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bjorn@sveinsvoll.net 1:33am on Wednesday, September 29th, 2010 
Toshiba offer a free motherboard remplacement and 1 additionnal year of warranty and a new bios version(3.9) go to their site to check! At 15 mothns old, the green screen problem occured. The video card/motherboard rendered it useless. Very poor repsonse from toshiba.
trikywayn 9:21pm on Tuesday, May 25th, 2010 
MARKED AS SPAM BY AKISMET Lots of features ; A year of great performance Battery life is terrible ; Video card/motherboard issues (issues??.. Toshiba offer a free motherboard remplacement and 1 additionnal year of warranty and a new bios version(3.9) go to their site to check!

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CHAPTER 3 - The Craft of audio Synthesis
Analog signals @ 250 Hz (approximately middle C)

Source

forth back higher lower forth back higher lower up down north south positive negative forth back

Air Pressure Wave

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 recording. Since the TimewARP 2600 is not constructed of electronic circuits, concepts such as voltage dont 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 Owners 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.
Microphone Diaphram Microphone Output Signal (Voltage)

Phonograph LP groove

Magnetic Field on Tape
Power amp Output (Volt * current = Power)

Loudspeaker Cone

Your Eardrum

in out

250 = 4 msec
Attributes of Signals The simplest possible signal is a sine wave. Its 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 dont have to get involved in that unless youre curious about it, but its 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. Its 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 yourself:

etc. amplitude

period

1 frequency = period

phase (important only in comparing two signals)
CHAPTER 3 - The Craft of audio Synthesis 3.2.1.1 3.2.1.2 3.2.1.3 3.2.2
Amplitude: whats 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 isnt 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 relationships. 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.

3.2.2.1
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:

3.2.2.2

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 signal. 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 3.2.2.2.1 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 spectrum.

Spectrum = the sine-wave component of a signal
@250Hz sine wave saw tooth guitar string
immediately after pitch just before dying out
A harmonic series is composed of numerically equal frequency intervals, and that means decreasing pitch intervals

101112

1/8 pulse wave

Increasing Power

6 (octaves)

Increasing Pitch

Increasing Frequency

middle C 250Hz

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.

250Hz octave 1K

1750 2K

5th maj min 3rd 3rd

100Khz 400

8 ANY ANY*2 *3 *4 *8 *16

2Khz 4K 6K 8K
CHAPTER 3 - The Craft of audio Synthesis 3.2.2.2.2 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 components.
Not all spectra are harmonic. An Arbitrary collection of sines at unrelated frequencies is "unharmonic".

1.47 2.43

2.725 3.157
In audio synthesis, you can use various modulation techniques such as AM and FM - to generate enharmonic spectra. 3.2.3 Attributes of Random Signals
+1.7 ANY Increasing Pitch
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. 3.2.3.1 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, dont sound properly balanced across our range of hearing. They sound too bright.
CHAPTER 3 - The Craft of audio Synthesis 3.3 Attributes of Auditory Events 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 its your native tongue (when its a language you dont 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 sounds:

Frequency in cycles / seconds
10KHz Each division is one octave. To add an octave, you have to multiply F x 2. 5KHz
This is a really important fact in audio synthesis; you will encounter it over and over again, in every 20Hz patch you create. It governs much 0 of the arithmetic of generating and controlling spectral distributions and patches:

625Hz 1250Hz

2500Hz

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 3.4.2 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. 3.4.2.1 How Adding Volumes Means Multiplying Amplitudes

Reference Power level

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.
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.

. and in the frequency domain, this cuts off higher-frequency components.
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. 3.6.4.2

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. 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.
A highpass filter makes it difficult for the medium to keep a new level - it "leaks" away.
. and so lowerfrequency components are weakened.
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 3.6.4.3 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. 3.6.4.4 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. 3.6.4.5 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. 3.6.5 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 wont 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 3.6.5.1 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, and the signal that signal provides the modulation pattern is called the program. (Guess why.)
Given this for a carrier: carrier spectrum
. and this for a program program spectrum
here's the AM result: lower sideband
carrier weakend upper sideband
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 components. 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.

