Sequence 15

Realizing the Realizer

2012-01-08T15:07:00.001-06:00

This post was inspired by Big City Music's announcement this week that they are offering a PPG Realizer for sale. The Realizer, as many synth enthusiasts know, was the legendary attempt by Wolfgang Palm's PPG company to pioneer the soft synth concept using 1980s technology. However, the development costs became the final straw for the financially struggling PPG, and led to the company liquidating itself in 1988. The Realizer never went into production; rumor has it that two prototypes were built. (If this is true, I don't know where the other one is; I've checked the listings of the Audities Foundation, the New England Synthesizer Museum, and the Eboard Museum. None of them list a Realizer, but that doesn't mean they don't have one in their collection somewhere.

Realizer control desk -- from ThePPGs.com

So what was the Realizer actually? Modern legend has it that it was the first virtual analog synthesizer. Actually, however, it was both more than that, and possibly less than that. According to Palm, the Realizer was an early attempt to build what we now call a "workstation"; it would have been capable of synthesis, multi-track recording, processing, and mixing. The unit you usually see in pictures, which PPG called the "control desk", is not the whole system; it's only the control unit and user interface. The control desk could interface with a combination of sound modules that actually did the audio processing, and hard disk units (HDUs) which provided audio and data storage, up to 8 units total. The sound modules and HDUs were intended to be somewhat mix-and-match with other PPG products, such as the Waveterm B.

The sound module was the piece of the Realizer that did most of the work. Like many of today's digital synths, it contained a bank of digital signal processor ICs -- in this case, eight of the TMS 32010 -- and a Motorola 68010 that managed everything. (The HDUs also contained a pair of 32010s, but it appears that they were not used in the Realizer configuration.) PPG developed three synthesis packages that ran on the sound modules. The one that everyone talks about in regard to the Realizer was the "Minimoog" virtual analog software that reproduced not only the features of the Mini, but also its panel layout. However, there was also application software for additive synthesis and for the sampling and wave scanning method of synthesis that PPG was know for. Further, according to Palm, the software allowed individual processing functions to be mixed and combined in the style of a modular synth. (You could say that PPG invented Reaktor in 1986!)

PPG HDU with stand-alone control unit (in foreground). From Synthmuseum.com

Further, the Realizer also incorporated the functions of what we now call a digital audio workstation. It was capable of multi-track recording and editing using the HDUs as storage. It had "plug-ins" for adding common studio effects (although, according to Palm, the reverb software was troublesome and was never completed to a satisfactory degree). And it could do mixdowns, producing a digital master that could presumably be transferred directly to CD mastering systems, although it is not clear how that would have actually worked.

The control desk is the part of the Realizer that everyone has seen. It contains a monitor (which was to have been color in the production version, but the prototypes had green-phosphor monochrome), 31 knobs, 6 faders, a data entry wheel, a keypad with numeric and function keys, and a graphics tablet. Each of the knobs and faders had line graphics drawn on the panel from the control to the edge of the screen opening; the software generated lines on the screen that led from the screen edge to the parameters on the screen, and by that method, the user got a visual indication of which knob or fader controlled what on a given screen. The graphics pen would have been used for drawing, and probably also for selecting commands from menus, in the fashion of the Waveterm. The best photo I've been able to find that shows a screen and illustrates the layout and the knob graphics is the following, taken from Palm's old blog on Myspace. Palm describes this photo as being a photo of an early mock-up, which accounts for the crude-looking panel.

So what made the Realizer so expensive to develop, enough that it killed the company? One thing that Palm mentions is that in the timeframe when the Realizer (and presumably the Waveterm B also) was being designed, PPG had decided that laying out the circuit boards by hand, which was how they had done all of their previous products, wasn't going to work given the density of the boards they wanted to design. So they purchased an electrical CAD software package (which would have been very expensive in 1986), and then leased a computer system to run it. Unfortunately, apparently the leased computer didn't have sufficient performance, and the system ran very slowly, harming productivity. There probably would also have been a learning curve for the engineers. Palm also mentions, in his account of the Waveterm B development, that the Waveterm B and Realizer were PPG's first products to use 16-bit sampling, and that they had a hard time getting their 16-bit analog-to-digital converters working properly. (In order to build the initial sample library to be shipped with the Waveterm B, they hacked a Sony F1 digital tape system and used its converters.)

However, I'm guessing that the real killer was the software. Even though there was probably some commonality with PPG's other products, there would have been a ton of new software to be written. They had to write a lot of new software for the control desk displays and user interfaces; the Moog emulation and the additive synthesis was new, and they had implemented a lot of improvements to the sampling which required new software. (Plus, knowing how things were done back in the day, the 12-bit sample handling software used in the Wave keyboards probably did a lot of "tricks" with unused data bits, which would have had to be re-written to handle 16-bit samples.) The sound modules required a bunch of new software to manage all of the DSPs, not to mention the actual DSP processing code. Also: they were writing all of this in assembly language. Palm states that a C compiler was not available for their systems at the time -- a statement which puzzles me, since C compilers were readily available for most processor families by 1986, and Sun, to name one, had one for the 68000-series CPUs.

The final factor was that, due to a combination of circumstances, PPG found themselves without a lot of money to spend on R&D. In 1986 they had invested money in moving production to a larger facility, in part due to robust sales of the Wave, but at about the time the new factory opened, Wave sales began to decline. The Wave, especially by the time of the 2.3 revision, had very sophisticated capabilities for wave scanning and manipulation of samples, but a lot of users didn't care about that -- they only wanted basic sampling and playback, or just playback of canned sample libraries, and so they gravitated towards less expensive samplers like the Emulator or the Ensoniq Mirage. The Wave was PPG's bread-and-butter product, so when sales declined, the company's revenues suffered. A few Waveterm B's were sold, and a few HDUs were sold as stand-alone products, but the Realizer never reached production before Palm and the other founders realized that they were going to run out of money. They liquidated rather than continue and be forced into bankruptcy.

Going back to the sale by Big City Music, I don't know exactly what it is that they are selling. The ad shows only the control desk. Although that item no doubt has significant collection value by itself, the point remains that if the goal of a buyer is to get the system actually running again, it won't do anything without at least one sound module and one HDU. Perhaps Big City has these items and is including them in the sale; the ad copy doesn't say.

Below is a link to a 1987 demo, from Palm's Myspace page. The Realizer's control desk can be seen on the left for much of the video. (The device that the demonstrator is holding appears to be a stand-alone control for an HDU, and not part of the Realizer configuration.) Note that the demonstrator never touches the Realizer control desk, which suggests that the software was still not stable at this point.

http://mediaservices.myspace.com/services/media/embed.aspx/m=,mr=60628615,t=1,mt=video

Finishing up with a historical curiosity: The astute observer may have noticed that in the photo of the control desk, following the first paragraph of this post, the desk does not have the data entry wheel. I don't know if this implies that prototypes were built both with and without the data wheel, or if it was added to the pictured unit after the photo was taken.

Thoughts on panel graphics

2012-01-01T22:30:00.001-06:00

Happy New Year to synthesists everywhere! Looking back at the last year, I see that for a while I've been focusing almost exclusively on gear. That was not really my intent. I'll let you in on a little secret: One of the reasons I created this blog was to use it, in a sense, as my own notebook; a lot of the things that I post are things that I want to be able to refer back to myself. For instance, since I did the detailed description of the MOTM-650 MIDI-to-CV interface back in 2010, I've referred back to it several times when I needed to sort out some parameter or other.

However, I'm not doing this just for myself. If I was, I'd just keep a notes file on my computer, and not bother with a blog. My New Year's resolution, in regard to Sequence 15, is to share thoughts of all sorts with regard to synthesis and electronic music. Just blogging about gear is too limiting, and accounts for the dearth of posts.

So here goes... In the ongoing discussion about the differences in the user community between modular synth performers who prefer the Eurorack format, and the performers who prefer the "5U" formats (MOTM, Dotcom, and Modcan-A), one thing that's often debated is the style of panel graphics commonly seen on Eurorack modules, versus the usual style of 5U modules. Let's compare: Here is a (rather murky) photo of my Encore Universal Event Generator:

By comparison, here, from Analogue Haven's Web site, is a photo of the Makenoise Maths:

The Encore UEG clearly maintains the tradition of the vintage Moog modulars: white graphics on a flat back background, and controls in neat rows and columns. (Further, it follows the MOTM format convention of putting the I/O jacks at the bottom, although it doesn't really conform to the whole MOTM standard grid due to the large number of knobs.) All the controls and jacks are labeled in a clean font, and the panel has index marks for the knobs. Line graphics are used to indicate associations between controls. Most MOTM, Dotcom, and Modcan-A format modules follow this pattern; in the world of 5U, Modcan's B-series modules (which are MOTM format) are considered a bit radical for having black graphics on a white background. There have been a few other makers of large-format panels who have used colored text and line graphics, but even they tend to stick to the black background and standard fonts.

Now let's compare with the Maths. White background with a red border around the edge of the panel. (And that's considered conservative in Euro-land.) Zig-zaggy graphics that show the flow of signal through the module. There's four input jacks; they are at the top of the module, and you have to read the manual to realize that they are the four jacks pointed at by the small arrows. Knobs and jacks scattered hither and yon, although the panel is symmetrical. (It has two processing channels; the two outside ones do basically the same thing, and the same goes for the two inside ones.) Functions of some of the jacks are indicated only by the signal flow graphics. You have to look rather closely to see the little math operator symbols that label some of the controls. The knobs don't have any indexing, and there are two illuminated pushbutton switches whose purpose is not indicated at all. And I don't know where the hell Makenoise came up with that font; maybe they made it themselves.

If that sounds like I'm ragging on Makenoise, I'm not intending to be. If you go to Makenoise's Web site and look through the descriptions of their modules, you realize that Makenoise has its way of doing things, and once you've studied it and gotten into that groove, most of those panel markings make intuitive sense. Where you start to run into problems in Euro-land is when you realize that the Makenoise way of doing things is not the same as the Harvestman way of doing things, which is not the same as the WMD way of doing things, etc.

Euro users put up with this, in part because it looks cool. But I think there is also more of an aesthetic in the Euro world of being more willing to patch something up, turn some knobs, and see what happens, where in the 5U world, users tend to want things to be more precise (or "anal" if you prefer). This is just a general statement based on anecdotal data; it certainly doesn't apply universally. However, I do note that there are a few small makers in the Euro world who are willing to silkscreen something on a panel that has nothing to do with a panel's function, or just leave a panel blank; almost no one in the 5U world would ever do that. I do note that even in Euro land there has been a bit of a reaction to some of the more excessive panel designs. Pittsburgh Modular makes a wry comment on it with their 1960s-embossed-label aesthetic.

Other aspects of small vs. large format have been discussed to death already: 5U takes up a lot more space; Euro/Frac knobs are too small for large fingers, 5U panels cost more to make, 3.5mm jacks break off too easily, etc. However, I think there's one other, very practical concern. It's been noted that 5U users tend to be, on average, older than Euro users. Here's the other reason us 5U guys like things nice and clean: when we look at something like the Maths panel above, we can't see the panel! Our eyes aren't as good as they used to be. If we had a Maths, we'd have to get a magnifying glass out every time we wanted to use it. Panels like that give us headaches. We have to stick with nice high-contrast panels with clean labeling that we can see.

