Friday, March 9, 2018

Review: SSL 1900 O'Tool

Always wanted an oscilloscope integrated into your modular?  Euro modular users have the Doug Jones Design O'Tool, a handy scope with a little color LCD display mounted in a module.  Fortunately for us 5U guys, Doug Slocum took it upon himself to reformat some into MU format, and the result is the Synthetic Sounds Labs Model 1900 O'Tool.  Since the days of Keith Emerson, modular users have wanted to find a way to mount an oscilloscope and be able to conveniently route signals to it.  Problems with this have always included the size and weight of traditional scopes, their incompatibility with the panel formats and mounting methods that modulars use, and signal compatibility issues.  (Who has room for a Tektronix 464 in their case?  Or the $$$ for a Tek MDO3000?  Me neither.)

The O'Tool solves these problem neatly, and provides many more capabilities than your average Ebay-special analog Tektronix.  The O'Tool consists of a digital signal processing system coupled to a color LCD screen, packaged in a modular synth panel format.  It is powered from conventional +/- 15V power, and easily accepts the usual modular synth signal levels and types.  No heavy CRT, no high voltages, and no four-figure price tag.  Available functions consist of scope screens, voltage measurement, frequency measurement, signal level metering, and spectrum analysis.

This version of the O'Tool is physically packaged as a 1U wide Dotcom/MU format module.  When I received it, I was a bit concerned at first because the screen is pretty small, and my eyesight is not what it used to be.  However, the contrast and resolution are excellent, and I've had no trouble reading the screen.  The screen does take up as much of the width as could fit without structurally compromising the panel (which would make it difficult to package this in an MOTM-format module).  There are six input jacks, a pair for each of the input channels, and a pair for an external trigger signal for the scope modes.  Each pair is simply wired together; this allows them to be used to "patch through" a signal that needs to go somewhere else, so you can conveniently insert the O'Tool into a patch without needing a mult.

Underneath the screen is a row of four small pushbuttons.  The leftmost one selects the operating mode and screen to be displayed.  Pressing the mode button repeatedly cycles through the screens.  The other three buttons are "soft keys" whose functions vary depending on the selected screen.  Each screen has a small legend at the bottom showing what the soft buttons do in that screen.  The available screens are:
  1. Single channel voltage/time scope, displays channel 1 only
  2. Dual-channel voltage/time scope, with the two channel signals overlaid.  Channel 1 is displayed in red, and channel 2 in green.
  3. Dual-channel voltage/time scope, split screen.  Channel 1 is displayed in the top half, and channel 2 in the bottom half.
  4. Bar-graph averaging voltage display, described further below.
  5. VU/peak level meter
  6. Spectrum analyzer
  7. X-Y oscilloscope
  8. Frequency counter
  9. Digital voltmeter

Scope Mode Screens

In the first three screens, the three buttons allow the user to select the time per horizontal division, the displayed voltage range, and the trigger source and mode.  Pressing each button cycles through the available values (which can be a bit tedious in the case of the time/div setting since there are many possible values).  The screen is underlaid with a division grid, shown in dark blue, which appears behind the signal traces.  The available time per division settings range from 100 microseconds per division to 5 seconds per division.  There are six horizontal divisions across the screen, so at the maximum setting, the time for one screen sweep is 30 seconds.  The voltage range differs from oscilloscope convention, and from the time range, in that it applies to the entire vertical span instead of per grid division.  Available ranges are

  • Plus/minus 5V DC
  • Plus/minus 10V DC
  • Plus/minus 10V AC (DC signals/offsets are filtered out)
  • 0-5V DC
  • 0-10V AC

Here are some screen shots of the three scope mode screens.  (All of the photos from here through the end of this post were photographed from the actual screen.  I cropped the shots, and added some contrast enhancement in order to get rid of room light reflecting off of the screen; otherwise, the photos are unretouched.  The somewhat fuzzy look is caused by magnification of the photos, and the fact that I had to use the camera's digital zoom because I don't have a proper macro lens.  Bear in mind that these photos are larger than the actual screen.  All of the waveforms are from a Q106 VCO.)

This is the single-channel mode, showing a sine wave:

The dual-channel stacked mode, showing two waveforms from the same VCO.  Channel 1 is shown in red and channel 2 is in green.  Here, channel 1 (the sine wave) is chosen as the trigger signal.

The dual-channel layered mode, with the same two waveforms.

The same screen, but with channel 2 (the sawtooth wave) chosen as the the trigger channel  

The trigger can be set to trigger on either of the two input channels, or on the signal connected to the external trigger input jacks.  It can also be set to no-trigger mode, in which the scope free runs.

Triggers and Triggering Modes

The concept of triggering, for a scope in general, can be a bit difficult to understand at first.  The reason that scopes have triggered modes is to make the waveform "stand still" on the display.  Considering what would happen if the scope was free running; that is, if it scanned continuously.  Unless the waveform you are trying to display happens to be divisible by the scan rate, the wave won't be stationary on the screen; it will begin in a different place in its cycle on each scan, resulting in a display that jumps around.

To solve this problem, a scope has some sort of detection of a certain part or feature and then generates a trigger signal, not unlike the trigger signals that we use in our synths.  The trigger causes one horizontal scan to happen; after that scan is completed, the scope waits until it sees the trigger again, and then it does another scan and updates the display, etc.  By doing this, the scan always starts at a chosen point in the signal cycle, so that the displayed waveform remains stationary and you can actually look at it. 

The O'Tool can use either input channel as the source to the trigger detector, or it can use the signal at the "Trigger" input  There are five trigger modes:

  • Trigger 1.  Uses channel 1 as the trigger source.  If the ±5V or one of the ±10V ranges are selected, the trigger is generated when the signal crosses the horizontal axis in the positive-going direction.  If the 0-5V or 0-10V range is selected, the trigger is generated when the signal crosses 1.25V in the positive-going direction.
  • Trigger 2.  Same as trigger 1 except that it uses channel 2 as the trigger source.
  • Ext 0V uses the external trigger input as the trigger source.  The trigger is generated when the signal crosses the horizontal axis in the positive-going direction.
  • Ext 1V is the same except that the tigger is generated   when the signal crosses 1.25V in the positive-going direction.
  • No Trigger is a free-running mode; the scan runs all of the time, unsynchronized to the input signals.  

The trigger selection allows you to select either channel to be fed to the trigger generator.  You can even do this in the single-channel mode; you can display channel 1 and trigger off of channel 2.  The display may look different depending on which channel you trigger from.  Consider the screen shots of the two stacked-mode screens above.  In the top one, the trigger is on channel 1, so it triggers when the sine wave crosses the X axis going up.  The nature of the Q106 VCO (as with most sawtooth-core VCOs) is that the positive peak of the sine wave is where the positive peak of the sawtooth wave is.  So the top half starts with the sine wave heading up from zero towards its peak, while the sawtooth display starts with the last 90 degrees of its cycle.  In the second photo, we switch the trigger to channel 2. Now we are triggering on the positive-going zero crossing of the sawtooth, which is pretty much instantaneous.  So we see the display start with both of the waveforms descending from their peaks.  

Level Displays

The bar graph display is interesting but kind of hard to describe.  Basically, what it does is show how much time -- what percentage of the cycle -- a signal spends at a given voltage level.  The more the signal is at that a given level, the brighter the bar will be at that level.  In the shot below, channel 1 is a square wave and channel 2 is a sawtooth.  The square wave, of course, alternates sharply between the positive and negative peaks; hence the two discrete bars.  The sawtooth falls linearly and so all of the voltage steps get the same saturation, resulting in a spread of evenly lit bars.  (Not sure why the top one is a bit dimmer; may have to investigate how the sawtooth waveform is looking coming out of that VCO.)  The display range can be adjusted, and "fast" or "slow" averaging can be selected.

The VU and peak level meters do what you expect them to: display the average and peak voltage level of an alternating signal.  The display shows VU and peak levels for both of the input channels; the VU displays are grouped on the left, and the peak displays on the right. The VU indicators appear to be a true RMS measurement, as they display identically to the peak levels when a sine wave is input.  I cannot say, however, that the ballistics of a true VU meter are emulated properly; I don't have any means to measure it.  There are three selectable scale modes, which effect what reference level is used for the meters, and how the scale on the peak meters is displayed.

Like many such meters, the display uses color bars to display different regions of the measurement levels.  The blue horizontal line indicates the reference signal level (the level that is considered a "100%" signal) for whatever scale mode is in use.  Levels below and at the line are displayed using green bars.  Above the line, on the peak side, the first three steps are displayed using yellow bars, and levels above that are displayed with red bars.  On the VU side, all levels above the line are displayed using red bars.