here's the ring modulation: the carrier is gone ?
and here's the FM result:
variable number of sidebands
and varying amplitude depending on the modulation depth

3.6.5.2

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 3.6.5.2.1 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 doesnt 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. Thats what a ring modulator does. (The expression ring modulator describes the appearance of the analogue circuit design thats 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. Whats 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 youre careful about the ratio of the two input signal frequencies. 3.6.5.2.2 Frequency Shifting Its 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 its not pitch shifting which would have to be accomplished by frequency multiplication but frequency shifting. Its an addition or subtraction process, and it really messes up any harmonic relationships that might have existed in the original spectrum. 3.6.5.3 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.

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 operations. 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 above).
CHAPTER 4 - Modular Components of the TimewARP 2600 4.1.4 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. 4.1.5 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 4.1.5.1 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 overlaps. 4.1.5.2 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. 4.1.5.3 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.) 4.1.5.4 Rename / Delete The Rename / Delete buttons appear within the Patch Manager dialog to Rename or Delete selected items. 4.1.6 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 synthesizers 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 machines clock speed.

CHAPTER 4 - Modular Components of the TimewARP 2600 4.1.7 Reset Button The Reset button removes all patch cords and returns all sliders to a standard position. 4.1.8 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. 4.1.9 Output-Level Meter This shows the output signal level. If it reaches into the red segment, your signal will distort. 4.1.10 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. 4.1.11 The Magic Logo At the lower right of the main panel is the TimewARP 2600 Logo. Clicking on this brings up a menu:

4.1.11.1 4.1.11.2

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 4.1.11.3 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 changes
4.1.11.4 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 4.3.1 VCO 1 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 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 4.3.3 VCO 3 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.

Mixer 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, theres 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 4.8 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 40Hz. 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 VCOs. The Envelope Follower output is an envelope and can be used in the same fashion as the output from either of the envelope generators. 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.

Appendices

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 6.1.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. 6.1.8 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 6.1.9 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 6.1.10 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 6.1.11 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 6.1.12 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 6.1.13 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 6.1.14 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 6.1.15 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 6.1.16 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) 1/1 81/80 33/32 21/20 16/15 12/11 11/10 10/9 9/8 8/7 7/6 11/9 5/4 14/11 9/7 21/16 4/3 8/5 27/20 11/8 7/5 18/11 5/3 27/16 12/7 32/27 6/5
10/7 16/11 40/27 3/2 32/21 14/9 11/7
7/4 16/9 9/5 20/11 11/6 15/8 40/21 64/33 160/81

doc1

Foreword

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? Its time travel. Im grateful to Way Out Ware for providing me that opportunity. When, at Alan R. Pearlmans invitation, I began work on the original 2600 manual in September of 1970, the 2600 itself barely existed. For the rst two months, I was writing blindwithout a machine in front of me. My rst hands-on experience with a synthesizer had been only six months earlier (it was a Putney VCS3). I nished the text in March of 1971, Margaret Friend created the graphics, and the Owners Manual for the ARP 2600 began what turned out to be a surprisingly long career. In spite of the many defects that my inexperience contributedthe gaps in coverage, and outright errorsit became quite popular. To this day, it still gets an occasional respectable mention in the analog-synthesis community. When Way Out Wares Jim Heintz called in early 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, thought about, and solved these problems. He owned a real 2600. He really aimed at getting it right and would not be satised with anything less. It was a pleasure, nally, to accept his invitation to do an Owners Manual. Its now clear that Way Out Ware has set a new standard for software-based audio synthesis. The behavior of the TimewARP 2600 softwareboth module-by-module and integrated into patchesis effectively indistinguishable from that of the analog hardware that it emulates. Soaring and swooping through the free air of analog synthesisa world of nothing but sliders and cords and continuously evolving patch congurationwas a capstone course at the Boston School of Electronic Music in the 1970s. 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 rstand I believe onlysoftware synthesizer to support real-time performance by sliders and patchcords alone. So here it is: your new Owners 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 ight?