And besides, we like the laboratory-test-equipment aesthetic. Our moms all say it reminds them of their father's ham radio gear.

Washington

2011-12-28T23:03:00.001-06:00

A new Statescape... named Washington, and done entirely with the Solaris. More at the Web page.

Solaris demo, and a few comments/corrections from John Bowen

2011-11-14T21:22:00.002-06:00

First of all, John Bowen has sent along some corrections and additional notes to my overview posts. (I warned everyone that it was a quickn-dirty...) John's notes:

The mixers allow each input to be modulated separately, plus the output can be modulated. So each mixer allows five different modulation inputs. I should have realized that; I guess I assumed that "ModSrc1:", "ModSrc2", etc., referred to the four mixers rather than the four inputs of the selected mixer.

The S/H waveform of the multimode oscillator is tunable and can track the keyboard; it isn't just a low pass filtered noise.

Samples cannot be loaded via the USB interface in the current version of the OS. To get new samples into the system, what you have to do is get a CF flash card reader connected to your computer (they're cheap), pull the CF card from the Solaris, put it in your reader, and then move the sample files to the card. Then, you plug the card back into your synth. I'll write some more about samples and sample loading after I've had a chance to experiment with it.

I forgot to edit something before I published it about signal routing to the FX channels, and as a result I got that part wrong. The four VCAs mix down into a fifth VCA, which you can't control directly, but VCA 6 is connected to it. When you select "synth" as the input to an FX channel, it is taking its input from the fifth VCA.

The overdrive part of the Minimoog filter algorithm was actually moved from the filter to the VCA section, which allows you to use overdrive with any filter. It's the VCA "boost" parameter (which I missed when I read that in the manual).

I mentioned some anomalous behavior in the velocity and aftertouch, which at the time I was writing that post, I had confirmed with MIDI Monitor. Well, guess what: the next day, it was behaving normally with the same patches. So I'm not sure how I did that. Possibly I messed up the sensitivity settings when I was playing with the system parameters (which, fortunately, I didn't save).

The envelope follower can be used with any signal source, not just the external inputs. I knew that, but the way I wrote it may have given the wrong impression.

I should have mentioned that the default routing of the mod wheel to the LFO 5 amount to pitch can be disabled. Then, LFO 5 is output at a constant amount set on the main page 2. You can always route mod wheel anywhere you want, just like any other modulation source.

I made a comment about the joystick on previous vector synths: John says that the Prophet VS and Korg Wavestation (neither of which I've ever had my hands on) were not capable of memorizing joystick movements. So I guess that's a characteristic that was limited, among vector synths, to the Yamaha SY77/TG33 (which John says he didn't work on). I have a TG33 and I know it does that.

Finally, John wrote me some good stuff about his history with Sequential Circuits and Creamware. He wants it to be known that he was not the author of Scope -- it was already written when he went to work for Creamware. John gave me some great info, and with his permission, I'll summarize it in a future post.

Now, finally, the Solaris demo. Note that this was also kind of a quick-n-dirty; there is some parameter tweaking, but it's all based on factory patches. There are four parts: in part 1, I demonstrate a patch that uses a rotor, and demonstrate some of the effects you can get by varying the rotor frequency. There's a quick demo of using the main screen and patch list to select patches, and then part 2 which demonstrates the different filter types. Part 3 is a quick demo of a patch that uses the arpeggiator, and part 4 demonstrates the ribbon controller.

Solaris Architecture, part 2: Control Sources

2011-10-31T21:52:00.001-05:00

In Part 1, we looked at the audio signal generators, processors, and routing. In this part, we'll look at the control signal sources. To recap, these are:

Six envelope generators
One looping envelope generator
Four lag processors
Arpeggiator
Step sequencer
One envelope follower
Velocity and aftertouch
Performance devices: Joystick, ribbon controller, assignable buttons, and assignable knobs
Expression pedal input jack
MIDI continuous controllers

The six envelope generators are of the DADSR (delay-attack-decay-sustain-release) type. EG 6 is hardwired to the final VCA, although any of the VCAs can be routed to other destinations. The minimum time for any segment is 0.1 milliseconds and the maximum is 20 seconds. A useful feature is the ability to slope the sustain segment, instead of it having it be a constant level. The time of each segment can be modulated by velocity, key tracking, mod wheel, or one of four MIDI continuous controllers. The looping envelope is actually a universal event generator; in fact it is quite similar to the Encore Electronics UEG that I reviewed last year.

The lag processors do what you expect; you can route other signals through them to smooth transitions. (You don't have to use the lag processors to produce key portamento; each oscillator has its own glide settings.) The envelope follower produces a control signal proportional to a signal fed to it (presumably from an external input).

The Solaris is unusual in that it contains both an arpeggiator and a step sequencer. I have not played with either one very much yet. The arpeggiator can use internal clock or sync to MIDI clock. It does the usual up, down, up/down, random, and as-played patterns. The sequencer is a four-row, 16-step sequencer with variable step lengths. and it can be routed to any destination -- it isn't tied to oscillator frequency. Both have the ability to use stored patterns that the user can create, but the software to edit the patterns is not finished yet.

The keyboard generates velocity, release velocity, and aftertouch. There are scaling and offset parameters which can be stored in each patch, which apparently are capable of making velocity and aftertouch do some rather strange things. According to the manual, the Solaris will receive and respond to polyphonic aftertouch, although it will not generate it. The keyboard has the usual pitch and modulation wheels to the left; the pitch wheel is spring loaded while the mod wheel is not. The pitch wheel does not appear to be routeable to any parameter other than oscillator and rotor pitch. The mod wheel is defaulted to control the amount of LFO 5 that is routed to oscillator and rotor pitch, but it can be routed to other destinations.

The Solaris has an array of performance controls besides the pitch and mod wheels, the most notable of which is the ribbon controller that runs the span of the keyboard. In my tests with it, I found the ribbon controller to be very smooth and glitch-free. It can be configured so that the point where you first touch it becomes the zero point, in the style of the much-vaunted ribbon controller on the CS80. It can also be configured to hold its last value.

There are buttons to turn the arpeggiator and sequencer on/off, and a "hold" button that does what a sustain pedal does, except that it is latching; once you press the button, you can take your hands off the keyboard and it will keep playing. There are two assignable buttons that can send a constant value to a modulation destination. One of the preset pages on the large window allows the bottom row of knobs underneath the window to be used as assignable knobs. And there is a jack for plugging an expression pedal.

In the MIDI Setup, there are five controller designations labeled CC1-CC5. You can assign any MIDI continuous controller number to these, and then they can be routed to any modulation destination. Finally, it appears that every patch parameter is accessible via the MIDI NPRN mechanism, although it is possible that not all will respond in real time -- it would take a long time to try every possible value.

This wraps up the quick overview of the Solaris architecture. It's quite possible that I will later find out that some of what I've written is wrong; I'll make corrections in future posts. I'm getting a lot of requests for a demo, so I'll get that up in a day or two.

Solaris architecture, part 1: Signal sources and routing

2011-10-29T17:12:00.000-05:00

Here's a quick look at the Solaris architecture. At this point, I'm still studying the manual and experimenting with the synth, so my understanding is incomplete and some of what I say here may be subject to correction later. So be aware of that. Nonetheless, here goes.

At first glance, the voice architecture of the Solaris appears to be a basic four-layer setup with conventional oscillator-filter-amplifier chains. However, the routing is far more flexible than that: components can be swapped back and forth between layers, components can appear in more than one layer at a time, and feedback loops between layers are possible. Strange as it may sound, the best way to understand the voice architecture is to start in the middle, with the mixers.

There are four mixers, each of which has four signal inputs and two control inputs. The mixers serve the purpose of combining up to four signal inputs, and also act as VCAs under the control of the two control inputs. On the input side of the mixers are all of the signal sources:

Four oscillators
One white noise source
One pink noise source
Two vector processors
Two rotors
Two AM processors
External inputs
Four mixer outputs
Four VCA outputs
Five LFOs
Six envelope generators
Four lag processors
Arpeggiator
Step sequencer
One envelope follower
Velocity and aftertouch
Performance devices: Joystick, ribbon controller, assignable buttons, and assignable knobs
Expression pedal input jack
MIDI continuous controllers

A few things to note here. The first is that mixers can process audio signals, control signals, or any combination. Mixer outputs can in fact be routed as control signals back to various places. The second is that feedback is possible: a mixer's output can be routed back to itself, or to other mixers.

On the output side of the mixers are these signal destinations:

Four filters
Four insert effects slots
Four VCAs

The possible routings on this side of the mixer are: An insert effect can be before or after a filter, and either the filter or the insert effect can be the input to the VCA.

Audio Signal Sources

Let's look at the input sources. The mixers can accept either audio or low frequency sources, but I'll do the audio sources first. The original source of all audio signals within the synth (other than external inputs) is the four oscillators. You can choose from six different implementations for each oscillator. Two of them are analog emulations, a Minimoog and a Curtis VCO (I presume this means the CEM 3340); they offer the same waveforms as the originals. The multimode "MM1" offers all of the standard synth waveforms, plus some continuously morphable waves a la the EML 1o1, proper white noise (computed, not a sample playback), a low-frequency rumbling noise referred to as "S/H", and a supersaw that appears in the display as "Jaws". There's a set of single-cycle waveforms taken from the Prophet VS, a set of wavetables (which can be scanned using the "shape" parameter) from the Waldorf Microwave, and a sample playback mode into which you can load your own samples via the synth's USB interface. An oscillator can accept up to four modulation sources (which can be low or audio frequency, and includes pretty much every signal source in the synth), and each modulation source can be routed to modulate frequency, shape (the effect of which depends on the mode and waveform selected), or linear FM.

The white and pink noise sources do what you expect. Note that they are both computed rather than sampled, which means that they sound the same no matter which note you play, and there is no clocking noise.

The vector processors emulate the vector synthesis method used on the Prophet VS, Korg Wavestation and Yamaha SY77. The vector is basically a four-way mixer, with two sources at each end of an X axis and two more at the ends of a Y axis. (What they don't have is the ability to memorize a manual joystick movement and store it with the patch, which the synths named above do have. However, you could do this via an external sequencer.) By default, the vector inputs are tied to the panel joystick, but you can route any modulation parameter to either axis. The rotors are a variation on the vector synthesis idea: imagine a vector synthesis machine with a motor tied to the joystick, capable of making it move in a circle at audio rates. That's what the rotors do, crossfade between the four sources in a circular pattern. This amounts to a form of audio-rate wave scanning. The rotors track the keyboard (or not, if you switch it off), and otherwise act like oscillators.

The AM processors implement several possible amplitude modulation techniques, with one of the choices being "standard" AM, and another choice being ring modulation. I haven't played with this much yet and I don't yet understand all of the algorithms or parameters. Any signal sources can be selected as the carrier or modulation.

