc is gradually attenuated until a sharp peak
remains at the cutoff frequency. (Gain at Fc is always unity.)
At this Q setting, just below the point at which oscillation begins,
the filter will ring distinctly in response to any sharply defined pulse
presented to its signal input. In this state it is effectively analogous to
a highly resonant physical system, and may be used for various percussion effects
depending on its resonant frequency (identical to Fc) and on the impulse spectrum
exciting it.
As the Q is raised still higher, beyond about the halfway point in the slider travel, the
filter will oscillate. Operating in this state, it generates a pure sine wave. even in the
absence of any signal input.
The VCF has five Audio signal inputs. They are fed through logarithmic attenuators to a
summing point, and then to the VCF itself. The default input signals are from the Ring
Modulator, VCO-1 Square, VCO-2 Pulse, VCO-3 Sawtooth, and Noise Generator.
The VCF has three frequency Control inputs. The first is normally from the Keyboard pitchcontrol. The slider that governs this input is a TimewARP 2600 extension; on the original
CHAPTER 4 - Modular Components of the TimewARP 2600
ARP 2600, the keyboard control depth was not adjustable.
The second and third FM Control inputs are governed by linear attenuators; prewired to
these are the ADSR Envelope Generator output and the VCO-2 Sine output.
Inserting a patch cord at an input jack automatically disconnects the default signal.
Envelope Generators
The Envelope Generators generate transient, positive-going waveforms, with
controllable rise and fall times. They are used primarily with the VCF and VCA, in
generating events whose time-varying spectrum and amplitude must be accurately
and repeatably controlled. The output from each generator is a positive-going signal
whose rise and fall time is set by slide controls on the generator, and whose onset and
overall duration is determined by a gate signal.
The maximum value that either envelope can reach is +10vV; thus, unattenuated,
either envelope is capable of driving a VCF or VCA from its minimum initial setting
(10Hz for the VCF, -l00Db for the VCA) all the way up to maximum. See 4.1.2 and
4.1.3, specifically the data on control input sensitivity. Reread too sections 2.1.6
through 2.1.7.
Gate signals for operating an Envelope Generator may originate with a Manual Start
button, the Keyboard Controller, or any +10vV square-wave or pulse signal. The twoposition switch just under the lower AR generator selects between the two latter
sources. The Manual Start button overrides both of these.
ADSR Envelope Generator
The ADSR Envelope Generator offers variable Attack time, initial Decay time, Sustain level,
and final Release time. Four vertical sliders control these parameters: note that three of
these are time parameters and the fourth – Sustain level – is not.
The generator produces an output only when a gate signal is present at its input. At the
onset of a gate signal, the output level rises to +10vV, at a rate set by the Attack slider.
When the level reaches +10vV, it immediately begins to fall, at a rate set by the Decay
slider, to a level set by the Sustain slider. It remains at this level while the gate signal
continues. Finally, when the gate ends, the output falls to zero at a rate set by the final
Release slider.
If the gate ends at any time during the ADSR cycle, the generator immediately advances
to its final Release phase: the current output level starts falling to zero from whatever
value it had at the moment the gate ended.
When the Envelope Generator is controlled from the Keyboard Controller, it responds to the
trigger signal from the keyboard – supposing that an envelope is already in process – by
repeating the first two stages of an envelope and returning again to the Sustain level.
For all other control gates presented to the generator input, circuitry internal to the
generator itself derives a trigger signal from the leading edge of the gate.
CHAPTER 4 - Modular Components of the TimewARP 2600
AR Envelope Generator
The AR Envelope Generator offers variable Attack time and final Release time. Two vertical
sliders control these parameters.
The generator produces an output only when a gate signal is present at its input. At the
onset of a gate signal, the output level rises to +10vV, at a rate set by the Attack slider.
When the level reaches +10vV, it remains at this level while the gate signal continues.
Finally, when the gate ends, the output falls to zero at a rate set by the final Release
If the gate ends at any time during the AR cycle, the generator immediately advances to
its final Release phase: the current output level starts falling to zero from whatever value
it had at the moment the gate ended.
The AR Envelope Generator does not require a trigger signal for any phase of its
Voltage Controlled Amplifier (VCA)
The Voltage Controlled Amplifier has a maximum gain of unity and a dynamic range
of l00Db. With the initial gain control at maximum, and with no control input, the
VCA will pass with unchanged amplitude any signal presented to its signal input. On
the other hand, with the initial gain control at minimum, no signal will pass through
the amplifier at all unless some positive signal level (the VCA does not respond to
negative control signals) is present at one or both of its control inputs.