This screen has three modes, which effect where the "100%" line is, and how the peak display is scaled.  The modes are:
  • +4dBu: In this mode, the VU scale conforms to the standard recording industry definition, in which zero VU = +4 decibel volts RMS, or dBu.  This in turn is defined as 1.228 volts RMS.  (It's defined at 1000 Hz, but that is not supposed to matter across most of the audio range.  I'll have more to say about this further down.)  The peak scale displays dBu and the blue line will pass through +4 on that scale.  (It always passes through zero on the VU scale.)  Red bars on the peak scale start at +8 dVu.
  • +2.5V: In this mode, zero on the VU scale corresponds to 2.5V RMS.  The peak scale will be re-scaled to show voltages up to 10 volts, and the blue line will pass through the 2.5V mark.  Red bars on the peak scale start between the 3.5V and 5V marks.
  • +5V: In this mode, zero on the VU scale corresponds to 5V RMS.  The peak scale will be re-scaled to show voltages up to 10 volts, and the blue line will pass through the 5V mark.  Red bars on the peak scale start between the 7V mark and the 10V mark.
An issue with the VU/peak display is that it does some preliminary high-pass filtering before it processes the signals for display.  This is common for VU meters; it prevents a DC offset in the signal from creating a false high reading.  However, it prevents the meters from working properly with low-frequency signals.  If you want to look at levels from an LFO, use the bar graph display, or one of the scope screens.

X-Y Display

The X-Y display emulates a feature of many of the old analog scopes, in which the X-axis, which is normally controlled by the scope's time base, can instead be driven by an external signal, producing two-dimensional patterns on the scope screen.  In this implementation, channel 1 drives the X axis and channel 2 drives the Y axis.

The old analog scopes depended on the persistence of the display phosphor for the user to be able to perceive the drawn figures.  The O'Tool attempts to emulate that with a setting that defines the "persistence" of each dot drawn; the dot is removed from the display after the equivalent of what would be that amount of time has passed, which determines how long each part of the figure remains on the display (which is, of course, also a function of the frequencies of the two waveforms driving the display).  To my eye, it doesn't work all that well; the continuously redrawn form is hard to perceive at faster settings, and it quickly fills the entire screen at slower settings.  The photo below was taken at a 1/15 second exposure and captures more of the drawn figure (which is made from a triangle wave driving the X axis and a sine wave on the Y axis) than was visible to the eye in real time.

Spectrum Analyzer

The spectrum analyzer surprised me with how well it works.  The update rate is pretty fast, and it seems to not have much of a problem with quantizing noise.  It has two display modes, "linear" and "log".  In the linear mode, there are four frequency ranges available, with a choice for the upper end of 20, 10, 5, or 2.5 KHz.  Vertical scaling is relative, but you can choose from 1x up to 4x.  If, in one of the higher vertical magnifications, one of the peaks exceeds the vertical range, the peak displays a red top, as you can see in the shots below.

This one is with a square wave on channel 1 (top) and a sawtooth on channel 2 (bottom.)  On the square wave, you can see the odd-harmonics pattern typical of square waves. 

This one is with a 25% pulse wave on channel 1, and a triangle on channel 2.  Notice how little harmonic content the triangle has. 

I have found the log mode to not be as useful in general, because it groups all of the frequencies into octaves and displays one bar per octave.  Depending on the range setting, it displays between 5 and 7 octaves.  Here's an example; unfortunately, I forgot to write down what waveforms I was using for this shot.

Frequency Counter

The frequency counter is straightforward and works well.  You can select channel 1 only, channel 2 only, or both.  There are three elements on the display.  At the top is the frequency, in Hz, for each channel.  Not having a calibrated frequency source, I can't really speak to whether these are actually precise to two decimal places.  In the middle, it displays the closest equal-tempered note for the frequency of each channel, and how far away in cents it is from the ideal equal-tempered value.  The bottom portion shows this deviation graphically.

For the note display, you can select concert A to be either be 440 Hz (the usual standard) or 432 Hz.  There is a lot of nonsense surrounding 432 Hz tuning; there's nothing special or magical about it.  Prior to the 20th century, orchestras were all over the map as to what standard they tuned to; Bach, Beethoven and Haydn are thought to have used an A of about 422 Hz.  Nonetheless, if you want to try something different, and you're using the O'Tool to tune instruments, you can give 432 Hz a try.  Note that some polysynths may not be capable of being tuned this far off of 440 Hz.


\The last display is a simple voltmeter, displaying the voltage present on each channel.  Only DC voltages can be displayed.  Keep in mind that the O'Tool is (in this case) being powered from a 15V supply; it most likely cannot display voltages exceeding the supply rails, and trying to do so could potentially damage it.  (I haven't tried.)  So don't use it to check the supply voltages on your Roland System 700.


The O'Tool is a useful thing to have in your setup.   And it looks cool.

Monday, February 19, 2018

Review: Q119 Analog Sequencer

The Q119 from Synthesizers,com is a 24-step analog sequencer. If you haven't used an analog sequencer before and don't know what its purpose is, it's a device that stores a set of control voltage values, and sends them to an output one after the other, under the control of a clock signal. As is the case with many analog sequencers, the “storage” for the control voltages consists of a set of knobs, each of which selects a control voltage within a given range. If you've listened to early Tangerine Dream or any other “Berlin school” electronic music, you've doubtless heard note sequences produced by an analog sequencer connected to the control input of a VCO. Repeating control voltage patterns have a huge variety of other uses, such as controlling filters, switching between different signals via connections to VCAs, and even using the output as an audio signal when the clock rate is high enough.

Like all products, the Q119 is formatted in the MU (Dotcom) format, which means it uses 1/4” jacks for all signal connections, and the standard Dotcom six-pin MTA-100 connector for power. (It does draw from the +5V power; the power supply must supply that voltage in order for the Q119 to function.) At a width of 8U, it is one of the physically largest modules that offers. The panel is divided into three basic sections: The section on the left has the clock controls and the various option switches that change the way the sequencer works. The middle and largest section consists of the 24 step controls, each having a control voltage tuning knob and an LED indicator. The section on the right is the output section, with the row outputs, and the master outputs with their offset and lag controls. Q119 analog sequencer, with a single-width Q128 A-B switch shown next to it for size comparison.

Clock Rate, Start/Stop, and Cycle Controls

Cycle option switches at the top,
clock controls at center,
start/run/stop controls at bottom
In the clock section, the most prominent controls are the oscillator frequency (RATE) knob and the GATE WIDTH knob. The RATE knob and the adjacent RANGE switch control the rate of the internal clock. With the knob full CCW and the RANGE switch on LOW, the slowest available rate is about 3 Hz, which to me is not slow enough. If you want slower, you have to use an external clock, The fastest available rate, with the RANGE switch on HIGH, is about 320 Hz. To the left of this knob is the external clock input and the SOURCE switch. As you might guess, when the SOURCE switch is in EXTERNAL, the internal clock is disconnected and the sequencer is driven by a clock signal received at the external clock input. This input should be a pulse wave (although the sequencer will square it up if it isn't), and the sequencer advances on the leading edge.

When the internal clock is being used, the GATE WIDTH control determines the “on” time of the gate outputs, as a duty cycle percentage (which means that as the frequency gets faster, the gate on time gets shorter). Unfortunately, the one on my Q119 does not work (I bought this unit used); it produces gates that are about 1 ms wide regardless of what I set the knob at. Fortunately, when an external clock is used, the gate on time follows the pulse width of the external clock; the GATE WIDTH control is ignored. This means that if you are driving the Q119 with a VCO that has pulse width modulation, you can change the gate “on” time by adjusting the VCO's pulse width, or better yet, make the gate “on” time voltage controlled by feeding a control voltage to the VCO's pulse width input.

The start/stop controls at the bottom of the clock section consist of four pushbuttons and three associated input jacks (one for each button except SET END). The START button, when pressed, causes the sequencer to start; it then runs continuously (unless the the SINGLE / CONTINUOUS switch is in the SINGLE position), until the STOP button is pressed. The GO button causes the sequencer to run as long as the button is held; when the button is released, it stops. The jacks under the START and STOP buttons accept trigger signals; receiving a signal on one of these jacks has exactly the same effect as pushing the associated button. The jack under the GO button accepts a gate input; the sequencer will run as long as the gate signal is high.  The SET END button, we'll cover in a minute.

How fast will it run?

With an external clock, I tested mine to see how fast it would run, and it made it up to 920 Hz; faster than that, and the sequencer freezes. ('s documentation only says that it will run “up to” 1 Khz.) This means that you can, in effect, use the Q119 as a sort of function generator at low audio rates; at this speed, a full 24-step sequence will cycle at about 38 Hz, and faster if you make the sequence shorter. There is no limit on the slowest rate; you can unplug the cord from the external clock jack, and the sequencer will simply wait until you plug it back in. When the sequencer is stopped, pressing the MANUAL STEP button next to the RATE knob causes the sequencer to advance one step. This is normally used to tune steps when setting up a sequence, but it can be used to “clock” the sequencer manually.