There are six external inputs -- four analog input jacks, and the left and right channels of the SPDIF input. They can be routed to audio and control destinations the same way that internal signal sources can. The mixer outputs can also be routed back to the mixer inputs. The Solaris makes no attempt to prevent feedback loops from being created, and in fact feedback loops can be used in patches.

Filters and Effects

The mixers output to a chain that consists of "enable part" switches, insert effects, filters, VCAs, and effects channels. Signal routing on this side of the mixers is more limited; any signal can be routed to the input of a filter, but the insert FX can only accept input from the mixer or filter of the same number (e.g., insert FX 1 can only accept input from mixer 1 or filter 1); the VCA can only accept input from the insert FX or the filter of the same number, and an FX channel can only accept input from the VCA of the same number, or from another FX channel or external input. (It does appear that clever use of the external outputs and inputs could get around some of these limitations, but I haven't tried that yet.) The "enable part" buttons, when turned off, cut off the output of the corresponding mixer to whatever comes after it, but they do not cut off the mixer from places where it has been routed to a modulation input.

The filters and insert effects come first, and can be placed in either order by means of selecting their respective inputs. (They can also be fed back to the mixer inputs.) The filters, like the oscillators, have several selectable implementations. The "MM1" multimode filter is the most versatile; it is a 4-pole filter that allows a number of pole combinations of low pass, high pass, bandpass, and band reject, in the style of the Oberheim Xpander. The "SSM" type emulates the 4-pole, SSM low pass filter as was implemented in the Rev 2 Prophet-5. The "Mini" type emulates the Moog 4-pole low pass transistor ladder filter, including its distortion and overdrive characteristics. "Obie" emulates the 2-pole filter as was implemented in the Oberheim SEM and other early Oberheim models; it is switchable to low pass, high pass, band pass, or band reject modes. The "Comb" fitler generates a comb-filter response as produced by a flanger or cardboard-tube echo; there are two variants. The "Vocal" filter produces vowel formants, and can be varied between vowel sounds. The insert FX are all waveform modification effects. The "Decimator" reduces the sample rate of the signal; I haven't tried it yet. The "BitChop" is a bit crusher, and there's a soft clipping distortion.

The next bit, I'm a bit confused about. The four VCAs each accept signal input from the corresponding filter or insert FX. There are two modulation inputs, one for level and one for pan, or they can be cut off which leaves the VCA wide open. The reason this doesn't result in an infinite sustain is that, apparently, all of the VCAs sum down to a fifth VCA which is hard-wired to envelope generator 6, and it provides the master control over the output. The effects channels then accept input from the master VCA, or from another FX channel or external input. The reason I say I'm not clear about this is because it's not quite what the manual shows, but I think it's correct, and you'll see why in a moment.

Each FX channel has four FX slots, each of which can hold one effect. The four available effects are the chorus/flanger, phaser, delay, and EQ. (They are all stereo.) The effects are "pooled" such that a given effect can only appear in one slot, in one FX channel, in a patch. Contrary to the manual, these appear to be after the master VCA, and here's why I say that: I played with the delay parameters and found that the maximum delay time is a whopping 20 seconds. And... if you set a long delay and then play some notes, the delay will continue to sound until the echoes die out, long after the master VCA has shut off. Statescape time! (And in fact, I'm already thinking about doing that... have to build a clever patch for it...)

Finally, there are five pairs of stereo output channels, four analog pairs and one SPDIF pair. Each FX channel can be routed to one pair, or the "dry" output of the master VCA can be routed out. This, for example, would let you send a dry output to an external effect, and a chorused output directly to your mixer or DAW.

Finally: I have found what appear to be a few bugs (not unexpected since the OS is version 1.0). I managed to crash the Solaris by twiddling the knobs under the large screen while I was on the second patch-store page (the one where you name the patch). Don't do that. Also, the keyboard velocity does not seem to work, and the aftertouch is odd -- it only outputs values of 0 or 63.

There's a lot more to the architecture; I haven't touched on the modulation sources yet. More in part 2.

Lookee what I found!

2011-10-25T22:31:00.000-05:00

Hmm, I wonder what's in this box...

It has a label on it printed in German. Could it be...

It is... it's a... SOLARIS!

The day that's been long in coming is finally here! To recap: John Bowen was a cohort of Dave Smith's at Sequential Circuits; it's not clear to me who did what at SCI, but I understand that John had a lot to do with the design of the later synths, the Prophet VS in particular. After Sequential went bust, John went into soft synths and developed something called Scope. A few years ago he had the idea to put Scope into a hardware implementation -- the Solaris. I was one of the pre-order customers, #37 in line last time I looked. I pre-ordered mine... well, it's been two jobs ago. I leave the math as an exercise to the reader. But I never had any doubt (well, almost never) that John Bowen and his team would come through. It was a long strange trip, full of redesigns, financial adventures, NLA parts, and a few untimely vacations. But it's here. It's sitting on the couch in my den. The Solaris is real.

Let's take a quick look at the panel. It's basically divided into four sections. Starting from the left:

Each of those windows is a two-line backlit LCD display (I took these pics before I turned the power on) that shows the parameters for some functional grouping. The upper half of this part of the panel is the oscillators and rotors parameters, and the lower half is the LFO and mod sources parameters. Within each of these sections, there is a row of buttons across the top where you select which specific unit you want to display (osc 1, osc 2, etc). A page comes up which displays names and values for five parameters, each laid out on the display above one of the knobs. Turning the knob changes the value of the parameter displayed above the knob. Pages are grouped into two sets, "main" and "mod"; you select the set using the button at the lower left corner of the LCD window, and you use the two buttons to the left of the LCD to scroll through pages. The row of buttons at the bottom of this photo is mostly performance parameters, such as unison mode, octave transpose, and arpeggiator on/off.

Next section:

The top half of this section controls the mixer and insert effects. The bottom half, with the larger screen (which looks crinkly because I hadn't taken the protective plastic off) is the main display where you select patches to load. It also contains a huge number of parameter pages. The buttons across the top select page sets for particular categories of control, such as the arpeggiator, FM and AM modulations, and the MIDI setup. This LCD contains two rows of parameters on each page, corresponding to the two rows of knobs beneath.

Next section:

This is similar in appearance to the leftmost section. The top half has the filter and VCA parameters; the bottom half is the envelope generators. The four buttons at the bottom enable layers; a patch can have four layers.

Rightmost section:

These are the patch storing and recall controls. There's a button for bypassing the effects. The numeric keypad can be used to enter values for any numeric parameter, as can the data entry wheel. The knob at the top left is the master volume. You can see a bit of the ribbon control at the bottom -- it runs across the whole span of the keyboard.

And last but not least:

Yeah, we got lit pitch and mod wheels (which are an option). OK, they're a bit cliched now. So what. The dark circle above is the joystick, which doesn't show up very well in this photo.

So I've played with it a bit, and I can tell you that it looks and sounds wonderful. So far I've tried only a handful of the factory patches (some of them voiced by Bowen himself), but I've come across some incredible evolving pads, some really grungy leads, some weird vocal-like sounds, and a few what-the-heck-is-its. I've barely touched the patch controls so far. But it appears to have tremendous potential, a synth that will take years to master. I'm looking forward to the challenge.

Ladies and gentlemen, may I present to you... the Solaris.

New MU-format Manufacturer: Corsynth

2011-10-19T20:25:00.003-05:00

Corsynth is a company based in Spain that has just introduced their first product: the MU-formatted C101 OTA-based lowpass filter. Nice to see a new player in the game, especially in the 5U formats, which the past few years has been taking a back seat to the Euro-format world.

OTA-based filters have a different sound and different characteristics from the more widely known transistor ladder filter, a design and sound usually associated with Moog. The classic Roland analog synths such as the System 100, the Jupiter-4 and -8, and the Junos used OTA-based filters, and so people who hear an OTA-based filter often describe it as the "Roland sound". OTA filters aren't very common in the modular world; the only other one I know of currently available in any 5U format is the Synth Tech MOTM-440, and it's not really a "typical" OTA filter because Paul Schreiber designed his own discrete OTA circuit for it. I have a 440 and it's a rather different beast. Judging from the sound samples on Corsynth's Web site, the C101 captures more of the classic, smooth Roland filter sound.

Here's a photo of the module, borrowed from their Web site:

As you can see, it has some interesting features. The most important ones are the voltage-controlled resonance and the 1V/octave control input. Getting any analog VCF to properly scale to 1V/octave is no mean feat, and it allows for some interesting tricks. There's obviously using the VCF as a VCO by turning the resonance up into self-oscillation, but the 1V/octave input also allows for things like harmonic tracking, e.g., you can run the output of a VCO into it, and feed the VCO's control voltage into the 1V/octave input. Then, you set the cutoff knob so that the cutoff centers on a harmonic of the VCO's output, and the filter will stay with that harmonic as the VCO frequency varies.

The two FM inputs, each with its own attentuators, will save you from having to use a CV mixer when doing complicated modulation. There's the two audio inputs, one with an attentuator, and the oversize cutoff frequency knob is a nice touch. Panel graphics are legible and easy to understand.

This thread at Muff's has a good discussion, and more photos, including a photo from the rear. The physical layout is interesting. There's a board that sits behind and parallel to the panel, Dotcom-style. But unlike most of the Synthesizers.com modules, this board carries only the panel controls. The core of the circuitry is on a daughter board behind the panel board, and connected to it via a ribbon cable. Probably the main motivation for doing this was to make it possible to adapt the design to other formats. The construction appear to be all through-hole, with all of the ICs socketed, which makes for easier repairs and mods. The board has both Dotcom and MOTM power headers. I've asked a question on the thread about what OTA was actually used, if it was the LM13700 or something else. (Unfortunately the CA3080, which was used for this type of application for years, is out of production and no longer available.)

The C101 is a pretty good-looking module, and a type of circuit that doesn't appear often in the modular world. The Web site lists the module at a price of E210, which currently translates into U.S. dollars at $290.

The Fizmo Project

2011-10-17T20:34:00.000-05:00

As I wrote on VSE a few weeks ago, I have re-started my aborted project from a couple of years back to document the patch dump sysex format of the Ensoniq Fizmo. The first time around, I was trying to tweak patches and then manually examine sysex dumps for changes, and it became very frustrating. Changes were hard to find, and when I did find them, they seemed to make no sense.

A couple of months ago, out of the blue, I had an idea for how I could use Unix shell scripts in OSX to format dumps and find changes automatically. I put together a procedure that involves using Metro to capture the dumps and then a shell script to format and compare them. It compares two dumps, finds which bytes differ, and then lists the differences and where they are.

With this, I was finally able to get a start on the project. If you're not familiar with the Fizmo, it has a rather unusual patch structure. There are 64 memory slots for "sounds", each of which consists of two layers. Then, there are 64 "presets" (using the terminology from the user manual), which has four sound slots. Each one of the slots loads one of the 64 defined sounds. A preset also includes an insert effect and an arpeggiator setup.