The first control input has linear sensitivity; the gain of the amplifier in response
to a signal at this input is S/10, i.e. dividing the signal level by 10 will give the gain
The second control input has exponential sensitivity; the gain of the amplifier in
response to a signal at this input will equal 10Db/vV.
There are two audio signal inputs to the VCA. The default connections to these are
from the VCF and the Ring Modulator. Inserting a patch cord automatically disconnects
the default signal.
The default signal at the linear control input is from the AR Envelope Generator, and
at the exponential input is from the ADSR Envelope Generator. Inserting a patch cord
automatically disconnects the default signal.
CHAPTER 4 - Modular Components of the TimewARP 2600
Mix/Pan/Reverb Output Module
The three functions in this module provide final processing of the output signal.
That, at least, is what they are intended for; you may actually use them in other
roles, for any purpose you please.
If you leave the default connections undisturbed, the module is configured as a
two-input Mixer, which feeds a Pan Control and a Reverb unit, which are themselves
mixed to feed the final left/right system output channels.
When the TimewARP 2600 is configured for mono operation, this section omits the
Pan control and provides only one output channel.
The two inputs to this Mixer carry default connections from the VCF and the VCA; they can
of course be overridden with patch cords.
The two jacks just above the sliders and below the Mixer graphic are not inputs; they
are the outputs from the attenuators. This lets you use the two sliders as “floating”
attenuators, in any situation where you need to set the strength of a signal. (Although if
you do this, there’s no way to get any signals into the mixer.)
Pan Control
The Pan Control takes its input from the jack just below the horizontal pan slider.
Normally, this signal comes from the mixer. Centered, the Pan feeds its input signal
equally to the left and right channel outputs; moving the slider left or right shifts the
signal balance accordingly between the two output channels.
Reverb Unit
The input to this unit is the rightmost jack in the row that runs across the middle of the
panel. By default, it carries the Mixer output. The output jack, to its upper right, provides
a 100% wet signal from the Reverb, at a fixed level. (There are interesting patches in
which this signal is subjected to further processing via, say, the Ring Modulator or the
Envelope Follower.)
The two sliders adjust the wet-dry mix fed to each output channel.
CHAPTER 4 - Modular Components of the TimewARP 2600
Envelope Follower
The Envelope Follower generates, from any audio-frequency input, a fluctuating DC output
level directly proportional to the average moment-by-moment input signal amplitude. Its
sensitivity is such that, with the input attenuator wide open, a 1V P-P square wave will
produce a +10vV output. The maximum output is +10vV.
The risetime, or time it takes for the Envelope Follower to respond to any sudden change
in the amplitude of the signal input, is 10 milliseconds to 50% of final value and 30
milliseconds to 90% of final value.
Like all similar circuitry, the Envelope Follower tends to “ride” on low audio frequencies
as if they themselves represented changes in signal amplitude; this is not critical, but
has been held to a ripple of less than 1% P-P down to l00Hz and less than 10% down to
The primary use of the Envelope Follower is with external instruments. Essentially it extracts,
from any audio input, a control signal representing the amplitude-envelope of that input:
this signal may control the VCF, VCA, or any of the VCO’s. The Envelope Follower output is an
envelope and can be used in the same fashion as the output from either of the envelope
The default input to the Envelope Follower is from the preamplifier. When the TimewARP
2600 is configured for stereo input, it is the first (left) channel preamplifier output.
CHAPTER 4 - Modular Components of the TimewARP 2600
Ring Modulator
The Ring Modulator is essentially a multiplier; from its two inputs A and B it produces the
output function A x B / 5. The kind of transformation this effects on input signals depends
to a large extent on what they are and on whether the modulator is AC or DC coupled to
them. This is selected by the Audio/DC switch at the bottom of the modulator.
When the inputs are AC coupled ((Audio position of the switch), any DC component
present in them is canceled before they are fed to the modulator. Thus a sawtooth that
starts from zero and goes to+l0vV will instead start at -5vV and move to +5vV so that
its overall positive and negative deviation cancels to zero. Under these conditions the
modulator will generate from any two periodic signals an output signal consisting of the
sum and difference frequencies that can be generated from the frequencies of the two
inputs. The input frequencies themselves will be suppressed.