Cycle options

The four switches across the top select various options for the sequencer's operation. The MODE switch, I'll cover in the next section where we go over the step controls. The voltage range OUTPUTS switch sets the minimum and maximum range of the step tuning knobs. When the switch is in the -5 / +5 mode, turning a step knob full CCW causes tha step to output -5V, and full CW outputs +5V; the 12 o'clock position outputs 0V. When the switch is in the 0 / +5 mode, full CCW on the step knob outputs 0V. (The 12 o'clock position doesn't output 2.5V; I'll say more about this later.)

When the CYCLE switch is in the SINGLE position, the sequencer always stops on the last step in the sequence. To make it run again, a START operation has to be performed again. In the CONTINUOUS position, as you might expect, the sequencer runs in a continuous loop until you stop it. (Note that when the “hidden” random mode is selected, this switch is ignored; the sequencer always runs continuously until stopped.). The SEQUENCE switch, when in the UP/DOWN position, causes the sequencer to reverse direction when it reaches the last step in the sequence, and again when it gets back to step 1. If the configured length of the sequence is 6 steps, then after step 6 the next steps will be 5, 4, and so on, back to 1. At that point it will again change direction and count through 2, 3, etc. When the up/down mode is selected, and the CYCLE switch is in the SINGLE position, the sequencer stops when it returns to step 1.

The SET END button serves two purposes. Its primary function is to allow you to set the desired length of a sequence. This is done by pressing the SET END button once and releasing it; the LED for either step 1 or the current end step will begin to flash rapidly. Repeatedly press the SET END button to advance the end step (you have to do it quickly); when it reaches the step you want, stop pressing the button. After a second or two, the flashing will stop, and then that step will be the final step in the sequence. This is effective for all sequence modes -- up, up/down, and random. Note that when you switch the sequencer to 3x8 mode, it will automatically set step 8 as the end step. When you switch back to 1x24 mode, step 8 will remain the end step, and you will have to use SET END to reset it to a longer sequence if you want. (Or cycle the power.)

The SET END button is used with the MANUAL STEP button to select two "hidden" modes of the sequencer. The normal start mode is the "reset" mode; in this mode, any time the sequencer starts, it first resets to step 1. Pressing MANUAL STEP while pressing and holding SET END selects the "continue" mode. In this mode, when the sequencer starts, it resumes with the step after the one it stopped on. Doing the opposite of that – pressing SET END while pressing and holding MANUAL STEP -- sets the cycle mode to the random mode. In this mode, each time the sequencer advances, it selects a step at random. Although I haven't attempted to do an analysis of the distribution, it seems to be pretty uniform. One thing to note is that the code presents the same step from being selected twice in a row. This is a nice feature when generating random notes; in a random-note sequence, it tends to be jarring to the listener to hear the same note sound twice. The CYCLE and SEQUENCE switches have no effect when the random mode is engaged; the sequencer runs continuously until stopped. Either of these hidden modes may be disengaged by repeating the button sequence for that mode, or by cycling the power. 

Step Controls

The heart of the Q119 is in the 24 step blocks, which are organized in three rows of 8 steps each. Each step block consists of a single knob, which is used to select the output voltage for that step, and a red LED that indicates when the block is active. To improve finger room for the knobs, the odd-numbered steps have the knob on top and the LED on bottom, while the even-numbered steps are the reverse. This results in a rather amusing pattern of lights moving in a zig-zag when the sequencer is running, which some performers object to, but I think it actually improves recognition of which step is active. The LEDs also function with the SET END button in selecting which step is to be the last step in the sequence. Changing the setting of a knob will be reflected immediately in the output if the sequencer is on that step (the step's LED is lit), whether running or stopped.

Q119 step controls and LEDs, with row outputs on the right.

The organization of the step blocks into three rows is not merely a visual presentation. The Q119 has two operating modes, known as “1x24” and “3x8”, and selected by the MODE switch. In the 1x24 mode, the sequencer drives a single sequence of up to 24 steps long, using the three rows in series. When the sequence runs, it will proceed across the top row until it reaches step 8, then resume on the second row at step 9, going to 16 and then jumping to the third row at step 17. At step 24, it jumps back to the first row and step 1. In the 3x8 mode, the sequencer drives the three rows in parallel, producing three sets of control voltages at the three BANK outputs. The first step is steps 1/9/17, then it proceeds to 2/10/18, and so on, up to 8/16/24, at which point it returns to 1/9/17. The LEDs for the proper steps in each row will light simultaneously, as opposed to the 1x24 mode, in which only one LED is lit at a time. (In either mode, the SET END button can be used to make the sequence shorter than the maximum, if desired.)

Control voltages

The control voltage knobs are not linear with respect to output voltage. With the OUTPUTS switch in the -5/+5 position, one might expect that the zero position (12 o'clock) is 0 volts, and each major hash mark is a difference of one volt. The first statement is true, but the second is not. From 0 to +1 on the indexing is a difference of about 0.6V. The steps get larger moving further away from the zero position, finally reaching plus or minus 5V at the +5 and -5 positions respectively. With the OUTPUTS switch in the 0/+5 position, something similar happens: the full CCW position (-5 in the indexing) is 0V; -4 is about 0.3V, -3 is about 0.7V, and so on. In both modes, the steps get larger as you move farther away from 0V. This is something of a benefit if you can use the ADD offset control (further down) so that you can keep most of the steps near the 0V position, which makes it easier to make fine adjustments. However, it is confusing if you expect to be able to look at the indexing and dial up a desired voltage; that isn't straightforward. If you need a specific voltage, it is best to check it with a voltmeter. If you are running the output into a VCO and trying to tune notes, it is usually better to either let the sequence run and tune it by ear, or if that doesn't work for you, single-step the sequencer with the MANUAL STEP button and check each note against a tuner. 
Output section, with row (bank)
outputs on the left, and the master
outputs on the right.


The output section contains the master outputs, a set of row outputs for each row (labeled BANK 1/2/3), a knob for adding lag (portamento), and a knob and jack for adding an offset voltage to the master output. The master output is usually used when the sequencer is operating in the 1x24 configuration. The master OUTPUT jack outputs the voltage from the currently active step. The GATE jack outputs a gate which rises when the sequencer advances to the next step, and falls some time after, as determined by the GATE WIDTH knob in the control section (or the pulse width of the external clock, if an external clock is being used). The LED next to the GATE jack lights when the gate is active. If the MODE switch is in the 3x8 mode, the master OUTPUT jack will have the sum of the active steps from each row. This isn't usually what you want, but it does have creative possibilities. Note that the OUTPUT jack is active all of the time, including when the sequencer is stopped. The GATE output remains low when the sequencer is stopped.

The row output jacks are active when the corresponding row is active. When the sequencer is in the 3x8 configuration, the top-row OUTPUT jack outputs the voltage selected from the currently active step in that row, and the other two row OUTPUT jacks perform the same function for their rows. All of the GATE jacks pulse together in this mode. In the 1x24 mode, the row output jacks are only active for the row that contains the currently active step. When the current step is not in that row, the OUTPUT jack outputs the minimum voltage (0V or -5V depending on the OUTPUTS switch setting), and the GATE jack remains low.

Master output modifiers

The GLIDE and ADD knobs only effect the control voltage master output. The GLIDE is a conventional lag processor that acts on the control voltage output. The ADD knob adds an offset voltage to whatever voltage is present at the master output; this has a number of obvious uses, such as transposing sequenced notes, or bringing them in tune with another instrument. If a cable is plugged into the ADD INPUT jack, that is also added to the master output.  To sum it up, the voltage at the master output consists of:
  •  The current step control voltage (or the sum of the three steps, in the 3x8 mode)
  •  The ADD knob voltage
  •  The signal present at the ADD INPUT jack

Interaction with another sequencer

The DONE OUTPUT jack sends a trigger signal at the time that the sequencer advances from the last step back to step 1 (or would have, except for the CYCLE switch being in the ONCE position). This allows you to operate two (or more!) Q119s in a round-robin fashion, by setting their cycle switches to ONCE, and then patching the DONE output of one into the START input of the next. When the first one finishes, it will start the second one, etc. By careful adding of the outputs, you can create sequences of 48 or more steps. (You could take the master OUTPUT jack of one Q119 to the ADD INPUT of the next one to combine the control voltages, but you'd need some external module to combine the gates.)

