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 modularsynthesis.com

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 modularsynthesis.com

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 MOS-LAB.com

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 modularsynthesis.com

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: Synthesizers.com 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 Synthesizers.com Q119 (bottom).  Top photo courtesy of David Brown at modularsynthesis.com; bottom photo courtesy of Synthesizers.com

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; Synthesizers.com 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. 

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.

Editing

(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.

Views

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.

Modulations

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

Effects

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.

Conclusions

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. 

The OTA, and why it's important

Many (in fact, most) analog synthesizers rely heavily on an integrated circuit called the operational transconductance amplifier, or OTA for short. The OTA is a variation on the common operational amplifier, or op-amp. From an electrical engineer's point of view, the main difference between an op-amp and an OTA is that, while an op-amp outputs a voltage that is proportional to the difference in voltage between its two inputs, the OTA outputs a current that is proportional to the difference in voltage between its two inputs.

This, however, is not what makes an OTA valuable from a synth designer's perspective. Analog synths rely on two behaviors in which the OTA differs from the op-amp:

1. The OTA has a third input, called the amplifier bias current or Iabc for short. (The letter I is the standard electrical engineering symbol for current.) With the conventional op-amp, the gain is determined by an external feedback resistor. The OTA's gain is set by the amount of current that is allowed to flow into the Iabc input.

2. The OTA's input impedence is proportional to the current flowed into the Iabc input.

What does this mean, in practical terms? Consider the first: It's a simple matter to convert a voltage to a current, or vice versa. So if you take an external voltage input, convert it to a current, and feed that to the Iabc input to control the OTA's gain -- well, that's what a VCA is. Another way of looking at it is that the OTA is an analog multiplier; the difference between the two input signals is multiplied by the Iabc input. That opens up all kinds of possibilities, such as ring modulators and wave shaping circuits.

As for the second characteristic, if a voltage is converted to make the Iabc input, then the OTA becomes, in effect, a voltage-controlled resistor. As it happens, a voltage-controlled resistor is a very handy thing to have when designing VCF circuits.

Now the problem: Other than analog synth circuits, there are few uses for OTAs these days. There was a time when OTAs were widely used in radio and microwave systems, but digital signal processing has taken over in these applications. Consequently, OTA integrated circuits are disappearing from the market. A survey of the casualties:

• Intersil CA3080: Originally designed by RCA and introduced in 1969. A good, basic OTA cicruit widely used in both commericial and DIY designs over the years. Because it was produced in huge numbers, there's some stock still around and it's still possible to find them. But of course, that stock is a finite number and it will run out eventually. (Rochester Electronics still has some, although they aren't cheap.) 


• Intersil CA3280: Long regarded as the Cadillac of OTA designs (and often priced accordingly), this was also widely used particularly in top-end VCA designs. Blogger Don Tillman made a heroic effort to rescue the 3280, but was unsuccessful. Because of the higher price, it was probably not produced in as large a quantity as the 3080, and existing stocks have nearly dried up.


• Intersil CA3060 and 3094: These were variants of the 3080. I believe the 3094 was a dual OTA with buffered outputs, and the 3060 was a triple OTA. They are both long out of production and there are no sources that I know of.


• National LM13600: I think this part was used mainly in DIY designs. It can still be found, but except for repair of existing circuits, it isn't as widely sought out because it's not the easiest part to incorporate in a circuit, and it's specs aren't as good. The still-in-production LM13700 is better in both respects.  (Internet rumor has it that the original design was undertaken by an intern, as an exercise.) 


• Rohm BA662: Heavily used in '70s and early '80s Roland gear. Long out of production. There is no source that I know of.


• Rohm BA6110: Widely promoted as an "almost pin compatible" substitute for the BA662 after the latter was discontinued. However, the 6110 is also now discontinued, and existing stocks have nearly dried up.

There are only two OTA ICs in production at this point, the above-mentioned National LM13700 and the On Semiconductor NE5517. The LM13700 is regarded as a decent design, but not as well spec'ed as the 3280 and not quite as versatile as the 3080. But by default, it is the part of choice for new designs. The NE5517 is a bit of a story; it was formerly manufactured by Phillips, but the fab in France that manufactured it burned down in 2003. On Semiconductor then purchased rights to the design and picked up manufacturing. I've seen some Web references that say that the On Semi parts aren't spec'ed as tightly as the former Phillips parts.