Jim Michmerhuizen Jim Michmerhuizen is the author of the original ARP 2600 Manual and Founder and Director of the Boston School of Electronic Music.
The ARP 2600, 1970 - 1981 and Onward
How this Manual is Organized
This is not a textbook; its a survival manual. Chapter 2 is about installing and conguring 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 softwarea piece of computer behaviorrather 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 le 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

Youll 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 youre new to audio synthesis, youll want to walk through the tutorial patch sequence in Patchman.pdf, referring back to Chapter 3 for concepts and Chapter 4 for detailed module specications 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 specications 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.

2.1.3 2.1.3.1

Setups 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 rst 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.

2.1.3.2

Processing Audio Signals The TimewARP 2600 can be used to process audio signals in a similar manner to a reverb or other effects plugin. 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 rst available insert selector. Choose the TimewARP 2600 plug-in from the appropriate menu. If the current track is mono, the TimewARP 2600 for this track will be congured 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 congured for stereo throughout. Conrm 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 tracks 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.

2.1.3.3

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 2.1.3.1: 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.

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 dont have a hardware MIDI keyboard handy.
The Craft of Audio Synthesis
This chapter is about the factsphysical, mathematical, and auditorythat 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. This is important because synthesizer equipment can only deal with signalsphysical commotion of one sort and another. When you are ddling 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 ashlight, or ambulance siren, referee ag 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 elds 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 amplier, 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 eld strength or stylus position, the signals are identical. Graphed or charted, they even look the same. This is analog recording.
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 recording. Since the TimewARP 2600 is not constructed of electronic circuits, concepts such as voltage dont 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 Owners Manual used the word voltage we will just say signal or signal level, and in the module specications we will refer to virtual Volts or vV.

How Signals and Sounds Go Together. Sort of
Signal activities, being entirely physical kinds of things, are easily nameable, measurable, catalog-able, and 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 well-established 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.

3.4.1.1

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: Equal pitch intervals require exponentially increasing frequency increments; Equal frequency increments make a harmonic series. In a harmonic series, as we go up the series we get smaller and smaller pitch intervals.
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-colorthan 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 examinevisuallythe recorded signal from a pipe organ, or even an orchestra, you may be surprised to nd 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 distributionthe most harmonic contentwill sound louder.
How Adding Pitches Means Multiplying Frequencies The range of audio frequenciesof human hearing, in other wordsis 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.

3.4.2.1

How Adding Volumes Means Multiplying Amplitudes 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 rst one, you have to double the amplitude again. 3.4.3 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 soundin its tone-color or spectrummust 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. Its fairly easy to nd waveform changes that listeners cant hear. For example, we humans simply arent 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. 3.4.4 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
Throughout the past century people have been noodling around with electronic ways of generating audio signals. In particular, beginning in the 1960s, 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 congurations of modules. The cables that connected modules were called patchcords, and so connecting modules together came to referred to as patching them, and so, nally, any working conguration came to be called a patch. Lets take a look at some modules that generate signals.
Oscillators 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.

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 amplied, 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.
How can you get access to the signals and processes weve just described, so that you can play with them on your own?
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.
Filtering and equalization can shape a noise spectrum into almost any sound. 3.5.3 Envelope Generators 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.
envelope generators are controlled by gate and trigger signals.
trigger ADSR envelope AR envelope
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
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.
Attenuators/Ampliers 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 amplier may increase the amplitude of a signal. But not necessarily; it is usual for a voltage-controlled amplier to have a maximum gain factor of 1.0. In fact, the purpose of VCAs is actually to chop the signals passing through them. It would make more sense to think of them as voltage controlled attenuators.

Inverters 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).

3.6.3.1

Gain Factor To describe the behavior of an amplier or attenuator, we may use the expression gain factor to mean the ratio of output signal amplitude to input amplitude. Filters A lter 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 lters.) Because of this frequency-dependent characteristic of lters, they change the shape of any complex waveform passing through. And so it will be important for you to get to know what lters do to signals in both the time domain and in the frequency domain.