If both signals are audio-frequency, a large variety of harmonic and inharmonic timbres
can be produced from the modulator, depending on the ratio of the input frequencies
and on their own harmonic content. If A is a sine wave and we represent its frequency by
Fa, and B is a complex waveform of frequency Fb with overtones 2Fb, 3Fb, 4Fb, etc., then
the output of the modulator will be a complex waveform with frequency components
Fb + Fa, Fb - Fa, 2Fb±Fa, 3Fb ± Fa, 4Fb ± Fa, etc. A moment’s experimentation with the
prewired sawtooth and sine inputs to the modulator will demonstrate the complexity of
the timbres that can be generated by this simple means.
If, still with AC coupling, one input is subsonic and the other at some audio frequency,
there will be an output from the modulator only when the value of the subsonic input
is changing, and the output will be roughly proportional to the rate of change. If, for
example, the subsonic input is a square wave, the modulator output will be a series of
short, decaying tonebursts – one at each rise or fall in the input signal.
When the inputs are DC coupled, any DC component in either one of the inputs will
pass into the modulator and affect the modulating process. The effect when both
inputs are at audio frequency is to allow into the output waveform some of the input
frequencies in addition to the sum and difference frequencies. The effect when one of
the inputs is subsonic is that the modulator operates as a voltage-controlled amplifier:
the output amplitude will be in direct proportion to the instantaneous amplitude of the
low-frequency input and will vary as its absolute value varies. Also, the output phase will
reverse when the low-frequency input signal changes from positive to negative or vice
The AC-coupling time constants are 235 msec, for the left input and 90 msec, for the right
CHAPTER 4 - Modular Components of the TimewARP 2600
Noise Generator (NG)
The Noise Generator has two manual controls: one for spectral balance (“color”) and one
for output level.
The spectral balance is continuously variable from white to red (low-frequency noise
output). In the latter case the output falls off at the rate of 6Db/Octave; the pink noise
position approximates a -3Db/Octave slope.
The level control, at minimum, cuts off the output signal completely. At maximum, the
output is clipped at 20vV P-P to produce binary, or two-valued, noise. Clipping begins
with the level control approximately half open. input.
Voltage Processors (VP)
The Voltage Processors are simple utility functions for mixing,
inverting, and shaping signals.
VP #1
VP #1 has four signal inputs and one output. Two of the inputs have attenuators. The
output signal is the inverted sum of all four inputs.
The attenuator-governed inputs carry default connections from +10vV and from the
keyboard pitch-control. Opening the +10vV slider thus produces up to -10vV at the
processor output.
VP #2
VP #2 has two signal inputs and one output. One of the inputs has an attenuator. The
output signal is the inverted sum of the two inputs.
The attenuator-governed input carries a default connection from -10vV, so that opening
the slider produces up to +10vV at the output.
The Lag Processor
The Lag Processor is a low-pass filter for processing control signals. The slider adjusts
its cutoff frequency. The corresponding rise-time ranges from 0.5ms with the slider at
minimum, to 500msec – about half a second – with the slider at maximum.
The Lag Processor can be used to process audio signals, as a -6Db/octave manual filter
with a maximum Fc of approximately 1KHz.
CHAPTER 4 - Modular Components of the TimewARP 2600
The Sample & Hold Module (S/H)
The Sample & Hold Module produces stepped output signal levels, by sampling
the instantaneous value of any signal at its input. The stepped levels produced
in this manner are useful for controlling oscillator and filter frequencies and
– occasionally – VCA gain.
The S/H circuit has a signal input (the waveform to be sampled), a trigger input,
and an output giving the result of the sampling operation. The trigger input is
defaulted from the internal clock, but any square or pulse wave, or the keyboard
gate or trigger signals, will work.
Upon being triggered, the S/H sets its output level to the same value as the input signal
at that instant. After the trigger, the output signal will hold that level until the next
trigger pulse.
Any signal whatsoever may be sampled. The default input is from the Noise Generator,
so that the step sequence is random. The accompanying diagrams show how, when
the signal being sampled is random noise, the output voltages are correspondingly
unpredictable. An infinite variety of cyclical output patterns may be obtained, on the
other hand, by sampling a periodic waveform. Different ratios of the sampling frequencies
to the frequency of the waveform being sampled create different melodic patterns (if the
output level is controlling a VCO).