So I set up a procedure for finding which bytes in a patch dump correspond with which parameters. Because there are so many parameters that you can't access from the panel, I'm using the Fizmo-specific version of Sounddiver that Ensoniq developed, running on an HP laptop. I have this connected to my MIDI network via a Turtle Beach 1x1 USB/MIDI interface. On my Mac, I'm using Metro to capture the patch dump sysex strings. The procedure steps go like this:

Change a parameter in Sounddiver.
Start Metro recording.
At the Fizmo, save the current patch and then dump it.
Stop recording.
In Metro, view the sysex string (in hex) and copy it to a text file.
Run my shell script to show me the differences between this dump and the previous dump.
Copy the dump text file to the "old" file name, for comparison to the next dump.

Once I started working on this, I realized one reason why I could never make any sense of the dumps before. I had had several people advise me that the dump format is similar to the MR/ZR sysex format, which is documented. Hah... not even close. The Fizmo patch dump format is one of the craziest things I've ever seen. First, the basics: it's 3097 bytes long. The dump for each sound (two layers) is 640 bytes. Why so long? I wondered about that too. Each layer has about 100 parameters, so 640 bytes seems a bit much.

Well... one thing I'm finding out is that the way that the parameters are arranged in the dump is just insane. There are random filler bits and bytes all over the place. Hardly anything lines up on a byte boundary. Parameters don't appear to be in much of any particular order either. And some data fields are oversized for the minimum and maximum parameter values. (It doesn't help that there seem to be a huge number of different and seemingly arbitrary choices about what the minimum and maximum of each parameter should be. Some go from -64 or -128 to +64 or +128; some go from 0 to 100, some from 0 to 49, and some are just random.)

So far, I've managed to work out a lot of the parameters for layer 1 of sound 1 in the preset. I've got most of the oscillator, filter, and amp parameters, plus the parameters for keyboard splits, velocity mapping, volume and pan, tuning, and portamento. I've mapped out all of the parameters for envelope 1, including figuring out a few that aren't explained on the Sounddiver screen or in its help instructions. I'm hoping that the parameters for layer 2 of sound 1, and the layers for the other three sounds, use the same layout. So far, in the few that I've checked, this seems to be the case, but I need to do more to make sure.

I've also found where it saves the on/off and edit-enabled status of the four sounds, and the two layers of each sound. I've also found where it saves the patch name (the Fizmo can't of course display the names, but Sounddiver can), although I haven't decoded the format yet -- it's not anything as simple as plain old ASCII. I found something else, by accident: did you know that it saves the values of the F-I-Z-M-O real time controller knobs? I didn't realize that either, but it does.

Another bit that I haven't yet decoded is how the wave selection parameters work. I suspect that the parameter value may be, or contain, an actual memory address. Apparently there is a table in the OS ROM that tells it where the start address is for each waveform that you can select. Potential for sysex mischief there.

What I'm doing with all this info is putting it into a spreadsheet. I don't have MS Office at home, so I'm using the spreadsheet editor in Open Office. Here's a screen shot of what I've got so far:

The color-filled columns represent the bit patterns of the various parameters; there is no particular system other that the color matches the color of the corresponding text. Each column represents one bit and there are seven. (Since the leftmost bit of a data byte must always be 0 in the MIDI protocol, I left that out.) I number bits starting with bit 0 on the right, to bit 6 on the left. The un-colored columns represent bits that are filler, or whose function I haven't discovered yet. The row number is the byte number. The middle column documents parameter ranges and anything unusual about the parameter format.

This version of Sounddiver is very buggy. It has a lot of trouble remembering its configuration parameters. Nearly every time I start it up, I have to engage in some kind of software gymnastics to get it to communicate with the synth. About half the time, it will start up and then inexplicably lose track of the MIDI interface, and I have to either kill it and restart, or manually reconfigure it, or both. It never does the same thing twice. Now and then, all of the numeric parameter values will mysteriously blank out of the display. And periodically, it goes off on a mental vacation -- it consumes 100% of the CPU for about ten seconds, locking everything else out while it is doing so. I get the impression that it's a hack; it will claim that you're communicating with an MR series synth, and the Fizmo is hardly every referred to by name. Every now and then an error dialog pops up with a cryptic message and the developer's name, as if the developer was still in the process of debugging it when it was released.

Which may be true... between the condition of this version of Sounddiver and the peculiarities of the software in the synth itself, I'm getting the impression that the whole Fizmo project was a rush job. I picture Ensoniq, in financial trouble, trying desperately to crank out a product that might save the company, in a very short amount of time. Developers grabbing pieces of existing products and jamming them together with things from the lab, and rushing the whole thing out the door with minimal verification. And it never had a chance to find its niche before Ensoniq sold out to Creative Labs and production was abruptly terminated. And someone forget to check the current draw to see if it was within the voltage regulator's specs...

I'll post more on the sysex format as I get time to work on it; right now I'm only finding about 4 hours a week to give to this project. I think it's going to take about 40 hours to finish it, so that takes me about up to Christmas. At some point, once I've got the spreadsheet a bit more complete, I'll start posting the latest version to my Web site. You'll need Open Office to read the file; it's available for Windows, OSX, and Linux, and you can download it for free here.

I'll finish this up with a screen shot of the Fizmo Sounddiver version. Shown is the patch editor window, with the parameters for layer 1 visible, and a piece of the effects parameters on the right. Note the rather primitive look and feel, and the amount of wasted space.

Where to get a Fizmo motherboard

2011-08-27T10:49:00.003-05:00

If you have one of the Ensoniq Fizmos that had its DSPs fried by a faulty voltage regulator, you probably know that E-mu used up its stock of spare motherboards several years ago. However, per a post on VSE, it appears that thesoniq.com still has some. How many is uncertain, so if you have a Fizmo that needs a new motherboard, get one now.

EML 101 -- calibrated

2011-07-17T21:25:00.004-05:00

It worked! Replacing that resistor moved the circuit characteristics enough that I was able to tune and scale all of the 101's three exponential converter circuits. I installed a 2.55K 1% resistor in place of the 2.2K resistor in the reference voltage circuit, and I measured 4.46V vs. the specified 4.4V. That wasn't a huge change from the 4.6V that it was outputting with the other resistor, but it was enough to give me the leeway I needed on the tuning and linearity trim pots. It now plays in tune at both ends of the keyboard. (Although, curiously, it's ever so slightly flat in the middle octaves. Maybe it's always been like that and I never noticed.) And I can now play all four VCOs in unison without much beating.

The process is a PITA if you don't have a high-accuracy frequency counter, and I don't. I use the A440 reference signal built into my Synthesizers.com Q123 standards module, and route that into the external input of the 101 so I can mix it with the 101's own signal. I set up a patch with VCO 2 only. I then play high A and tune it to eliminate audible beating. Then, I unplug the A440 from the 101's external input, and connect a Q106 VCO which is driven by control voltage from the Q123. I turn the Q123's octave knob up to the fourth octave and tune the Q107 to match the EML's high A. Then, I turn the Q123 down three octaves and compare it to the Q107's low A. Adjust the tuning trimpot until it matches. Set the Q123 back to the high octave; check the high A and adjust the scaling trimpot. Then, check the low A again, etc. It took me about 10 tries. Once VCO 2 and the keyboard 1 expo converter are tuned and scaled, do the other VCOs, and then the keyboard 2 expo converter. Finally, scale the filter (involves moving a few wires around).

EML 101 repair -- reference voltage circuit

2011-07-16T22:29:00.004-05:00

The EML 101 on its test fixture on the workbench.

I wrote in my previous entry (before unemployment and tornado disasters intervened) that I have not been able to get the EML 101 to scale. I think I figured out why. The 101 has a circuit that generates a 4.4V reference voltage that several other circuits, including the expo converters, use as a standard for developing other control voltages. Here is the circuit in question, from an old set of photocopied schematics:

The zener diode at the left, marked 1N823A, is supposed to have a reverse bias drop of 6.2V. The two resistors to the right of it comprise a voltage divider that drops this down to the 4.4V reference. That is filtered by the cap and fed to the 741 configured as a unity-gain buffer.

I had to hunt around for where this circuit physically was. I knew from a legend on the schematic page (not visible in the scan above) that the circuit was on board #1, the expo amp board. In the photo below, showing the reverse side of the panel, board #1 is at right, just to the left of the power supply:

EML 101 guts. The power supply is at farthest right, then the three main boards are numbers 1 thru 3, right to left: the expo amp board, the oscillator board, and the filter/VCA board. In this photo, the expo amp board has been detached from its standoffs for ease of access. The rear of the patching jacks can be seen across the top.

I figured it had to be near one of the 741s on the board, but I was confused at first because I didn't know what the 1N823 actually looked like. There were many ordinary glass-bodied ordinary diodes on the board; I figured it didn't look like those, but I wasn't sure what I was looking for. The data sheets I found for the 1N823 were no help because none of them had an actual photo.

I finally focused my attention on what appeared to be an oddly-marked resistor at the lower right edge of the board. In the photo below, the screwdriver points to the vicinity:

I had been looking at this "resistor" because the marking didn't make any sense. If you double-click on the photo below to expand it, you can see that it is marked sorta-blue, red, orange, and then a blue T that points towards the right end. It finally dawned on me to decode the stripes: if you allow that the sorta-blue might have been gray 35 years ago, they spell out "823", the part number of the zener I was looking for! I've never seen a diode marked like this. The blue T points towards the cathode end. Sure enough, this was it. The 2.2K and 5.6K resistors are right above the zener; the 1K resistor is to the left, and the round metal thing that appears in this photo to be a four-legged transistor is the 741. (This is how ICs were packaged in the early days, before the DIP package came into use. The other four legs of the 741 are obscured by the green wire.)

The 4.4V reference circuit. The "708" stamped into the board is this 101's serial number. This photo was shot through a 2X inspection magnifier. The finger is not part of the circuit (hopefully).

I measured the voltage across the zener and got 6.39V, higher than the spec'ed 6.2. Doing the math with this input voltage, I got 4.6V for the voltage divider output, which, sure enough, is what it measured. I considered replacing the zener, but researching the 1N823 further, I noted two things about it: (1) it is a temperature-compensated part, and (2) it appears that the only present-day suppler is NTE, and Mouser has it marked "not recommended for new designs", which suggests that it may not be in production much longer. And there doesn't seem to be an exact current-day substitute. So I figured that as long as the one that's there is still working at all, it would behoove me to find a way to make the circuit work.

The obvious answer is to tweak the voltage divider by replacing one of the resistors. I did the math for what value I would need in place of the 2.2K resistor, and I got 2.53K. Off to the Mouser catalog to see what they had in the way of 1% resistors. 2.53K isn't an available value; the two closest are 2.51 and 2.55K. So I ordered 10 of each, and I'll measure each one to see which one gets closest to the 2.53K value I need.

I've got the shipment now, and next week I'll replace the resistor. And then we'll see if my grand theory is right: that the 4.4V reference being off by 0.2V is what is preventing the 101 from playing in scale.

EML 101 and MIDI

2011-04-26T22:56:00.002-05:00

I've had several people asking me how I got a MIDI interface to work with my EML 101. I haven't answered the question yet because... well, to tell the truth, I don't remember exactly what I did, and unfortunately I can't remember which forum I posted the article where I did it. And Google isn't turning up anything.