Interaction with other modules

Some performers who use an analog sequencer to produce note sequences find it easier to set up the sequencer when they can run the outputs through a quantizer. offers a quantizer, the Q171, which has features designed to make it complementary to its sequencers. In particular, it has three quantization channels, so that you can quantize all three rows when using the 3x8 mode, and it has gate inputs to force quantization to only occur on the note gates, which can help avoid the “dithering” problem (where the quantizer jumps back and forth between adjacent notes). However, other quantizers could certainly be used. 

Output selector switches, such as the Q962, have potential uses with the Q119.  The DONE OUTPUT can possibly be used to cycle between different bus selections or outputs, for various purposes.


It seems a bit unfair to describe the Q119 as an “entry level” sequencer, since it is a quite capable module. It is not as full featured as, say, the Moon Modular 569, the GRP R24, or's own Q960. Then again, it also costs a lot less than those others; the direct-sale price of $560 USD is a bargain in the world of analog sequencers, which generally tend to be expensive. (Moon's direct-sale export price, excluding VAT, for the 569 is E1258.77, which at the exchange write on this date, 7 Feb 2018, works out to $1545.41 USD.) The main thing that those sequencers have that the Q119 lacks is flexibility; they typically have features like individual gate outputs for each step and reset trigger inputs. Then again, they sometimes require either additional aid modules or fancy patching to perform functions that the Q119 has built in. So yes, the Q119 is a good choice for someone who has no experience with analog sequencing and wants to get practice with it, but it's a module that will continue to be useful in your case even after you purchase one of the higher-end sequencers.

Demonstration videos

This first video is a basic demonstration of the Q119's different cycle modes.  The 1x24 and 3x8 modes are demonstrated at different speeds, with up, up/down and random sequencing, and the single and continuous cycle options.  The use of the SET END button is also demonstrated.

This second video illustrates using the Q119 in the 1x24 mode, with a sequence length of 14 steps, to generate an approximation of a familiar Synergy sequence (the one from which this blog takes its name). Driven by a pulse wave from a Q106 VCO in LFO mode, it is modulating another Q106, whose triangle output is going into an MOTM-440 OTA filter. Envelope is from a Q170 Envelope++, and it is controlling a Q109? VCA. Note that this actual patch is only an approximation of the original, for demonstration purposes.  Please excuse the rough tuning; I don't have a quantizer and I didn't spend a lot of time on tuning the notes. Nonetheless, if you listen to much Synergy, you should recognize it.  I use an external clock and gradually speed up the sequence, in the same manner as the original.  Just before the end, I take it up to a faster speed than Larry Fast's old Moog 960 was capable of, just to show off the Q119 a bit.

You will notice something at the start of the video: there seems to be a "skip" at the very start of sequence, between the first and second notes.  This is due to the fact that I'm using an external clock in this video.  (When I reach to something above the top of the picture, I'm reaching for the requency control of the Q106 that is serving as the clock source.)  The Q119 syncs its own clock when it is instructed to start, but it has no way of making an external clock sync to it.  So when I start the sequencer, it starts at some random point in the external clock's cycle.  If this is part way through the cycle, then the first step will be short, time-wise, and that is what you hear here: the first step occurs on the START button press, and then the next step occurs on the next clock transition, but I hit START at some point in the middle of the clock cycle, so the interval between the first step and the second step was short.  If I had wanted that interval to be precise, I could have watched the LED on the Q106 and pressed START at the start of the cycle.  Or I could have fed an external trigger source to the Q119's START jack, and to the Q106's hard sync input.  

This third video illustrates using the sequencer in 3x8 mode. What is happening here is that the top row is being used to modulate a Q106, whose sawtooth wave is going into an SSL 1310 digital delay that is being modulated by an LFO. (There is no filter in the patch.) The bottom row is being used to generate a gate signal – turning the knob up causes the gate to be “on” on that step, and turning the knob down causes it to be “off”, so that that step does not sound. As the sequence plays, I play with the bottom row to make different notes in the sequence sound.

Sunday, November 1, 2015

Review: SSL 1250 Quad LFO

The Synthetic Sound Labs Model 1250 Quad Low Frequency Oscillator is what it says it is: four LFOs in one panel, formatted in the MU (Dotcom) modular synth format.  It's a pretty simple module.  The panel is divided into five sections: four sections are each for one LFO, and a bottom section contains the output jacks.  Each LFO has three controls: a rate knob, a waveform select switch (sine and square waves are available), and a peak indicator lamp which is also a pushbutton.  Pressing it switches the LFO between high range and low range.  The lamps are red LEDs and actually look much redder than in the picture to the right; I think the infrared filter on my camera prevented the deep red from registering.

This is an LFO meant to drive slow, evolving patches.  On the high range, with the knob full clockwise, the period is about 22 milliseconds, which works out to 45 Hz.  With the knob at 5 (straight up), the period is 50 ms, or 20 Hz.  As you turn the knob further left, the period increases linearly, which per the law of reciprocals means the frequency decreases exponentially.  With the knob at 2 (the 9 o'clock position), the period is 150 ms, a frequency of 6.6 Hz.  At the low end of the knob's travel, between 0 and 1, the change is much more than linear -- with the knob full CCW, I measured a period of 80 seconds.

If you want really slooooooooow, switch to low range.  With the knob full clockwise, the period is about 1200 ms, or around 0.8 Hz.  At the 5 setting, it's 3 seconds.  At the 2 setting, it's 8.5 seconds.  At the 1 setting, it's 36 seconds.  With the knob full counterclockwise… I was not patient enough.  After three minutes, it had climbed from zero volts to +0.45V.  If I've done my math right, that's a cycle time of about 45 minutes!  The cycle indicator light starts to light up when the sine wave rises +1.5V, and reaches full brightness by +3.5V; it goes out when the sine wave drops below 1.5V.  (This is true whether the sine or square wave is selected.)  At moderately slow rates, it's rather hypnotizing to watch.  I did a quick check of all four oscillators to make sure they were all calibrated the same, and didn't see any noticeable differences.

Looking at the waveforms on the scope: The square wave looks good.  The sine wave is a bit distorted; it looks a bit triangle-ish.  There's a distinct corner at the turn point, and the rise and fall portions look a bit straight-lined on either side of the horizontal axis.  (A perfect sine wave is straight only right on the axis; it has at least a little bit of curvature everywhere else.)  It's not as bad as that makes  it sound; most of the waveform looks like a good sine wave, and using it to modulate a VCO, I didn't hear any abrupt reversals in pitch rise and fall, as one would if the VCO were modulated with a triangle wave.  Also, the sine wave doesn't quite make it to the 5V rails; it turns at about +/- 4.5V.  The square wave looks good.  There are no visible changes or variations in the waveform with frequency.

The build quality looks good, up to SSL's usual high standards.  There is one main board and a smaller jack board, as you can see in the photo to the right.  (That blurry white cable with the colored wires coming out is my tacky homemade power cable.)  Most of the components are surface mount.  The main board is flush to the back of the panel, and the jack board only stands off about one inch (2.5 cm), so there should be no problem installing the 1250 in the most shallow cabinet or skiff imaginable.  The panel is standard MU construction and all of the dimensions are correct.

The SSL 1250 serves a basic but essential function in a modular synth: to avoid highly repetitive modulations that can become fatiguing to listen to, you need to be able to mix several LFOs to create modulation shapes that are more complex but not totally chaotic.  The 1250 does this job admirably.  And the blinkylights factor is high too.  The one improvement I might suggest is some onboard way to output a combined waveform without having to use a separate mixer.  If the output jacks were chained -- that is, the output of a given jack combines with the next higher numbered jack when no cord is plugged in -- that would be useful.

SSL is at  They sell both direct and through dealers. 

Thursday, August 27, 2015

Analog computers and synths

A few weeks ago, a poster at VSE asked a good question: To what extent, if any, did the design and use of analog computers in the mid-20th century influence the development of music synthesizers?  My first thought was, "probably not much".  Then I did some research...

First, let's go over what an analog computer is.  An analog computer, put simply, is a device that accepts input parameters which are represented by something inside the computer.  It performs computing functions through mechanisms and/or electronic circuits, and the outputs are expressed by quantities of something the mechanism can produce.

A Philbrick K-3 analog computer, circa 1950.  From the Philbrick Archives.

Analog computers preceded the development of electricity.  The first, simple analog computing devices go back to the Middle Ages, but significant ones started appearing during the pre-industrial scientific discovery period from 1600 to 1800.  Generally they relied on sliding or rotating parts to represent measurements which were input or output.  A simple but important example is the slide rule, invented in the 17th century.  A basic slide rule multiplies two numbers by positioning one operand on a sliding scale against a fixed scale; the amount by which the sliding scale is moved represents (by reading off of a scale) the product.