The level control attenuates the input signal before it is fed to the S/H circuit. The rate
control actually belongs to the internal clock; when that is disconnected from the S/H
circuit, the rate control has no effect on the operation of the S/H circuit.
The Internal Clock / Electronic Switch
The Internal Clock is a manually controlled low-frequency square-wave oscillator. It is the
default trigger source for the S/H device. It is also hardwired as the clock source for the
Electronic Switch.
Under MIDI control, the Internal Clock may be synchronized to incoming MIDI Beat
Clocks; see section
The Electronic Switch has two connections on one side and one on the other, as indicated
by the panel graphics. For clarity, let’s call these three jacks A-1, A-2, and B. The switch
alternates between connecting A-1 to B, and A-2 to B. It doesn’t matter which side is the
signal source and which is the destination; the switch works the same regardless.
The switching rate is governed by the Internal Clock. This is a permanent feature of the
CHAPTER 4 - Modular Components of the TimewARP 2600
The Virtual Keyboard
This has five octaves, 60 keys. The control panel, at the left, is modeled on
but not identical to the original ARP 3620 keyboard.
Low Frequency Oscillator (LFO) Section
The keyboard unit has its own LFO section, independent of any of the standard VCO’s. It
can be used in two ways: for vibrato, or for automatically repeated keyboard gates (as, for
example, in imitating the repeated notes of a mandolin). Three sliders govern the Speed,
Delay, and Depth of the LFO.
Under MIDI control, the keyboard LFO may be synchronized to incoming MIDI
Beat Clock signals; see section
Dual-Pitch Control Output
Like the original ARP 2600, the TimewARP 2600 virtual keyboard can generate a second
pitch-control signal when two keys are depressed. This signal is available at the two jacks
labeled Upper Voice at the lower left of the keyboard module.
To use one of these, simply patch it to an oscillator. That oscillator will now track the
uppermost key depressed rather than – as with the standard keyboard control signal – the
lower key.
Gate and Trigger Control
Two switches in the upper right quadrant of the keyboard module govern the logic of the
keyboard gate and trigger signals.
When the Trigger Mode switch is set to Off, the keyboard generates a continuous gate
signal as long as any key is depressed, and generates a trigger signal only on the transition
from no key depressed to any key depressed. In this operating mode, you have complete
performing control over the production of trigger signals; to avoid them, play legato,
and to generate them, play non-legato. This is the baseline logic of the original ARP 2600
With this switch set to On, the keyboard will generate a trigger on every new keypress,
regardless of your performing habits. The gate logic is not affected.
The three-position switch labeled Auto Repeat is Off in its center position. This is the
In its lower position, the keyboard gate and trigger are taken from the local LFO. Actual
key depressions no longer play a role in gating. In its upper position, the LFO and the
keypress are ANDed together; when you press a key, there is a series of pulses from
the LFO, and when you release the key, the series stops. This is the mandolin effect we
mentioned above.
The keyboard Gate and Trigger signals are available on the main panel, from two jacks in
the Envelope Generator section.
Patching the TimewARP 2600
Please see the Patchman.pdf file found on the install disc or at www.wayoutware.com for suggestions.
If you are new to audio synthesis, consult the tutorial patch collection.
Table of Alternate keyboard tunings
Tuning Presets, compiled by Robert Rich
12 Tone Equal Temperament (non-erasable)
The default Western tuning, based on the twelfth root of two. Good fourths and fifths,
horrible thirds and sixths.
Harmonic Series
MIDI notes 36-95 reflect harmonics 2 through 60 based on the fundamental of A = 27.5
Hz. The low C on a standard 5 octave keyboard acts as the root note (55Hz), and the
harmonics play upwards from there. The remaining keys above and below the 5 octave
range are filled with the same intervals as Carlos’ Harmonic 12 Tone that follows.
Carlos Harmonic Twelve Tone
Wendy Carlos’ twelve note scale based on octave-repeating harmonics.
A = 1/1 (440 Hz).
1/1 17/16 9/8 19/16 5/4 21/16 11/8 3/2 13/8 27/16 7/4 15/8
Meantone Temperament
An early tempered tuning, with better thirds than 12ET. Sounds best in the key of C. Use
this to add an authentic touch to performances of early Baroque music. C = 1/1 (260 Hz)
1/4 Tone Equal Temperament
24 notes per octave, equally spaced 24root2 intervals. Mexican composer Julian Carillo
used this for custom-built pianos in the early 20th century.