I have the 101 on the bench now for calibration. While I've got it out, I'll reconstruct the mods that I made in order to be able to input a control voltage and gate. One thing I have to point out: I did not mod my 101 so that it will work with 1V/octave scaling! Several years ago I purchased a JKJ Electronics CV-5, which is scalable to the 1.2V/octave EML scaling, and that's what I use. Unfortunately the CV-5 is now out of production. I don't know whether other MIDI interfaces on the market can be scaled to 1.2V/octave or not. However, it would not be too hard to build a little circuit that would re-scale a 1V/octave control voltage to 1.2V/octave. I'll draw up one and post it.

It will probably be a few days because... now that I have my 101 on the bench, I can't get it to scale with its own keyboard. The linearity and scaling trim pots are all on the stops and it's still not close; I'm only getting a span of about 2 octaves across the keyboard. The really odd thing is that it's happening on both the KB1 and KB2 busses, which are almost totally independent circuits. Tomorrow I'll start by checking the 4.4V internal reference voltage.

Review: Oakleysound EFG envelope follower

2011-04-24T18:46:00.000-05:00

I recently purchased an Oakleysound EFG, pre-assembled from Krisp1 in the UK. (Oakley itself does not sell assembled modules.) This is a combination preamp, envelope follower, and gate generator. The module was on sale for (if I recall correctly) £110, which after exchange to American currency and shipping, worked out to a total cost of $195 US. Shipping time was about a week, and the module arrived well packed and protected.

This particular assembly is based on the issue (version) 4 board from Oakley. The one I bought is formatted as a 1U width MOTM-format module:

Oakleysound EFG, as assembled by Krisp1. Note that this panel layout is different from the 1U panel shown on the Oakley Web site.

What is an Envelope Follower?

So what is an envelope follower? Well, you know what an envelope generator is. It creates a signal whose (primary) purpose is to represent the volume of a sound over time; the signal is intended to be used as the control input to a VCA. An envelope follower, rather than creating an envelope from scratch, accepts an audio input and then outputs a control voltage that is proportional to the volume of the input audio at any given moment. The louder the volume of the input signal, the higher the control voltage output.

The Oakley EFG also contains a gate generator, that turns on when the input signal rises above a set level, and turns off when it drops below that level. For instance, as an alternative to having a VCA track the input level of your source, you could use an envelope generator connected to the VCA to re-shape the input signal's envelope. For this, you connect the EFG's gate generator output to the gate input of your envelope generator. The EFG's gate serves as an alternative to the normal gate signal that you would get from a keyboard connected to the synth.

EFG Features and Controls

An obvious application of an envelope follower is to make a VCA follow the envelope of some sound produced externally from the synth, such as the sound of a conventional instrument picked up from a microphone or electromagnetic pickup (e.g., electric guitar). These instruments and devices produce signal levels that are far lower than the signal levels used in a modular synth. Towards that end, the EFG incorporates a high-gain, high-impedance preamp, into which you can plug in a dynamic microphone, a self-powered condenser microphone, or an electric guitar/bass directly. The preamp has a very wide gain range, so that it can also accept line-level signals. It provides its output directly to the envelope follower's input; the preamp output is also available directly at a panel jack. A red "PEAK" LED on the panel indicates the onset of clipping in the preamp.

The envelope follower itself follows whatever signal is input to the preamp. A green "FOLLOW" LED on the panel lights up to indicate the control voltage being output; brighter indicates a higher voltage. This version of the EFG provides only one control for the envelope follower itself: a panel switch that selects slow or fast tracking. (Apparently the board can accommodate a pot that allows the envelope generator tracking speed to be varied.)

The gate generator outputs a gate signal that indicates the presence of a signal at the audio input. The gate is at 5V when a signal is present, and at ground when the signal is absent; this is pretty much the standard for modular gear. (It might or might not work with other synths; for instance, it won't work with Moog or Yamaha gear without conversion circuits.) The yellow "GATE" LED lights when the gate is active. The primary use of the gate is to control an envelope generator, in the same way that a keyboard gate does. You could then use the envelope generator to re-shape the audio signal. For instance, by setting your envelope to a fast attack, slow decay, and zero sustain level, you could make an instrument that sustains indefinitely (say, a saxophone or a violin) fade out like a piano.

The THRESHOLD control on the panel determines the level that the input signal has to reach in order to be considered "active". The RESPONSE control adds hysteresis to the threshold level; as you turn the control to the right, the turn-on level goes higher while the turn-off level goes lower. You can use this sort of like a noise gate. If your input signal is a source that fades out gradually, like a guitar or piano, you can set the control to a slow setting so that it the gate remains active for a time after the signal fades below the turn-on threshold. It helps prevent the gate from thrashing back and forth when the signal level is near the threshold. I found that when using an electric bass as the source, I had to set this to a high value to prevent the gate from chasing the waveform; in other words, turning on and off with each cycle of the waveform.

Using the EFG

I tried several things with the EFG. My first patch was to plug an electric bass into the EFG's audio input. I routed its audio output to an MOTM-190 VCA configured as a ring modulator, and fed the output of a VCO to the 190's other input. The output of the 190 went to a Dotcom Q109 VCA which served to actually control the level of the output signal.

I started by connecting the EFG's control voltage output to the Q109's control input. With this configuration, I could play the bass and have its output ring modulated against the VCO. The output from the Q109 tracked the envelope of the bass, so that notes played on the bass attacked and decayed the same as if I were playing directly through an amp. That worked well with the envelope follower tracking switch on SLOW; when I put it on FAST, the envelope generator chased the waveform somewhat. That was actually kind of interesting, but not what I was wanting at the moment.

Next, I plugged in an electric guitar. With the MOTM-190 still configured as a ring modulator, I unplugged the control voltage output from the Q109 VCA. I then plugged the EFG's gate output into a Q108 envelope generator, and plugged its output into the Q109. I was able to obtain steel-drum-like sounds by setting the envelope for minimum attack, short decay, and zero sustain. With a longer attack and high sustain level I could get things that sounded like radio interference, or detuned violins; it helps if you turn the RESPONSE control up so that the gate remains active long enough to reach the sustain phase. One thing that doesn't work is to set up for a long release and then mute notes on the guitar: no matter what you set the envelope generator release to, the VCA can't amplify a signal that isn't there! That's one thing you have to get used to when you use an instrument with an envelope follower.

I did find that it was rather difficult to set the THRESHOLD control to get the gate to behave exactly the way I wanted. With a moderate setting of the RESPONSE control, the threshold range between the point where the gate would go off too soon, and the point where it would stay on indefinitely due to noise picked up by the guitar, was pretty small -- between the 2 and 3 positions. Turning the gain up helps some, but then you get into clipping in the preamp section. If I were seriously going to use the guitar as an input to the EFG in a patch, I'd probably put a compressor in line ahead of the EFG.

Technical Details

The EFG circuitry consists of a single board, mounted at a right angle to the panel and secured to it via the three pots, in the style of most MOTM 1U width modules. The build quality is excellent; the soldering is very clean and the leads to the panel-mount components are nicely dressed. The board came equipped with the MOTM 4-pin power connector; there is also provision on the board to install a Dotcom power connector. A MOTM-style power cable, with latching connectors, was shipped with the module.

All ICs are socketed, for easy replacement if needed. The board has ferrite beads to clean up in the incoming power, and diodes to protect against the power cable being plugged in backwards. I couldn't figure out what brand the pots are, but the feel is smooth and I didn't notice any glitches when turning the knobs. The jacks are Switchcraft enclosed jacks, which are top-quality parts.

I did some tests to look at what kinds of voltages were coming out. For the envelope control voltage, with no input, the output is at zero volts, as you might suspect. I set up an input signal and varied the gain so that it was just below the point where the PEAK light comes on; at this level, the envelope follower outputs 6 volts. It will go higher, past 10V, if you let the input stage be overdriven, but of course at this level the preamp's audio out won't be very useable.

The gate outputs 7V when active, and I verified that it will gate an Encore Electronics universal event generator (a module that is known to be picky about its gates). With the gate THRSHOLD and RESPONSE controls set full counterclockwise, the gate threshold occurs at the point where the envelope follower is outputting about 1 volt.

The board has two unused features which could be exploited. There is a provision to add a lag control to the envelope follow control voltage. Also, by cutting a jumper, the preamp section can be separated from the input to the envelope follower and gate generator circuits, which would allow one to use the preamp for an instrument and then put its output through additional processing before inputting it to the follower.

Conclusion

To be honest, I had not previously considered buying any of the Oakley/Krisp1 modules solely because of the sucky (for Americans) dollar/pound exchange rate, which meant that a similar module from a North American source would nearly always cost significantly less. But when Krisp1 had their close-out sale on this module, I couldn't resist. I'm glad I made the purchase. This is going to be a very useful module, both for running conventional instruments through the modular, and for interfacing the modular to other synths. For instance, I could run the output of a polysynth into the EFG and use the control voltage output to control the cutoff of a VCF, adding an envelope-filter capability to the polysynth. Or, using the EFG in combination with a waveshaping circuit, I could make an approximation of a guitar synthesizer. My only comment is the small useable range of the gate THRESHOLD control. Other than that, it's a great module.

Remove Before Flight: the making of a trance track

2011-04-17T00:25:00.001-05:00

I've been wanting for a while to take a shot at an electronica track. I wanted to do it as a new challenge to myself -- I've listened to lots of electronica, but have not tried writing any before -- and as a way of putting some rhythm and melody back into my music. I've been doing a lot of atonal experimental stuff lately and I needed to get back in the groove, figuratively and literally. Here is the result, from my Web site.

The Web site writeup describes the who-did-what for this track, but I wanted to go a bit more into the decision-making process here. Remove Before Flight started out as a drum pattern. A very complicated, cluttered drum pattern. I started with a low tom (not a kick; more on that in a moment) hammering a 4/4, and added drum after drum until I had, well, a mess. Then I started taking stuff back out. I kept doing that until I had a recognizable pattern. The snare and cymbal play a bit while the tom keeps the beat pinned down. One thing I quickly recognized about it is that there is no obvious place for the "one" beat. When I put together the place where the first heavy pad sound comes in, I realized that I had started it on the wrong beat! I went with it, and there are several places in the track where "one" gets redefined, although it isn't glaringly obvious.

I had several specific things that I wanted to do with this track: (1) get the Kawai K5m involved; (2) build a modular patch involving the MOTM-510 WaveWarper; (3) experiment with Shepherd tones; and (4) put together an idea that I've had laying around for a while called the "drum console". The first two sounds that come in after the drum pattern starts -- the bass and electric-piano-like chords -- are both out of the K5m. I've done some stuff with the K5m before, but like a lot of people, I always found the additive synthesis method hard to get a feel for. However, after doing the Mississippi Statescape entirely with additive synthesis (albeit using the simpler Minky Starshine plug-in), I felt like I was ready to take another crack at it.