In the early 20th century, a number of powered analog computers were invented to do specific calculations.  An early driver behind the development of this technology was the need for a device called a "gun director".  This was a computer that computed the elevation and azimuth angles at which an artillery piece needed to be pointed in order to hit a target, given the range to the target, the wind, the weight of the shell being fired, and possibly other factors.  The Norden bombsight was a famous electro-mechanical analog computer deployed by the Allies during World War II.  To use it, a bombardier looked through a sight glass to find the target to be bombed.  From the pointing angles of the sight, and the rate at which the bombardier had to move the sight in order to keep it on target, the bombsight computed the heading that the aircraft needed to fly, and the time at which the bombs should be dropped.  In this computer, the quantities being computed were represented by the movements of levers or gears.  (The bombsight was usually coupled to the airplane's autopilot so it could actually fly the aircraft during the bombing run, and to the bomb racks so it could release the bombs at the right time automatically.)

Norden bombsight (top left) and servos controlled by the bombsight.

Electronic analog computers started to appear around 1930. As was generally the case of the mechanical analog computers, most of the early electronic devices were hard-wired to perform a specific computation; because of this, early uses were limited to problems that were both important and difficult, enough that it was worth the cost to build a computer.  An early example was a device known as the "AC network analyzer", which was built to solve problems that electrical power utilities were encountering as individual power stations were being combined into large grids.

In 1938, electrical engineer George A. Philbrick, then employed by the Foxboro Company of Massachusetts, wrote a proposal for an electronic analog computer that would model various types of closed-loop manufacturing processes.  One of the problems that Philbrick had to solve was how to design circuits that would perform the needed math operations in a general sense, that is, not specific to a particular problem.  In 1943, Philbrick was working on a contract with the U.S. Army to devise improvements to the M9 gun director, which had been built by Bell Labs.  It worked, but it was too slow to compute in real time.  Philbrick came in contact with Loeb Julie of Columbia University, who had devised the first experimental operational amplifiers.  (Yes, there were op amps decades before the first integrated circuits.)  Philbrick realized that Julie's op amps could be used to perform a variety of analog computing math functions, and he began working on his own improvements.

Philbrick K2-P op amp

After WWII, Philbrick started his own company, George A. Philbrick Researches.  The company was heavily involved in both analog computing and commercial op amp design and manufacturing.  The company published a widely regarded collection of papers and notes concerning analog computing -- system design, circuit design, programming, and operations.  Analog computers were becoming more compact, and general-purpose units were appearing that offered a number of function modules which could be interconnected by the user in any desired configuration using patch cords.  In fact, Philbrick's company developed the idea of a "modular computer", in which individual function blocks could be purchased and combined as needed to apply to a problem -- a concept very similar to the modular computers that would come later.  At some point Philbrick hired a certain young electrical engineer, one Alan R. Pearlman, who took an interest in the op amp business.  So much so that, in the early 1960s, Pearlman and another Philbrick employee broke away and established their own company, Nexus Research Labs, which continued their work in the op-amp and analog computing business.

Philbrick K3 analog computer modules.  From the Philbrick Archive.

If Pearlman's name doesn't sound familar, look at his initials -- A.R.P.  In 1966, Pearlman's group sold Nexus Research Labs to Teledyne.  The sale made Pearlman wealthy, and he used some of that wealth to found ARP Instruments.  Look at the photo above.  Looks vaguely familiar?  The Philbrick analog computer systems were modular.  There were about 10 function modular that the user could purchase and configure in a case as needed.  Compare to this:

ARP 2500 model 1947 voltage controlled filter

However... The first of what we consider synthesizers today didn't come from Pearlman.  The two men who are generally credited with developing the basic building blocks of the analog synthesizer -- the voltage controlled oscillator, filter, and amplifier -- are Robert Moog and Don Buchla.  Moog has an obvious, if indirect, connect to Philbrick via Columbia University, where Philbrick and Loeb Julie worked on the first op-amp designs in the 1940s, and where John Ragazzini and Rudolf Kalman had continued to work on analog computing concepts through the 1950s.  The Columbia-Princeton Electronic Music Center opened at Columbia in the mid-1950s, but it is not clear how much cross-fertilization there was between it and the analog computing labs.  Moog just missed experiencing the RCA Synthesizer, which was installed at the center in 1958; he had graduated in '57.

Little is written about what Moog actually studied or did at Columbia (far more is written about his theremin side business by which he paid his way through school), so further investigation is difficult.  He did get his degree there in electrical engineering, and in a mid-1950s electrical engineering curriculum, he most certainly would have had instruction on computer circuits, both analog and digital.  There were probably analog computers to use, and possibly they were Philbrick units like the one pictured above, thanks to the connection to the university via Loeb Julie.  Where did Moog come up with the idea to make his first synths modular?  Did he spend some time with a Philbrick analog computer at Columbia?  Did he, perhaps, try to coax sound synthesis out of it? 

Buchla is even more of a puzzle.  There is almost no information available on the Internet about what he did prior to founding Buchla and Associates in 1962.  It is known that he was involved with Morton Subnotick and the San Francisco Tape Music Center, which was a tape studio and had little if anything to do with analog computers.  He was involved in some way with the University of California, Berkeley (it's not clear if he was actually a student or faculty there or not), which at the time was the world's foremost center of nuclear physics research, a field in which a considerable number of analog computers were used to model nuclear reactions.  Buchla studied physics (along with several other fields) and probably would have come into contact with the nuclear physics program's analog computers.  To what extent this influenced his later thinking about synthesizers is difficult to say.

So to answer our question: did analog computers influence the development of analog synths?  The answer, at this point, is "maybe".  We know that Pearlman was heavily involved in analog computers, but he came in a little after Moog and Buchla.  We know that Moog was at Columbia at a time when the school was involved in both analog computing and electronic music, and we can see similarities between his modular synth designs and some of the modular computer designs that he might have worked with.  Buchla is less certain, but he probably would have at least seen analog computers at Berkeley. 

For more information about George Philbrick and his pioneering company (it's a worthwhile read for anyone interested in electrical engineering history), see the Philbrick Archive at 

Sunday, October 19, 2014

Fetishizing the Moog modulars

Lately I'm noticing a big surge of interest in the vintage modular Moogs. Now, this in itself is not a bad thing. It's a good thing, and not only from the historical preservation sense.  It's always good to have a perspective of history, and to see how Bob Moog and his compatriots made their decisions and went about doing things without access to all of the technology we have today. Remember, in 1963 when Moog and Buchla built their first modules, the integrated circuit was still largely confined to Fairchild Semiconductor's labs. The commercially package operational amplifier was a big ugly box that plugged into a tube socket and contained a pair of 12AX7 tubes inside it. There were no OTAs, no 4000 CMOS logic; Doug Curtis was in elementary school, and Ron Dow had not yet gone to Dave Rossum and Scott Wedge to beg for money (which was a good thing, since Rossum and Wedge were themselves high school students and didn't have any money).

Yes, things were different back then. Moog (and, independently, Buchla) had just thought of the idea of “voltage control”, in which he imagined that a generated signal might be able to remotely control the functioning of another circuit, thereby increasing the possibilities for more animation in electronic music, e.g., that the output of one oscillator could control the frequency of another in order to introduce vibrato, without a person having to constantly turn the frequency knob up and down. This was new territory; at first Moog had no idea how to do it with components that were available to him. As he attacked the problem, he made it work, but there were a lot of compromises: many components being made to do things that they weren't designed to; use of some expensive components which forced cost cutting in some other areas, and the necessity to keep the circuits confined to a reasonable sized package. There were also things to consider like what we now call the “user interface” was to function. (We all know the story of how the synthesizer came to be primarily a keyboard instrument: the switches of an organ keyboard, wired to a resistor grid, worked a lot better than primitive pitch-to-voltage converters and provided an interface that looked familiar to musicians.)

Consider Moog's first voltage controlled oscillator, the model 901A/B duo. At a list price of several hundred dollars in 1964, what you got for a 901B VCO was a basic oscillator with four waveform outputs. If you wanted volts/octave response (which was essential for any kind of tonal music), you had to also buy the separate 901A driver module which contained the exponential converter. And oh by the way, the VCO contained absolutely no temperature compensation, which meant you had to constantly re-tune as the circuits warmed up and/or the room temperature varied.

As another example, consider the Moog 904B VCF. Here's a photo of one:

Moog 904B.  Photo courtesy of David Brown at

Note that it's a 2U wide module and how big it is and how much empty space there is on the panel. Why is it so big? Because the circuit board behind the panel needed to be that big in order to cram all of the circuitry in.  Here's another, more drastic example of that sort of thing:

Moog 905.  Photo courtesy of David Brown at
This is the 905 reverb.  Lots of wasted panel real estate?  You bet.  It's that large because it uses a spring reverb tank, which is mounted in the module itself, right behind the panel.  Modern modulars that offer spring reverb modules mount the tank remotely, somewhere in the rear of the case.  Although, oddly, the Club of the Knobs reproduction of the 905 retains the same 2U wide panel design, even though it uses remote mounted tanks:

Club of the Knobs C905 reverb.  Photo from COTK's Web site.
This is taking authenticity to a bit of an extreme.  Clearly, a 1U panel would have been sufficient.  The Eurorack users always say that all of the large format modulars take up too much space, and this sort of thing doesn't help.