3.6.4.1

Low-Pass Filters Any device or mechanism that passes along slower motions better than faster ones can act as a lowpass lter. 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. Thats a lowpass lter. 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 ltering. For subaudio signals, it is usually the time-domain effectschanges in waveshapethat we care about. In the time domain, a low-pass lter rounds off any sharp transitions in the signal. A good example of this is the lag processor described in section 3.6.4.5 below. In the frequency domain, it weakens spectral components that are higher than the lter cutoff frequencythe frequency at which the lter begins to have an effect on the signal.

3.6.4.2

High-Pass Filters Any device or mechanism that passes along faster motions better than slower ones is a high-pass lter. 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. 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 downit 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.

3.6.5.1

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 kind of thing 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, and the signal that provides the modulation pattern is called the program. (Guess why?) 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 frequencyand at least one upper sideband, at a frequency equal to the component plus the modulating frequency. If the carrierthe original signalis 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 signalthe programis complex, the arithmetic applies separately to each of its components.

3.6.5.2

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.

3.6.5.2.1

Ring Modulation While 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 graphsit doesnt matter what they areon the same timebase and vertical scale, and use a pocket calculator to work out the result of multiplying the two signals together. Thats what a ring modulator does. (The expression ring modulator describes the appearance of the analogue circuit design thats 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 singlecomponent program, generates a three-component AM spectrum but only a two-component ring-modulation spectrum. Whats 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 inharmonic if youre careful about the ratio of the two input signal frequencies.

4.1.14

To see a simple example of this, note that the rst 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 pre-wired 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 fth of the ve audio inputs to the VCF is similarly defaulted from the Noise Generator (counting across from left to right this is the 21st jack):
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 ltered 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 complications: Experiment with one signal at a time. With the VCF, for example, when you have listened to everything the lter 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 level.
If the TimewARP 2600 is running in full stereo congurationas a plug-in to a stereo trackthe 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 lter 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.
Normalled Jacks The most commonly used signal connections are normalled (i.e. defaulted). A default signal is identied by a small icon at an input jack.
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 ne-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.

Voltage Controlled Filter (VCF)
The Voltage Controlled Filter has variable cutoff frequency (Fc) and resonance (Q). The response below Fc is at 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 1Hz and as high as 20kHz. Fc is controlled manually by a coarse tuning slider (labeled initial lter frequency) and a ne tune slider. Fc may also be controlled by external voltages; the sensitivity under voltage control is 1.0vV/oct.
At this Q setting, just below the point at which oscillation begins, the lter will ring distinctly in response to any sharply dened 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 lter will oscillate. Operating in this state, it generates a pure sine wave, even in the absence of any signal input. The VCF has ve 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 rst is normally from the Keyboard pitch-control. The slider that governs this input is a TimewARP 2600 extension; on the original 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 repeat-ably 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, specically the data on control input sensitivity. Also, reread 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 two-position switch just under the lower AR generator selects between the two latter sources. The Manual Start button overrides both of these.

Mix/Pan/Reverb Output Module
The three functions in this module provide nal 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 congured as a two-input Mixer, which feeds a Pan Control and a Reverb unit, which are themselves mixed to feed the nal left/right system output channels. When the TimewARP 2600 is congured for mono operation, this section omits the Pan control and provides only one output channel.
Mixer 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 oating attenuators, in any situation where you need to set the strength of a signal. (Although if you do this, theres 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 xed 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.

Envelope Follower

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 nal value and 30 milliseconds to 90% of nal 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 40Hz. 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 VCOs. The Envelope Follower output is an envelope and can be used in the same fashion as the output from either of the envelope generators. The default input to the Envelope Follower is from the preamplier. When the TimewARP 2600 is congured for stereo input, it is the rst (left) channel preamplier output.

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 4.1.11.4. The Electronic Switch has two connections on one side and one on the other, as indicated by the panel graphics. For clarity, lets call these three jacks A-1, A-2, and B. The switch alternates between connecting A-1 to B, and A2 to B. It doesnt 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 switch.

The Virtual Keyboard

4.14.1
Low Frequency Oscillator (LFO) Section The keyboard unit has its own LFO section, independent of any of the standard VCOs. 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 4.1.11.4.

4.14.2

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 thanas with the standard keyboard control signalthe lower key.

4.14.3

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 Single, 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 keyboard. With this switch set to Multiple, 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 default. 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.

 

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