I started with a patch that I had built previously for part of Solar Flare. I tweaked on that, looking for an interesting melody patch, but it turned out to work better as a bass patch. Then I created another patch for the pulsing electric-piano-like sound. I recorded both of these as virtual tracks, so when I did the mixdown I had to put the K5m in multi mode, which I've never used before, and take the two patches out of separate outputs. And by doing that, I discovered something: the K5m is a lot quieter in multi mode! I'm not quite sure why this is; I think the K5m sums all five outputs together when it's in single mode. The K5m is known for being rather noisy; I've done the mod on mine to increase the output levels, but it's always tough to get the gain staging with it set up right. But from now on, whenever I record the K5m, I'm going to use the multi mode even if I'm only using it monotimbrally. (I found out something else interesting: in the multi mode, you can decide how many voices to allocate to each patch! I don't know of any other synth that lets you control the voice allocation in multitimbral operation this way.) Using poly mode 2 chops off the release phase of each note when the next note starts, giving more of a feel of a monosynth without using the 106's notoriously odd-sounding unison mode.

The high-pitched thing that sort of sounds like a pipe organ is from the JD990. Except for at the beginning, this is always layered with a big pad patch from the Matrix-1000. This turned out to be a good combination, with the somewhat shrill patch from the JD990 layered with the bassy and rather muddy Matrix-1000. I created the 990 patch specifically for this; the M1000 patch is one that I've had laying around. There are several things in the M1000 patch that are modulated by aftertouch to produce a somewhat unsettling detuning effect, which you can hear as the "chorus" portions of the track trail off into the quiet parts. It's a great demonstration of the M1000 modulation matrix.

The track actually uses three different bass patches. I started out with the K5m bass in the quiet parts, but I decided it was too up-front. I remember a patch that I had put together on the JD800 to emulate an electric bass. The basic patch is two layers; I had defined it with a third, disabled layer in an attempt to emulate an 8-string bass. That didn't really work out, but playing it way down at the far left end of the keyboard, it made a great near-subsonic bass for the quiet bits of this track. I knew that neither the K5m nor the JD800 basses would work as the sustained bass that I needed for the big choruses, so for that I created a third bass patch using the V-Synth. This patch has the ability to vary the low end harmonic content using the panel performance knobs (and corresponding MIDI CC #'s), which I use at the final chorus part. I mentioned using a low tom instead of a kick in the drum pattern. I did that because I knew I would be using a deep bass patch at some point in the track, and I don't like a kick drum cluttering up the bass spectrum. My observation is that when you do a track like this, you have to make a choice -- either use a deep kick and keep your bass patch in the mid-bass region, or use a deep bass and don't use a kick. Most electronica producers usually do the former, but I think I prefer the latter. (Must be my progressive-rock background coming through...)

I wanted a sequence that would capture the simplicity of a modular driven by an analog sequencer, but I didn't want to actually use an analog sequencer because (1) I didn't want the sequence to be quite that tick-tock, and (2) I don't have an analog sequencer. Having Metro drive the Juno 106 with a repeating pattern turned out to be just the thing. Funny thing about this: The sequence was one of the first things I did after putting together the drum pattern. I started it using a piano-like patch on the 106 as a placeholder, and I kept using that while I worked on other parts of the song. When the track was almost finished, I finally went back to it to create the patch I had intended, but nothing seemed to work quite as well as the piano patch, even though that patch wasn't what I had in mind and wasn't at all right for the track. I finally sort of compromised with myself by creating a patch that sort of sounds like what you'd get if you took a Wurlitzer and ran it through a wave shaper and an octave divider. This got me closer to what I wanted the sequence to sound like, without sounding too retro. The key to this patch was using gobs of the sub-oscillator, which has a different sound than the primary pulse waveform set to 50% duty cycle.

The Discombobulator plays a big part in Remove Before Flight. In the first quiet segment, there are a variety of sound effects produced using FM techniques. At the first of the year, I purchased three MOTM-310 micro VCOs from someone who was breaking up a large modular, with the specific intention of using them for FM patches. I set up a three-operator patch with one VCO modulating the second one, which modulates the third one. The first VCO is operating at a lower frequency and itself being modulated by an LFO. This patch uses triangle waves as the modulating waveforms; I've found that this often produces more interesting results than the sine waves used in the classic Yamaha FM synths. For this patch, no keyboard or other controller was used; I produced a track of about five minutes' worth of various sounds by tweaking the knobs. I recorded the audio in Metro and then chopped the track up into bits, about two minutes' worth of which actually made the final cut.

In the second quiet part, there's a series of atonal melodies. This was done using the MOTM-510 WaveWarper, which is sort of a super-duper ring modulator that takes three audio inputs and performs an analog computation on them. Two of the 310s were set to an interval (minor third, more or less) and sent to the X and W inputs of the 510. The third 310 was set to a lower frequency and modulated slightly by the low-frequency noise output of an MOTM-101. To play this, I used a keyboard interfaced via the MOTM-650 MIDI/CV interface. I configured a group of two channels and set the group to unison mode. An interesting feature of the unison mode is that if you play one note, both channels output that note. However, if you play two notes, the two channels split and each one takes one of the two notes. Each channel was fed to the control input of one of the 310 VCOs. By doing this, I could play the fixed interval that the 310s were tuned to by playing one note, or play another interval by playing two notes. Unlike the FM patch, this was not pre-recorded as an audio track; I recorded it as MIDI and played the track along with the other synths during the virtual mixdown.

The Discombobulator did one other function: That organ-like patch from the JD990 that I mentioned? It went through the Encore frequency shifter. I configured an aux output on the MOTM-650 to respond to mod wheel (MIDI CC #1) and used that control voltage to control the shift amount. The up-shift and down-shift outputs went to two channels of the mixer and were panned left and right for a stereo effect.

I did the Shepherd tones with Csound, as I have described in a previous post. I captured this as an AIFF file directly from Csound's output. This was imported into Metro and processed as described in the previous post.

The drum console is something that I'll say more about in a later post. One of my frustrations with conventional drum machines is that there is no easy way to change the pitch of the drums you select; for example, on the DR-202, you can change the pitch by going into the drum kit and changing it for that drum assignment in that kit, but it requires menu diving and is something you can't do at performance time. I've been wanting a way to play a particular drum sound over a range of pitches, to simulate e.g., roto-toms. So what I did is: I wrote a Csound program that loads a set of eight drum or percussion samples, and maps each one to a range of eight MIDI notes. Depending on the note played, it shifts the sample up or down, over a range of about half an octave. To play it, I use a Monome 40H, which has 64 keys in an 8x8 arrangement. Each row of the Monome corresponds to one sample; as you go from left to right on the row, it plays the sample at higher pitches.

Because the keys of the Monome aren't velocity sensitive, I coded the Csound program to take input from a Korg Nanocontrol to set a "velocity" parameter for each sample. Each of the 1 through 8 sliders on the Nanocontrol maps to one sample. When the slider is full up, the sample plays pretty much unaltered in dynamics. As you move the slider down, the attack time is increased; the volume is decreased, and a bit of low-pass filtering is applied. At low settings this actually produces a rather unrealistic attack sound, but I though it was interesting so I left it that way. Also, the rotary knob that goes with each channel is used by Csound to pan the sample.

I recorded the drum console parts as MIDI data from the Monome and Nanokontrol (which was tricky because I had to use the OSX IAC bus, and anytime you use the IAC bus you have to be very careful that you don't create a MIDI thru loop). So during the virtual mix I had this playing back through the Csound program. I routed the audio to its own output pair on the audio interface and to the mixer, where reverb got added. So, the resulting audio went from the digital domain to analog, and back to digital again. I learned a lot doing all this. It produces some interesting results, especially with the volume and attack manipulations. I want to extend this idea so I have a lot more control over various parameters and modifications of the samples. One other thing I learned is that the Monome isn't ideal as a drum pad controller; the buttons are kind of spongy and they have too much give for good drum pattern playing, in addition to not being velocity sensitive. But I don't know of any drum pad controller on the market that has 64 pads.

For the mixdown process, I had 12 MIDI tracks going on the various instruments, with everything being fed to the Mackie CR1604 mixer (of which I used all but one input channel, and all four of the effects returns). Nearly all of the mixing was pre-automated by setting up MIDI volume and pan commands to the various synths, so that once I had set the initial levels at the mixer, I didn't have to touch the faders during playback. There was a little bit of riding the master levels to avoid too much clipping -- I didn't use any compression. After that had been recorded to two-track in Metro, there were two additional (stereo) tracks that had to be mixed with the main two-track inside the box. One was the FM sounds from the Discombobulator, which had been chopped up from a longer track. The other was the Shepherd tones, which I wanted to keep in the digital domain, and anyway I had not set up the Shepherd program to be MIDI-controllable. That final in-the-box mix produced the track that you hear.

Remove Before Flight is Part I of the Flight Trilogy, all three of which will be electronica of some sort. I'm just starting to think about Part II. It will be called either "Sea of Crises" (a geographic feature on the Moon) or its Latin equivalent, "Mare Crisium". Right now I'm thinking it will be a downtempo piece, but I haven't decided that for sure.

The Discombobulator at work

2011-04-07T21:19:00.003-05:00

Here it is, hard at work on stuff for "Remove Before Flight":

Electricity for Synth-DIY'ers: Operational Amplifiers

2011-03-12T23:00:00.000-06:00

Now that we've covered the basic semiconductors and passive components, let's begin covering what happens when you put a bunch of them together on a slab of silicon -- the integrated circuit. One of the most basic integrated circuits, used in both analog and digital systems, is the operational amplifier, or op amp for short. Op amps are one of the most useful ICs; they can be used to amplify or attenuate a signal to any extent desired, invert a signal, and solve signal buffering and impedence-matching problems. Further, they can be used to conveniently implement active filters in the audio range, do voltage level comparisons, and a host of other analog signal processing functions.

The development of the operational amplifier actually precedes the advent of the integrated circuit. Op amps were first devised in the 1930s as a means of solving various amplification and impedence-matching problems in long line telephone circuits. At the time, they were obviously constructed using tubes since the transistor had not yet been invented; as the circuits were mass produced, they started being packaged into forms that allowed them to be treated as a single component -- some tube types were packaged such that the entire circuit could be plugged into a socket as if it were a single tube.

Philbrick Labs K2-W tube-based operational amplifier. Photo from Wikipedia Commons.

Much research in the pre-IC days went into miniaturizing these circuits, and making them work at lower voltages. A few op amps intended for high-end audio applications are still made as "bricks" implemented with discrete transistors, but nearly all other types are made as ICs these days. (An aside: One engineer who played a significant role in the development of the early solid-state op amps was a certain young Alan R. Pearlman, who would later play a much larger role in the development of the music synthesizer.)

The first monolithic IC op amps were developed by Fairchild in the early 1960s. They introduced the first one, the µA702, in 1963; it had some unfortunate characteristics and was not a success. However, in 1965 they followed this up with the µA709, which fixed the 702's problems and went on to see widespread use; it remained in production into the 1980s. Then things picked up; National Semiconductor challenged Fairchild by introducing its LM101/201/301 series in 1967. (The series of numbers represented different ambient operating temperature ranges, with the 3xx being the commercial series and the 1xx being mil-spec; this pattern continues today for most ICs designed by National.) Fairchild said "oh, yeah?" and the next year they introduced what would become the canonical op amp for decades: the µA741. The 741 was the first internally-compensated op amp (we'll get to compensation later). Variants of the 741 are still in production today.