19 Tone Equal Temperament
19 notes per octave (19root2) offering better thirds than 12 ET, a better overall
compromise if you can figure out the keyboard patterns.
CHAPTER 6 - Appendices
31 Tone Equal Temperament
Many people consider 31root2 to offer the best compromise towards just intonation in an
equal temperament, but it can get very tricky to keep track of the intervals.
Pythagorean C
One of the earliest tuning systems known from history, the Pythagorean scale is
constructed from an upward series of pure fifths (3/2) transposed down into a single
octave. The tuning works well for monophonic melodies against fifth drones, but has a
very narrow palate of good chords to choose from. .
C = 1/1 (261.625 Hz)
1/1 256/243 9/8 32/27 81/64 4/3 729/512 3/2 128/81 27/16
16/9 243/128
Just Intonation in A with 7-limit Tritone at D#
A rather vanilla 5-limit small interval JI, except for a single 7/5 tritone at D#, which offers
some nice possibilities for rotating around bluesy sevenths.
A = 1/1 (440 Hz)
1/1 16/15 9/8 6/5 5/4 7/5 3/2 8/5 5/3 9/5 15/8
3-5 Lattice in A
A pure 3 and 5-limit tuning which resolves to very symmetrical derived relationships
between notes. A = 1/1 (440 Hz)
1/1 16/15 10/9 6/5 5/4 4/3 64/45 3/2 8/5 5/3 16/9 15/8
3-7 Lattice in A
A pure 3 and 7-limit tuning which resolves to very symmetrical derived relationships
between notes. Some of the intervals are very close together, offering several choices for
the same nominal chords.
A= 1/1 (440 Hz)
1/1 9/8 8/7 7/6 9/7 21/16 4/3 3/2 32/21 12/7 7/4 63/32
Other Music 7-Limit Black Keys in C
Created by the group Other Music for their homemade gamelan, this offers a wide range
of interesting chords and modes.
C= 1/1 (261.625 Hz)
1/1 15/14 9/8 7/6 5/4 4/3 7/5 3/2 14/9 5/3 7/4 15/8
CHAPTER 6 - Appendices
Dan Schmidt Pelog/Slendro
Created for the Berkeley Gamelan group, this tuning fits an Indonesian-style heptatonic
Pelog on the white keys and pentatonic Slendro on the black keys, with B and Bb acting
as 1/1 for their respective modes. Note that some of the notes will have the same
frequency. By tuning the 1/1 to 60 Hz, Dan found a creative way to incorporate the
inevitable line hum into his scale.
Bb, B = 1/1 (60 Hz)
1/1 1/1 9/8 7/6 5/4 4/3 11/8 3/2 3/2 7/4 7/4 15/8
Yamaha Just Major C
When Yamaha decided to put preset microtunings into their FM synth product line, they
selected this and the following tuning as representative just intonations. As such, they
became the de-facto introduction to JI for many people. Just Major gives preferential
treatment to major thirds on the sharps, and a good fourth relative to the second.
C= 1/1 (261.625)
1/1 16/15 9/8 6/5 5/4 4/3 45/32 3/2 8/5 5/3 16/9 15/8
Yamaha Just Minor C
Similar to Yamaha=92s preset Just Major, the Just Minor gives preferential treatment to
minor thirds on the sharps, and has a good fifth relative to the second.
C= 1/1 (261.625)
1/1 25/24 10/9 6/5 5/4 4/3 45/32 3/2 8/5 5/3 16/9 15/8
Harry Partch 11-limit 43 Note Just Intonation
One of the pioneers of modern microtonal composition, Partch built a unique orchestra
with this tuning during the first half of the 20th century, to perform his own compositions.
The large number of intervals in this very dense scale offers a full vocabulary of
expressive chords and complex key changes. The narrow spacing also allows fixed-pitched
instruments like marimbas and organs to perform glissando-like passages.
G = 1/1 (392 Hz, MIDI note 67)
81/80 33/32 21/20 16/15 12/11 11/10 10/9 9/8 8/7 7/6
32/27 6/5
11/9 5/4
14/11 9/7
21/16 4/3
10/7 16/11 40/27 3/2 32/21 14/9 11/7
27/20 11/8 7/5
27/16 12/7
7/4 16/9 9/5 20/11 11/6 15/8 40/21 64/33 160/81
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