There were a lot of things about the Moog modulars that were different from today's modulars and made them not so easy to interface to or work with. Most Moog VCOs and other signal generators output a signal that is only 1.5V peak to peak. This I assume was a choice made based on typical use of signals as modulation sources, but, for example, it means that the output of a VCO or an envelope generator can't be made to drive a VCF though its full frequency range without being amplified.  For reasons totally unclear to me, MOS-LAB recently decided to go back to Moog's 1.5V standard for its reproduction of the 901B and 921B VCOs. 

MOS-LAB 921B VCO.  Photo courtesy of
And there's the infamous S-trigger signals. On a Moog modular (and other vintage Moog synths such as the early Minimoogs), something that generates a trigger or gate signal does not output a voltage pulse. Rather, the output is a simple transistor that is saturated in the “low” or “off” state, shorting the output to ground, or cut off in the “on” or “high” state, which leaves the output “floating” electrically. The output expects that whatever trigger/gate input it is patched to will “pull up” the output by applying a voltage through a resistor. When the output is in the high state, its voltage rises to the pull-up voltage; when it is in the low state, it shorts the output to ground, and the pull-up resistor limits the current that flows to ground. We've all seen that the modular Moogs use the infamous “Cinch-Jones” two-bladed connector for trigger outputs and inputs, requiring a separate type of patch cord to connect them (and thus the modular Moogs do not have fully unified patching). This is why; if a trigger/gate input, with its pull-up, were inadvertently connected to a signal output, it could potentially damage the output circuit.  But it's a pain because of the special cable needed, and because you need an adapter to interface any external trigger or gate source.  Mercifully, neither MOS-LAB nor COTK has chosen to use the S-trigger on their Moog reproductions, even though the connector itself is still available

Moog 911 envelope generator; note Cinch-Jones gate input connector at bottom left.  Photo courtesy of David Brown at
And last but not least, there's the cost of construction using those "authentic Moog" methods and circuits.  As I wrote above, there were a lot of places where the Moog designs had to use methods and techniques that were a lot more expensive (such as building op-amps out of discrete circuitry) because more capable components weren't available at the time.  Consider: offers two step sequencers -- the Q119 and the Q960.  The Q960 is a fairly faithful recreation of the Moog 960 sequencer design, up to and including the incandescent lamps which indicate the active stage (which most users replace with LEDs because the lamps burn out frequently).  The Q119, on the other hand, has most of the same capabilities and controls but is microprocessor controlled, and all of the indicator lamp are LEDs.  The two share many capabilities -- but the Q119 is about $300 less expensive, plus in order to duplicate the Q119's 24-step mode with the Q960, you need to add a Q962 sequential switch, at an additional $160. 

Moog 960 (top) and Q119 (bottom).  Top photo courtesy of David Brown at; bottom photo courtesy of

So given all of the above, I'm starting to wonder if the current market isn't fetishizing the modular Moogs a bit much. Of course, the Dotcom/MU format was based on the physical dimensions of the original Moog modulars, and Roger Arrick's designs continue to take certain design cues from the Moogs, such as the black panel background and the knob style. But Arrick started out with fresh circuit designs using contemporary electronics technology. And he's no slave to the Moog look and feel; he has never hesitated to make a module smaller than the functionally equivalent Moog module when the circuit design allowed for it. The other notable thing was that Arrick avoided both the weird mix of power supply voltages and the edge connectors that Moog modules used; set the standard power for the MU format at +/- 15V and +5 volts, and the flexible power supply harness doesn't limit the modules' board mounting geometry the way the Moog edge connectors did. (The Dotcom MTA-100 power connectors are also a lot less prone to corrosion problems than the Moog edge connectors are.)  As for Club of the Knobs, they started out copying the Moog modules, but soon realized that simply duplicating the Moog lineup would be too limiting.  And although they continue to stick to the general Moog format, they have long since blown past the limitations of the original Moogs with module designs that Moog could never have thought of or implemented with the technology available at the time, such as the C950A MIDI interface / arpeggiator. 

That's why, to be honest, I really don't want to see a big comeback of slavish Moog-modular clones. Even putting aside the difficulties of obtaining exact replacements for the vintage Moog parts, the 1960s Moog modulars were just all-around limited compared to what is available today. Yes, it's great that Moog has been able to sell several of the $1.5M copies of the Keith Emerson modular; the units will instantly be valuable collector's items as well as being highly educational, and more power to Moog for being able to build them and sell them at that price. What bothers me is the people might get the idea that the 5U formats are all about duplicating what has been done in the past, with the implication being that you have to turn to the Euro format to find any modern or fresh ideas. That would be a self-limiting move for 5U.  And as someone who wants modern capabilities but prefers to work with 5U formats, I don't want to see that happen. 

Thursday, October 9, 2014

Review: Izotope Iris

Iris is a plug-in from Izotope, a company that's better known for products like Trash, Stutter Edit, and "make my track the loudest thing on the dial" mastering tools.  Despite what that might suggest, Izotope has a lot of background in Fourier analysis and spectral processing tools, and they put most of it into Iris.  The difference is, with Iris, you get to control it yourself. 

Iris employs a method of synthesis whose availability to the masses is relatively recent, and that I had no previous experience with -- spectral editing.  So when I first installed it and fired it up, I wasn't at all sure what to expect.  But I have been pleased, if at times a bit baffled, at the results I've gotten so far.  With that said, let's dig in.

Spectral Editing

What exactly is spectral editing?  Well, it involves taking a sample of a sound, and picking out bits of its spectrum over time that you want to reproduce, excluding the rest.  Iris presents a sample that you choose in a two-dimensional window, with time on the horizontal axis and frequency on the vertical axis.  Energy at a given time/frequency coordinate is represented by a pixel on the screen, with the brightness of the pixel indicating how much energy is present.  You use various methods of selecting regions of the time-frequency space that you want to reproduce.  Then, when you play the note, it reproduces the regions you select.

The User Interface

Here is a shot of the screen you are presented with when you first start the plug-in:

Iris initial view

The big window with the Izotope logo is where you will load a sample to edit.  The toolbar over to the left consists of drawing tools that you use to select regions to be reproduced.  The controls on the right allow you to select a layer, tune an edited sample, set up an amplitude envelope for the layer, and route modulations.  For each patch, Iris allows up to three layers, plus a "sublayer" which uses fixed waveforms but still allows for spectral editing.

The buttons at the top right switch between different views, or presentations of the user interface.  Any of the three layers or the sublayer may be selected for editing by clicking on the "1", "2", "3", or "Sub" buttons.  There are two other views available by clicking their buttons at the upper right of the window: the All view and the Mix view.  We'll discuss these views later.  For now, we'll concentrate on the single-layer view, which is where you will probably do most of your work.

Sample Loading and Manipulation

When you load a sample, either by dragging and dropping one into the window, or by selecting one from Iris' library via the browser, you are presented with a window that looks like this (click on the image to see a full size version):

(If you have a copy of Iris, the above sample is "SK8 Circuit Singer".)  What you're looking at here is a spectral energy (or power, if you want to think of it that way) display. 

Note the frequency scale along the left edge; it indicates that this sample has its highest-energy bands in the 1.5/2.0 KHz range.  If you look at the right side of the on-screen keyboard, you'll see a little tag at F#6; this indicates what the sample's "natural" note is, i.e., at what note the sample will play back with no pitch shifting.  You can click and drag this tag to change the setting.

Immediately above the sample window you can see a gray bar (shown above this text) with an arrow at each end.  You use this to set start, stop and loop points (although Iris will do it automatically, as will be explained later).  This is all fairly conventional and similar to other types of sample editors; it allows for one-shot, forward looping, and alternate forward-and-back looping.  The bar above this graphically portrays the sample's amplitude envelope, and the gray region indicates the bounds of the currently selected loop, which may be different from where the arrows are set if Iris has chosen the loop points automatically.  Below and to the left of the screen are rulers; the one on the left (see to the left of this text) displays frequency, and the lower one (below the next paragraph) displays sample word counts.  You can scroll the view horizontally and vertically by grabbing one of these rulers and dragging it.

(If the rulers don't appear, click on the wrench icon at the upper left to bring up the preferences dialog, and click on "Show Time Ruler" and/or "Show Freqeuncy Ruler", as you prefer.  You can also choose time or number-of-samples scales for the time ruler, and there are several choices of units for the frequency ruler.)  