If you are interested in reading a very good and detailed story of the development of the op amp, see this paper published by Analog Devices.

Opamp Basics

A basic opamp has two inputs and one output. The two inputs are known as the inverting and non-inverting inputs; as one might guess, the output will be inversely proportional to the signal at the inverting input, and directly proportional to the signal at the non-inverting input. That's a bit of an over-simplification, though; it often leads beginners to believe that what the opamp does is amplify the difference between the two inputs. There's more to how an opamp works than that, though.

Here, in three short rules, is what an opamp does (or tries to do):

The opamp wants the voltage at its two inputs to be equal.
It assumes that a feedback path exists between the output and the inverting input.
It uses this assumption to vary the output to try to make the two inputs equal.

If no feedback path exists, the opamp will drive its output to its minimum or maximum voltage, which depends on the power supply voltages and the opamp's circuit design. (Some opamps will go "to the rails" and some will stop a volt or two short.) Thus, most useful opamp circuits include a negative feedback path. Another way of putting this is that without any feedback, the opamp has extremely high gain -- almost any non-zero inputs drive it to clipping, in one direction or the other. (The theoretical ideal opamp has infinite open-loop gain. Real opamps of course cannot go to infinity, but most types have open-loop gain factors within the audio range in the tens of thousands.) The feedback path is what controls this gain. From there, it follows that how much feedback is supplied can control the opamp to provide more or less gain, and this is exactly the case.

Symbology

We'll introduce the symbology at this point. The symbol for a basic opamp is:

The two pins show on the left are the two inputs: inverting (indicated by '-') and non-inverting (indicated by '+'). The pin on the right is the output. The power and ground pins are frequently not show, but when they are, they are show sticking out of the top and bottom; any other special-purpose inputs that the op amp has will also be shown at the top or bottom. In isolation, illustrators will usually draw the opamp symbol with the non-inverting input above the inverting input. However, in a schematic, the reverse is often true; conventionally the feedback path is drawn above the opamp symbol, and since it connects to the inverting input, to reduce the number of crossing paths the inverting input frequently gets placed on top. Some schematics will use the '+' symbol to indicate the non-inverting input and the '-' symbol to indicate the inverting. But others will use the pin numbers of the specific IC, in which case you need to know what kind of opamp it is and have access to the pinout. In this article, we will keep the non-inverting input on top throughout, to reduce confusion.

The Basic Amplifier Circuits

There are two basic configurations of amplifier circuits using an opamp, the inverting configuration and the non-inverting configuration. The inverting configuration has the effect of the output being the inverse of the input; with the non-inverting configuration, the output is identical to the input except for gain. Although the inverting configuration is used more, the non-inverting configuration is easier to understand and it does have its uses, so we'll introduce it first. It looks like this:

Note the resistor connected between the output and the inverting input. This is the feedback path. Thinking back to the three rules, picture what happens when a positive voltage comes to the opamp from the external circuitry. It raises the voltage at the non-inverting input. The opamp then wants the inverting input to rise to the same voltage as the non-inverting input. In order to accomplish this, it raises the voltage level at the output.

That voltage goes over to the inverting input, where some of it is bled off to ground by the other resistor. The two resistors form a voltage divider, so the voltage at the inverting input will be some fraction of the voltage at the output. Therefore, the output voltage will have to go up higher than the voltage present at the non-inverting input before the voltages at the two inputs become equal. For instance, if the two resistors have the same value, then they form a divide-by-2 divider, and the output will have to rise to twice the voltage that is present at the non-inverting input before the circuit settles.

All of this happens very fast. The speed at which the opamp is capable of responding to its input determines its maximum response frequency, or bandwidth. Nearly every opamp on the market today has a bandwidth that extends well beyond the audio range, so in an audio application, you will rarely have to worry about it; you can assume that everything happens instantaneously. (If you get into radio, or high-speed data circuits, that assumption is no longer valid.)

The ratio of the two resistors determines the gain of this circuit. The ratio in the middle of a two-resistor voltage divider is the total resistance divided by the resistance of the "bottom" resistor; in this case R2. So the gain of the non-inverting amplifier configuration is (R1+R2)/R2, or (R1/R2)+1. In the simplest case, where R1 is replaced by a straight wire (i.e., zero resistance) and R2 is omitted (infinite resistance), the result is a unity-gain amplifier; the output will be identical to the input. What good is a unity-gain amplifier? Well, one use is for re-driving signals that can't take being loaded down, such as signals from guitar pickups. Many opamps have extremely high input impedance, and a unity gain buffer is perfect for taking signals from high-impedance sources and repeating them with lower output impedance. Note one implication of the gain formula: in the non-inverting configuration, you can't achieve a gain of less than 1.

(By the way, in practical circuits, it's not a good idea to totally omit R2. Opamps often "bleed" a very small amount of current internally back into their inputs. If this bleed current doesn't have a path to ground, it can cause the opamp to go open loop and drive the output to the rail. If the need is for a unity-gain circuit, a 1M resistor can usually be used for R2 without any practical effect on the gain ratio, and it will allow the leakage currents to bleed off.)

The inverting amplifier configuration looks like this:

How does this one work? Per the three rules, when a positive voltage comes from the external circuit, it applies a positive voltage to the inverting input. Because it is the inverting input, the opamp will respond by pushing the output negative, in an attempt to counterbalance the positive voltage and bring it back to the voltage level of the non-inverting input, which is at 0V since it's grounded. The voltage level that the output has to go to in order to accomplish this depends on the ratio of the two resistors. Since the resistors directly determine the proportion of the input signal and the feedback that get combined at the inverting input, the gain equation for the inverting configuration is simply R1/R2. Unlike the non-inverting configuration, the inverting configuration can achieve a gain of less than 1; as you can see from the formula, as R2 gets large, the gain goes towards zero. Thus the inverting configuration can be both an amplifier and an attenuator.

So when do you choose one configuration over the other? Well, for one thing, the inverting configuration obviously inverts the signal, which is useful in its own right sometimes. More importantly, though, is that you will see below that most of the more sophisticated op amp circuits require an inverting configuration -- they don't work the other way. So why not just use the inverting configuration for everything? Well, for one thing, this may result in unintended inversion of the input, which will make you have to add an extra inverting stage to get the signal right side up.

However, a more important reason is in the fact that, in the inverting configuration, the two gain-setting resistors provide a "sneak path" around the op amp between the input and the output. This can cause a problem regarding the input impedance of the circuit. The op amp itself has incredibly high input impedance, which can be a very useful property (in, e.g., a preamp for a mic or guitar). However, if the op amp is driving something else with low input impedance, the path through the resistors will lower the effective input impedance. You could fix that by using high values for both resistors, since the ratio of the resistors and not the values per se is what determines the gain. However, past a certain point (when one of the resistors gets up past about 100K), you will start to have noise and voltage offset problems.

The non-inverting configuration does not have this problem because there is no connection between the input and the feedback path. So what you often see is that, for a given device, non-inverting circuits are often used for the external inputs and outputs, and most everything in between uses inverting circuits.

Stuff You Need to Know at This Point

Now that you understand the basic principles behind the opamp, before we proceed let's go over some of the requirements and practical limits of real opamps. Most opamps require both a positive and a negative power supply, depending on the type, and these must usually be of the same magnitude in order for the opamp to work properly. In other words, if the negative supply is -15V, a positive supply of +5V won't do; you need +15V. The power supply voltages limit how far the output can swing in either direction. They also limit what input voltages are permitted; with most opamp types, the inputs must be kept at least 1-2V away from the supply voltages. If the input limits are exceeded, the opamp may go into a condition called "latchup" where it no longer responds to its inputs. Latchup generally requires that the power be removed to reset the circuit, and it may harm the IC. If there is a possibility of the inputs exceeding the opamp's specified voltage, such as an opamp connected to an external input jack, it is a good idea to protect the inputs. Usually this is done with zener diodes. There are opamps which can accept input voltages all the way to the supply voltages; check the vendor catalogs if you need one of these.

There are some opamps made to work with single supply voltages. These are sometimes useful particularly in battery-powered applications. With these, the input voltages must be kept between the power supply voltage and ground, so if you want to feed in an AC signal, you will need to add DC offset to the input and then remove it from the output.

Most modern opamps are protected against having their outputs short-circuited; if you short the output to ground, nothing bad will happen. However, there is a limit to how much current the output can supply, so if you short it to ground, the output will go to zero and stay there until you remove the short. The output current limits are printed in the specifications. If you are building a circuit to drive a very long cable -- say, an output module for a modular that needs to drive to the house mixer 75 feet away -- an opamp will a high-current output capability will be useful.

One other thing to be aware of: All op amps "leak" very small amount bias currents that occur internally on the inputs. These small currents have to have a path out via the input pins. Here's something to watch out for:

Note the capacitor on the non-inverting input; this would be typical if it was desired to block DC voltages from reaching the input. The problem is: those micro-leakages at the input are "trapped" by the capacitor; they don't have a path out of the circuit! The result is that the circuit will never reach equilibrium and the output will probably go to the minimum or maximum voltage and stay there. Here's how you fix that:

The extra resistor allows the leakage currents to bleed to ground. A 1M resistor will usually be large enough to not have any effect on the input signal.

Analog Arithmetic with Opamps

A frequent problem in synth circuits is the need to add or subtract control voltages. For instance, the control voltage fed to a VCO might need to be the sum of control voltages from the keyboard, the pitch wheel, and an LFO. Here is a circuit that will sum up any number of control voltages:

You can add any number of inputs to the input ladder at the left. To keep things simple, all of the Rx resistors should be of the same value; if you do so, the gain is Ry/Rx. Use Rx values of at least 1K to prevent the input signals from trying to back-drive each other. Note that, because this is based on the inverting-amplifier configuration, the output will be the negative of the sum of the inputs. That's easy to fix; just run it through a second inverting amplifier, with the gain configured as needed. Note that this circuit can also be used as an audio mixer; after all, a mixer is really just an adder. The one thing you might want to do if it is being used for audio mixing is add a DC-blocking capacitor (10 uF or so, non-polarized electrolytic) in series with each input.

You can subtract signals using the configuration above, preceded by an inverting amp that inverts the signal to be subtracted. However, here's a one-op-amp solution for computing the difference of two signals:

Note that this circuit is identical to what is commonly called a "difference amplifier" or "differential amplifier" (the latter is not-so-good terminology, but lots of people use it). The "balanced line" signal connection often used in recording studios consists of two signals, one of which is the negative of the other. You see this commonly in equipment such as microphones which use the 3-pin "Cannon" plugs and jacks; two of the pins carry the signals and the third is connected to the cable shield. You can use the difference amplifier circuit to convert this to a single-wire signal that you can process inside a piece of equipment. The difference amplifier produces an output which is proportional to the difference between its inputs. An interesting implication of this is that if both signals change by the same amount, the output does not change at all, because the difference between the signals didn't change. In the studio environment, this is called "common mode rejection" and is a big help in reducing noise pickup. The basic circuit should have R1 = R3 and R2 = R4; assuming so, then the gain is equal to R1/R2.

(Note that this is not the optimum version of this circuit; the main problem is that it's difficult to make the impedances of the two sides match exactly, which limits the circuit's accuracy. Here is an improved version, if you have a need for it.)