So where does all this leave us? We have no indication as to what National's future plans are, regarding how long the LM13700 will be in production. Probably for a while, since there is little competition now, but one never knows. There are people in the synth design world who know how to design OTA circuits from discrete components, and to name one example, the Synth Tech MOTM-440 uses just such a circuit. But it's not as DIY-friendly an approach; there are a lot of subtleties to designing a good OTA circuit, and it's more expensive and more trouble then using an IC. But as far as technological progress goes, it's definitely a step backwards.

Miscellaneous thoughts for August

Some things I've taken note of recently:

Buchla Music Easel back in production

After about 40 years, the Buchla Music Easel is back in production.  The original Music Easel was produced in the mid-1970s; it basically consisted of a semi-modular, two-VCO analog monosynth coupled to a 2-1/2 octave, capacitive touch sense keyboard with variable tuning and scaling. 

Both the new Easel, and most copies of the original, took nearly all of the internal circuits out to a big edge connector called the "program card" connector.  Back in the day, Buchla sold cards that plugged into these connectors that the (soldering-iron-equipped) user could use to program patches.  This was done by soldering resistors onto the cards to make certain connections.  Unknown to most people, though, the program card connector contained additional interconnects that made it possible to actually play the Easel remotely.  Buchla at the time was doing some experiments with computer control, e.g., the Model 700 that connected to a DEC PDP-11 minicomputer.  However, no interface for the Easel ever reached production.

Now, 40 years later, it has.  Buchla has introduced the iProgram interface card, which connects via wireless Ethernet to an iPad.  Using software supplied by Buchla, the iProgram can function as both a patch editor/librarian, and a MIDI interface.  That's not a bad trick, and it says something about the Easel's original design that it can be interfaced to something that would not be developed for another 40 years. 

The Return of Richard D. James

The first album of all-new Aphex Twin material since 2001 is nigh.  Syro, announced by means of this bizarre press release (via Synthtopia), will be available in September.  Oddly enough, I didn't realize it had been that long.

Modulation Wheels

I've been doing a lot of patching with the Solaris recently.  Now, I "grew up" as a synth performer using mostly Roland keyboards (that is, I've owned lots of synth, but the ones of other makes have been mostly rackmount units.  Of the ones I've had with actual keyboards, most have been Rolands.)  So I've long since grown accustomed to the Roland "T-handle" control for pitch bend and modulation, including the fact that it's spring loaded in both the pitch and modulation axes. 

The Solaris, on the other hand, has traditional pitch and modulation wheels, and the mod wheel is not sprung.  I've often found this vaguely disconcerting.  However, last week, I was working with a three-oscillator bass patch I had built, and I wanted to assign a performance control to detune one of the oscillators.  Assigning it to aftertouch did not do what I wanted.  I was going to use a soft knob, and then it occurred to me -- assign it to the mod wheel.  By doing this, I was able to put it in various positions to achieve the amount of detne that I wanted at different times.  This really led me to appreciate the virtues of a mod wheel that stays where you put it.

Stuff I Want

I've got my eye on Madrona Labs' Kaivo plug-in.  I've long wanted a physical modeling synth that provides the performer with access to the parameters inside the guts of the modeling algorithms.  Kaivo appears to be all that and a lot more.  I'm also looking for something that will give me a much larger variety of tuned and semi-tuned percussion sounds that one will get from the typical drum machine, and it looks like AAS' Chromaphone is just the thing. 

I've also gotten interested in some of the effects boxes from Strymon, particularly the Timeline delay, which has MIDI control.  I can think of all kinds of possibilities for that.  I've read that Strymon includes a lot of ex-Alesis employees, from the time when Alesis was doing good stuff (before Keith Barr lost control of it), so that's a good sign.

Next Week

I have a review of Izotope's Iris spectral editor, which I'll try to have up by this weekend.