Other tools for view manipulation are at the top left corner.  When you click on the top one (the one that looks like a screen with a magnifying glass over it), it zooms in or out enough to show you all of the drawing that you have done in the layer.  The second one (the magnifying glass with an X) does a full zoom out and shows you the entire sample.  The third one is an arbitrary zoom tool; when you click on it, you can draw the diagonal of a rectangle that you want to zoom in on.  The hand tool at the bottom is a scroll tool; click on it, and then you can grab and drag the content of the window.


(A note here: Izotope refers to edited/painted regions of spectral content as "selections".  To avoid confusion with the conventional use of the word "selection" in computer UI phraseology, I'm avoiding the term here; I'll refer to such areas as drawn/painted regions.)

You actually choose spectral content to be played by drawing in the window using the tools from the center left margin (shown to the right of this text).  A better analogy for most of the tools is actually painting; you move the tool and as you do so, it paints a certain area which is to be played.  The three tools at the top select rectangular areas.  The top tool paints a region of time; you choose the tool and then drag horizontally across the time band you want, and it paints all frequencies within the time band.  The third tool does the same for frequency bands; you drag up or down and it paints all content across time within that frequency band.  The middle tool selects as rectangular area.

(When you first load a sample, before you paint anything, if you play a note you will hear the unedited sample.  This is turned off when you do your first painting operation on the sample.  If you paint an area and then erase it, so that nothing is painted, you will hear nothing.)

The fourth tool, the brush tool, allows areas to be painted freehand.  Click and hold and move the mouse around to paint.  Painting is additive, so if you paint over any areas that are already painted, nothing happens.  You will notice when you select the brush tool that a slider control appears above the on-screen keyboard, just to the left of the word "KEYBOARD".  This allows you to set the brush diameter.

The bottom tool in the tool selection is the eraser.  It works similarly to the paintbrush, except that instead of painting, it un-paints already painted areas.  The size slider also works with the eraser.

The lasso tool allows you to cause an arbitrary-shaped area to be painted by drawing a boundary around it.  Click and hold and freehand the boundary.  If you don't complete the loop, it will be completed for you with a straight line connecting the start and end of your draw when you let the mouse button up.

The remaining tool is the "magic wand" tool.  This is a bit hard to explain and I don't quite have a handle on exactly what it does.  When you use it to click on a painted region, it will select content that is harmonically related and coincides in time with the region you selected.  So far, for me it's a "try it and see what it does" thing.

The next set of tools allow you to make bulk changes to the areas you have painted.  Iris maintains an idea of the bounds, with respect to frequency and time, of the areas you have painted.  When you click on the first one (with the horizontal areas) it inverts all of the painted and un-painted areas within the frequency bounds of where you have drawn.  Similarly, the tool with the vertical arrows inverts within the time bounds.  The one with the diagonal arrows inverts across the entire sample.  The last two tools, with the four arrows and the box with an X, paint or un-paint the entire sample, respectively.

The two tools below that are the undo and redo buttons for painting and drawing.  The bottom tool is a preview button; it plays the spectral-edited sample without any pitch shifting.

An example of an edited sample:

Note the four painted regions, including the narrow one at the bottom, and the two vertical dashed lines that indicate the loop start and end points.  These illustrate the results with various drawing tools: The area at the left was freehanded; the area at the lower right was drawn with the lasso tool, and the area at the upper right was drawn as a rectangle and then attacked by the eraser. 

Brushes and Erasers Don't Scale with Zoom

Note that when you zoom in or out in the view, the size selection for the brush and eraser tools doesn't scale; it remains the same on-screen width at all zoom levels.  This means that if you paint a line, zoom in, and then paint another line, the second one will be narrower than the first one.  

Auto Loop Points Adjustment

Whenever you do any drawing or add or alter painted regions, the sample loop points automatically move so as to move the leftmost and rightmost edges of all the painted regions.  This is done to eliminate periods of silence that would result if the entire sample were played.  Refer back to the above figure and notice how Iris has automatically set the loop points in this example, as indicated by the vertical dashed lines, and gray area in the waveform display above the editing window.  You can override this and set the loop points manually if you wish, and you can also set the sample start point independently of the left loop point.

Dragging Painted Regions

When any of the above drawing tools is selected, except for the eraser or magic wand, moving the cursor over an existing painted region causes the cursor to change to a hand icon.  When this occurs, you can grab and drag the region that the cursor is positioned over.  Note that Iris keeps track of contiguous painted regions, as indicated by the crawling dashed lines surrounding them, and it updates these each time you make a change; touching or overlapping painted regions will be combined for dragging purposes.  Also note that dragging moves the region, not the sample content underneath.

The Undo Trap

You might notice that there are two sets of undo buttons on the user interface: one down near the bottom of the left margin, above the preview button, and another set at the top margin to the right of the patch selection controls.  The ones on the left, you can use to undo and redo drawing operations.  Beware of those ones at the top.  They undo operations like sample selection and parameter changes.  If you select a sample, do a bunch of spectral editing on it, and then hit the undo button at the top of the window… it will de-select the sample and you'll lose all the drawing you've done.

Layer Controls 

 There's a small set of layer controls for each layer, corresponding to other layer-specific features.  The four knobs above the amp envelope allow you to set tuning for the layer, control its overall volume, and pan it in the stereo image (the samples are all mono).  Each layer has its own amplitude shaper; the amplitude envelope is an entirely conventional ADSR envelope, which you set up graphically by dragging on the little squares.  Each layer has an LFO which can be routed to pitch or amplitude (you make it active by clicking on the little power-button symbol next to the word "LFO").  You can set the effects sends for this layer or for the whole mix, depending on the effects mode; I'll say more about this further down.  
Several significant controls are in the text box under the word "CONTROL".  Clicking next to the word "Loop" allows you to choose the loop mode for this layer.  Forward, reverse, alternating, and one-shot in either direction are available.  

The "Pitch" item is a pop-up menu with three choices.  Iris' normal mode for sample playback is to pitch the sample by speeding up or slowing down playback, the way that early hardware samplers did (this is misleadingly labeled "Resample" in the menu).  This of course means that the sounding of a higher-pitched note will be compressed in time compared to a lower note.  Choosing "Radius RT" engages a pitch-shifting algorithm that makes all notes play back in the same amount of time.  On my i7-based Mac Mini, I found the Radius RT default settings, and the user manual, to be excessively cautious about CPU usage; the manual warns that the pitch shifting algorithm is CPU intensive and accordingly the factory default is to only allow 4 voices at a time to use the algorithm.  However, with four notes playing on a layer set to Radius RT, I didn't observe CPU usage to increase much, and I was able to boost the max voices up to 10, and the octave range to +/- 3 octaves, without any problems.  I would not hesitate to use Radius RT for any patch where I wanted the playback or loop time to be constant.  (On the other hand, allowing loop times to vary with pitch can add thickness to sustained chords and help cover up naff loop points.)

Retrigger mode prevents new notes from starting at the beginning of the sample when playing legato.  In mono mode, you can use it to play legato and the new or retriggered note will pick up the playback at the point where the first one was.  In poly mode, it's a bit strange: If you set up the amp envelope with a long release time and turn retrigger mode off, and then play and hold one note while playing a second note, the second note will began playing at whatever point the first note is at in its playback when you struck the second note.  Of course, if you are in resample pitch mode, they won't stay together because one will play back faster than the other.  

To the right of the word "CONTROL" are mute and solo buttons.  These work similarly to on a mixer. If you hit "Mute", the current layer is muted.  If you hit "Solo", every layer other than the current layer is muted.  A muted layer is indicated by its layer button turning red.  

The modulation routing controls at the bottom right are not specific to the layer; they are global.  We'll talk about modulation later.


There are three basic views available in Iris.  The layer view we've already reviewed above; it allows you to see a single layer at a time and perform spectral editing.  There are two other views available by clicking their buttons at the upper right of the window: the All view and the Mix view.

The "All" view 
The All view is similar to the individual layer view, but it displays small versions of all four layer samples in four panes in the sample editing area.  You can scroll these up and down using the small buttons to the right of the panes, or grab and drag the frequency rulers.  You can actually do spectral editing in these panes, although it's awkward due to the small size.  (Recall what we said a while ago about the painting tools not scaling with the zoom level.)  This view contains the same patch parameter settings on the right side as the individual layer views.  You choose which layer they effect by clicking in its pane in the sample viewing area.

The Mix view
The Mix view shows all of the patch parameters for all four layers, plus the global patch parameters and effects settings.  The sample view area shows only an overview of the sample with none of the spectral editing, and you cannot edit here.  However, you can change all of the patch parameters.  The area on the right contains the filter controls and the global LFO settings, plus the effects mode setting and the MIDI Learn button, which are only accessible from this view.