Reviving M1000X

I've finally found the time to get Xcode going on my Mac Mini and rebuild M1000X, the program I published some years ago for doing patch editing on the Oberheim Matrix-6/6R and Matrix-1000.  It now runs on Intel Macs under OSX 10.8 (it should also work under 10.9, but I have not tried it).  There were some challenges involved in getting it going again.  (OSX coding geekdom ahead...)  I originally wrote M1000X in 2005 in pure C, using the Carbon libraries.  Apple in its infinite wisdom has since deprecated Carbon, and is pressing hard to get all remaining Carbon apps converted to use the Cocoa libraries.  Unfortunately, making a Carbon app written in C work with Cocoa involves a significant re-write, object-izing the code and reworking it into Objective-C.  Additionally, the current version of the interface builder built into Xcode refuses to open the Carbon nib files, although the runtime still works with them.  Fortunately, I still have my old PowerPC Mac up and running, so I was able to do the interface resource editing that I needed to do there and copy the nib file over to the Intel-based machine.  Additionally, due to a problem with the way that bit fields in structures are specified in C (a problem that the language has had since the original K&R specification in the early '80s), I had to re-write some large structures that handle patches in memory.  While I was at it, I took the opportunity to fix a couple of bugs, and to put code in to work around two bugs in the Matrix-1000 OS that I didn't become aware of until after I had released version 1.0.

Anyway, version 1.1 of M1000X is available here.  Just open the disk image file and drag the application to your Applications folder.  There is a help file in there and a set of release notes also.

So while we're here, this is a good place to talk about the Oberheim Matrix series of synths.  These consist of two groups -- the Matrix-12 series, and the Matrix-6 series.  The Matrix-12 series consists of two synths, the Matrix-12 and the Xpander, which basically is a "tabletop" Matrix-12 with no keyboard and half as many voices.  Both of these synths are now very expensive collectors' items, and they also don't have a lot in common with the Matrix-6 series, so I won't dwell on them further here.

The Matrix-6 series consists of the Matrix-6 itself, a 6-voice keyboard; the Matrix-6R, a rack-mount (2U) packaging of the Matrix-6, and the Matrix-1000, with the same voice architecture but more patch memory and some improvements, and packaged in a smaller (1U) rack-mount box.  All of these synths are based on the CEM 3396 "synth on a chip", which provides an all-analog signal path consisting of two DCOs, one low-pass VCF, one noise generator, and two VCAs.  The VCF can be FM'ed by one of the DCOs.  The synth adds to this a bevy of modulation and signal sources: two LFOs, three DADSR (delay-attack-decay-sustain-release) envelope generators, two ramp generators, and a tracking generator. The patch memory consists of 100 patches on the 6/6R.  The 1000 has 800 patches in ROM and 200 in RAM (hence the name).

The "matrix modulation" capability is what gives the synths their name.  A radically different idea at the time it was introduced, the modulation matrix is basically a virtual version of the old EMS pin matrix (which is displayed graphically by M1000X).  There are ten "pegs" that you can insert at any intersection in the matrix, and you can set the level (inverted or non-inverted) of signal to be routed from the selected input to the selected output.  All of the internal modulation generators are available as sources, as are external expression input jacks and MIDI continuous controllers.  Almost any parameter of the DCOs and VCF can be a matrix destination, as well as most of the envelope segments, the LFO rates and amplitudes, and the portamento rate.  It really did at the time help break out of some of the limitations of its contemporaries, which at the time usually had very limited signal routing -- for instance, on many synths there was one LFO that could only be routed to oscillator frequency, and maybe one other one that could only be routed to VCF cutoff or VCA level.

While the voice architecture is quite capable, the user interfaces for patch editing are limited.  The Matrix-6 and 6R both use the same mechanism, a membrane panel with three mode select buttons, a numeric keypad, and some forward and back buttons.  Visual output is by way of a one-line vacuum fluorescent display with 16-segment characters; it's bright and easy to read (except for some punctuation characters that come out looking funky in the 16-segment format), but it can only display 20 characters.  The display basically switches from display of patch numbers and names to displaying names and values of patch parameters, and patch editing is quite tedious.  As for the Matrix-1000... well, it has no patch editing controls at all.  Clearly there was a cost and packaging decision at some point (apparently Oberheim really wanted to get the Matrix-1000 crammed down to a 1U package).  The only thing you can do from the panel is select patches, set up MIDI, and change a few system parameters.  Output is limited to a 3-digit numeric LED display and a few status lights.  This is why you need an external editor. 

Yes, these are analog synths.  They do use DCOs, and some (most?) of the modulation signal generating is digital, but the audio signal path is all analog.  Oberheim produced many of these synths, and because of the limited user interface they are regarded as less desirable than many other analog synths, so they can be had at reasonable prices.  If you're looking to get into vintage analog inexpensively, this is a good way to do it -- get one of these synths, and then use M1000X to overcome the UI limitations.