The filter effects the overall mix, including the outputs of the effects.  There are 11 algorithms; three each of low pass, bandpass, and high pass, plus a peak booster (a sort of very narrow bandpass).  Most of them sound like they are intended to emulate familiar analog filter circuits.  Cutoff and resonance are adjustable on all, plus there is a dedicated ADSR envelope that can modulate cutoff.


Each layer has an LFO that can be routed to pitch, amplitude, or pan for that layer.  These are the only parameters that are modulatable on a per-layer basis.  Seven waveforms are available, and the onset of the LFO can be delayed using the "attack" parameter.  The rate can be synchronized to tempo.  When "Restart" is on, the waveform starts at zero for each note played; when it is off, the LFO is free running and all voices will have their LFOs synchronized in phase.  There is also a global LFO which can be assigned to global amplitude, global pan, or the filter cutoff frequency.  The global LFO's controls are in the Mix view.  

The "Mod Routing" controls at the bottom right control global routing of velocity and aftertouch (they cannot be routed on an individual layer basis).  Clicking on the arrow symbol next to "Mod Routing" brings up a pop-up window that allows velocity and aftertouch to be routed in varying positive or negative amounts to amplitude, depth of the global LFO, filter cutoff, and filter resonance.  Note that all of these are global parameters that apply to the whole patch.

If this all seems a bit limited, it is.  The saving grace is that almost every parameter that is visible in the Mix window can be assigned to a MIDI continuous controller.  In the Mix window, click on the "MIDI Learn" button at the upper right, and all of the assignable parameters turn blue.  Click on the parameter that you want to assign.  Then, send a controller message by moving the controller of your choice (or programming it into your DAW), and Iris will associate that controller # with that parameter.  

Mix view in MIDI Learn mode


There are four effects: distortion, chorus, delay, and reverb.  Only one instance of each is available.  The effects can be configured in either of two modes: send effects, in which each layer can be routed to one or more effects, and master effects, in which the combined mix is routed through the four effects in series.  In order to change the mode, you must select the Mix view.  Then at the top right, to the right of  the word "MASTER", there is a box labeled "Effects Mode" with icons for the two modes.  Click on the one you want.

In Send mode, the Send Effects knobs on each layer allow you to send some amount of that layer to the effects; they work like effects sends on a mixing console.  A mix of one or more layers can be sent to each effect.  The effects outputs are mixed with the dry outputs of the layers to form the sum output.  A slight annoyance is that before you can use an effect, you must click on the power-button icon to activate the effect.  A convenience is that, from the layer views, you can click on the arrow icon to the upper right of the knob to bring up a pop-up window that allows you to adjust the effects parameters without having to go to the Mix view.  

In Master mode, the four effects are in series: the distortion effect receives the summed dry mix of the four layers, and then the signal flow proceeds from left to right as shown in the mix view.  In this mode, the Master Effects knobs, as shown in both the layer views and the mix view, control the dry/wet balance for each effect; if the knob is at zero, the signal will effectively bypass the effect.  There is no way to control the balance of individual layers in the effects send when using Master mode.

The quality of the effects is decent.  I actually found the distortion effect to be the most versatile.  It allows the choice of five different distortions, ranging from the fairly mellow "Tube" to the absolutely brutal "Asymmetrical", plus an aliasing effect that did particularly interesting things to higher frequencies.  The chorus is basic but functional and pretty effective for adding some depth to some of the flatter-sounding patches.  The delay is also basic but it does what it is supposed to do, and it has a max delay time of 1.5 seconds on each stereo channel.  The reverb was the only effect I didn't care for; I found it cloying and limited, and it tended to obscure the more subtle aspects of patches.  (As you are going through the factory patches, try listening to some of them with the reverb off.)  

Macro Knobs

At the lower right of the screen, in every view except the Mix view, next to the modulation routing knobs is a mechanism called the macro knobs.  This allows you to assign a number of parameters which can all be changed by turning a single knob.  Minimum and maximum ranges can be assigned on a per-parameter basis, and the range can be inverted so that a given parameter can be made to decrease as the knob is turned up, or vice versa.  To access the knobs, you click on the pop-up icon next to the word "Macro".

The macro facility comes with some templates, called "styles", with pre-assigned parameters, or you can create your own using the parameter's pop-up menu.  The way you do this is: locate the parameter you want to assign, then bring up the parameter's pop-up menu (right-click on Windows, control-click on OSX), and using the "Assign" menu.  There are eight macro knobs available (which you can name, on a per-patch basis).  Right-clicking or control-clicking on a macro knob brings up a pop-up menu, in which you can change the knob's name, look at the parameters assigned to it, or clear the existing assignments.  (If you want to remove just one assignment, or update the range on an existing assignment, you need to go back to that parameter and access its pop-up menu.

The X-Y pad doesn't seem to work quite the way that the instructions describe.  It seems to describe the  two dimensions of the pad cursor movement being two additional macro knobs, but what I'm seeing is that they are hard coded to the #1 and #2 knobs -- moving the pad cursor horizontally causes the macro knob #1 to increase and decrease, and the vertical does the same to knob #2.  The only thing about the pad that I see you can change is the four legends; right-click or control-click on the pad to bring up a pop-up that will let you rename them.

Izotope describes the macro knobs as being a way to quickly achieve a certain style of sound, by choosing the appropriate template, once you have chosen a patch.  I'm not actually sure that's the best use for them.  Yes, it may be handy sometimes for that purpose, but I see a lot more potential in using them for on-the-fly patch morphing and automation / MIDI control weirdness.  The key to this: You can assign a macro knob to a MIDI controller using the MIDI learn mode.  To do that, go to the Mix view, and then find the cleverly concealed pop-up icon for the macro knobs -- it's in a different place in this view, to the lower right of the Master knob.  After you've done that, you can click on the "MIDI Learn" button, and then you can assign a MIDI controller to a macro knob the same way as for the other parameters.


Iris is a pretty cool piece of software.  This is the sort of thing for which soft synths really stand out: a synthesis method that would be all but impractical to implement in dedicated hardware, because of the cost of designing it and the need for a screen to do the editing.  I've done a lot of playing with it and I feel like I've only scratched the surface; it's a totally new method of synthesis to me, and I have a lot to learn yet.  The edit rendering seems to work well, without any aliasing or Fourier analysis artifacts that I've noticed.  The effects, as noted above, are good with the exception of the reverb which I'm not a big fan of.  I do wish there was more flexibility in the effects routing.

I only have a couple of criticisms of my own.  My main one is that I don't understand why, in this day and age and in this context, the envelopes are restricted to ADSR; the standard for advanced soft synths these days is multi-segment envelopes, and Iris is one synth that could particularly benefit from them.  The envelopes would also benefit from both a start delay and a time limit on the sustain phase, which would allow samples to be layered in and out on sustained chords and drones.

I would like to see more drawing tools offered for the spectral editing.  Circular and elliptical shapes could be useful, as well as the ability to draw arbitrary polygons by point-to-point clicking.  Similarly, the ability to erase by some means other than just freehand is desirable; I'd like to see an "anti-paint" function which works like region painting, but has the effecting of erasing any painted areas that are anti-painted over, and is capable of performing any draw operation that painting can.

A lot of the commenters at KVR would like to see a time stretch function that can be applied to samples, so that loops can be set up to exact lengths.  I haven't found a pressing need for that, though; if I need a sample time stretched, I can export it to some other software to do that job.  However, the ability to set loop start and end points to exact times would be helpful.  I'd also like to see the ability to set up a separate release loop, like some samplers have.

Another common request at KVR is to be able to set a volume level on a painted region, so that some choice in between full off (not painted) and full on (painted) is available for a region.  Not sure what I think about that... I think that if there were multi-segment envelopes, it would go a long way towards fulfilling the same purpose.

I will say that, running Iris as an AU plug-in under Metro [version] on OSX [version], I have found the software to be completely stable.  I have encountered no crashes, no functions that ceased working after a while, and no audio or visual glitches.  Iris integrates well with this uncommon DAW software, so the software engineers must have done their homework regarding the AU specification.  I have also run the stand-alone version and encountered no problems.  I have not tried the VST installation -- although Metro accepts both, when I have a choice, I go with the AU version unless it doesn't work for some reason. 

Sound Samples

All of the below are uncompressed WAV files, so they may take a few seconds to load.
  • Factory patch:  Black Galaxy.  Demonstrates Iris' capability for creating slowly evolving sounds.
  • Factory patch: Toddler Squarepusher.  Chosen mainly because I liked the name, and it does fit.
  • My patch: Zombie Alarm.  Shows the ability to do rhythmic repeating sounds.
  • My patch: Coffee Jackhammer.  Just plain screwy.