Differentiators and Integrators

Here's a pair of circuits that can be very useful particularly in a modular synth. The first is called a differentiator (not to be confused with a difference amplifier). What it does is produce an output which is proportional to the rate at which the input is changing; you can think of it as sort of a speedometer for the input signal. If the input is increasing, the output will be a positive voltage; if the input is decreasing, the output is negative, and if the input is not changing, the output is zero.

The gain of this circuit is proportional to the frequency of the signal fed into it, but is influenced by the values of R and C. The output voltage Vo = -RC(dV/dt), where R is the resistor value in ohms, C is the capacitor value in farads (not microfarads or nanofarads, but whole farads), and dV/dt is the rate at which the input voltage is rising or falling, in volts per second. (Don't worry about what the individual parts of the dV/dt stand for; just accept that it stands for "change in voltage with respect to time".) Note the minus sign, which is a result of this being an inverting-amp configuration. An easy way to think about this circuit is to assume an R of 1M and a C of 1 uF; this way the product of RC is 1, and then if the input is rising at 1V per second, the output is -1V as long as the input keeps changing.

The integrator is, mathematically, the opposite of the differentiator. To continue with the car analogy, you can think of the input as an accelerator for the output. When the input is positive, the output rises; its rate of rise is determined by the magnitude of the input. Same goes for a negative input; the output goes lower in proportion to the input. When the input is zero, the output remains where it was. Given that an input that remains positive or negative for any length of time will push the output to its power-supply-determined limit, it's useful to have a reset capability. This is shown in the schematic as a switch, but it would be a simple matter to substitute an NPN transistor or a mini relay so it could take a reset trigger signal.

Basic Active Filter Circuits

Consider the following circuit:

Note the capacitor in the feedback path. What is this capacitor doing to the feedback? Well, start by considering a very low frequency: At a low enough frequency, the capacitor is essentially an open circuit, so only the R1 resistor in parallel with it has any effect. This reduces to your basic inverting amplifier with gain determined by the formula for an inverting amplifier: R1/R2. But what happens as the frequency goes up: The capacitor starts to let some current through, in addition to what goes through R1. In effect, the capacitor is reducing the value of R1 by allowing some current to bypass it. This gets more pronounced as the frequency goes up, until the frequency is reached where the capacitor has effectively shorted out R1. At this point, R1's effective resistance is zero, and the full amount of feedback is being applied to the inverting input, which cancels out the input signal at that frequency and reduces the op amp's gain at that frequency.

This circuit is a low-pass filter. It has a certain gain at low frequencies and less gain as the frequency goes up; at a sufficiently high frequency, its gain is zero. Doing the math, we find that, just like any filter, there is a cutoff frequency determined by the capacitor's value and above that value the gain rolls off at 6 dB/octave, making it a single-pole filter. The cutoff frequency is: 1/(6.2832R1C) where R1 is in ohms and C is in whole farads.

This circuit is a high pass filter. From the above discussion, you can probably already guess how it works. At low frequencies, the capacitor is essentially open, so full feedback is applied to the inverting input, and the op amp's gain is zero. As the frequency increases, C starts to conduct and at a high enough frequency it essentially becomes a straight wire. At that frequency, the circuit's gain then is determined by R1/R2. The formula for cutoff frequency is the same as for the low pass filter. Note that this circuit, unlike the low pass filter circuit, is non-inverting with respect to the input. Inverting and non-inverting configurations exist for both low pass and high pass filters.

What if you want more than a one-pole response? Well, an obvious answer is to put several of the above circuits in series. For instance, if you use two, you get a two-pole filter with a 12 dB/octave rolloff. However, using this method for multi-pole filters means using an op amp for each stage, which can make for an expensive circuit. So, electrical engineers have devised all kinds of devious circuits for implementing multi-pole filters that do not need an op amp for each pole. For instance, the Sallen-Key circuit often used in synth filters implements a two-pole low pass or high pass filter with only one op amp.

Sallen-Key filter, from Wikipedia Commons

Op Amp Filter Topologies and Response Types

Here is something that people often get confused about when they read about op amp-based filter circuits. The topology is the logical arrangement of the components in the circuit. There are a bunch of ways of designing filter circuits, particularly multi-pole filter circuits. The Sallen-Key is one such topology; others often found in synths are the biquad and the multiple feedback. The response class refers to the general shape of the response curve. The three types you will hear most often in audio systems are Butterworth, Chebyshev, and Bessel; most synth filters belong to the Butterworth class. The differences between them are in exactly what the frequency response curve looks like; for instance, the Butterworth type is very flat in the passband, while the Chebyshev has some variations ("ripple"). All three of these filter types can be implemented in various topologies, and usually the difference is in the ratios between certain components. There is some pretty complicated math behind these, but you don't have to do the math; there are all kinds of design tools that allow you to plug in the desired characteristics and it will calculate the component values for you. Here is a simple one which will calculate a Sallen-Key (12 dB/octave) low-pass filter of the Butterworth class; all you have to put in is the desired cutoff frequency and one capacitor value. I'll do a future post which explains some of these active-filter types in more detail.

Compensation and Offset Nulling

All op amps require what is called compensation, in order to keep the op amp from going into self-oscillation under certain circuit conditions. The compensation consists of one or more capacitors that are placed into the op amp's internal circuitry. The first op amps were externally compensated -- they required the designer to add capacitors to compensation terminals on the device. However, most op amps today are internally compensated; the compensating capacitors are built into the op amp itself. In terms of ultimate performance, internally compensated op amps are a compromise; using external compensation allows the designer to extend the op amp's bandwidth and/or reduce its power draw, depending on the need. However, in designing synth circuits, these are seldom important factors. Therefore, there is usually no reason not to use internally compensated op amps in synth circuits.

The ideal op amp should, if both of its inputs are grounded, drive its output to exactly 0V. Real op amps will usually have an offset ranging from microvolts to a few millivolts, depending on how the inputs are configured. This is caused by small current leakages in or out of the input pins. Some op amps have inputs to which you can connect a trimmer pot that you can adjust to null out the offset. This can be important in, for instance, a control voltage input to a VCO. If you need this, look for an op amp type that has "balance" or "offset null" inputs.

Op Amp Types and Specifications

These days, nearly all op amps are packaged in the familiar dual-in-line plastic package, in either through-hole or surface mount configuration. (A very few of the oldest types are still available in the metal can that looks like an oversize transistor with more legs.) A single part often contains more than one actual op amp; dual, triple, and quad packages are common. Single types nearly always have 8 pins while types containing more than one per package may have up to 16 pins.

Things that you (maybe) need to be concerned about when you are searching for an op amp for a synth application are:

Number of individual op amps in the package
Minimum and maximum power supply voltages
Input impedances
The gain-bandwidth product
Whether the op amp can run off of a single supply voltage
The voltage range of the output
Output drive capacity

We had a lot of fun back in the history writeup at the beginning of this article talking about the 741. However: Unless you are replacing a part in a vintage synth, don't even think about using one today. There are much better types available now. A popular general-purpose type is the TL08x series, originally designed by Texas Instruments but available now from multiple sources. These are available in various packaging containing 1, 2, or 4 individual op amps per unit. The inputs are JFET circuits and have very high input impedance. They can be powered at ±12V or 15V (the voltages used in modern modulars), and the permissible input voltages range to within 2V of the supply voltages. Another useful general-purpose type is the LM358, also available from multiple sources. It has most of the same virtues as the TL081 (single op amp per device); the input impedances aren't quite as high, but still adequate for most synth circuits. And they are inexpensive enough that you can buy a tube of 50 or so, to keep on hand for circuit prototyping or other off-the-cuff uses. Here is a table containing some potentially useful types that I extracted from the Mouser catalog. (The file is a .CSV file, which you can download and import into any spreadsheet application.)

More Shepherd Tones from Youtube

2011-01-17T22:14:00.003-06:00

These are not mine... they are things I found while doing various searches for Shepherd tone examples.

First example:

I think the author is using this Reaktor ensemble, from the Native Instruments user-contributed library. It starts out using tones at octave intervals, and I think the cycling is rather obvious. It becomes more convincing when he sets it to narower intervals, starting about 0:55 in.

Second example. This one uses discrete quantized intervals rather than a continuous glissando:

As you can see, the author added a note indicating that he doesn't think it turned out very well. However, I actually find it more convincing than the first example. This is despite the fact that, according to the author, it only uses two tones at a time. I think this may have to do with the chime-like tonality of the tones used; there's a lot of upper overtones and it sounds like some of the overtones are enharmonic. The addition of the drum machine track, while not necessary for demonstrating the principle, adds a nice touch and illustrates a possible use in a musical context. Unfortunately there is nothing in the author's notes about what hardware or software he used to make this.

Third example. I was utterly floored by this one:

Now obviously there's a lot more going on here than just sweeping tones up and down. The author mentions that it uses FM; I don't know if it's using the FM to do some of the tone sweeps and spreading, or just for tonality variations. He apparently did this with a Reaktor ensemble that he built himself. Note the stereo effects. In some places the illusion is very convincing; in other places cycling is somewhat obvious. It seems to hit a place at about 5:10 where the sweep can't go any further, and then at 5:58 it starts back in the other direction. Frequency shifting? The author's notes don't say.

(After you finish listening to this one, go to the author's Youtube page and check out his favorites -- rather interesting list.)

Shepherd tones

2011-01-15T22:38:00.004-06:00

I took a crack at making some Shepherd tones today, for a tune I'm working on. (He's trying to do trance! Run! Hide!) Shepherd tones are a form of audio illusion; they are supposed to give the impression of a tone that rises or falls forever without ever moving out of the audio range. This is often referred to as a "barber pole" effect; Shepherd tones are one mechnism for creating it. I had hunted around for some Shepherd tone samples on the Internet, and until today I had not had much luck; in most of the examples I found, either the "cycling" was obvious (it sounded like they rose/fell to a certain point and then obviously started over), or there was no sensation of moving pitch at all. I tried copying and running the example given in the Csound manual, which employs frequency shifting, but all I got out of it was phase cancellation (sounding like pulse width modulation of a square wave).

So I decided to code up my own algorithm in Csound. I chose to build the tone out of six sine waves, at intervals of an octave. Each time rises six octaves before it is faded out. I wrote the code to gradually increase the frequency exponentially, so that it is constant-time with respect to octaves (in other words, it takes the same amount of time to go from one octave to the next at low and high frequencies). Below is my code -- first, the orchestra code:


sr = 44100
kr = 4410
ksmps = 10
nchnls = 1

1nstr 1

ifreq     = 200 ; base freq * 2
idur        = p3
iatk      = p3 / 5
irel        = p3 / 5
ireps      = p4 - 1

kruntime init 0
; At the octave break point, start the next one
if kruntime < 1 kgoto keepgoing
    schedwhen 1, 1, 0, idur, ireps
keepgoing:

; If repeat count has decremented to 0, do nothing
if p4 > 0 kgoto dontquit
    turnoff
dontquit:

; Compute the exponentially rising frequency
; (necessary to make it linear with respect to half-steps per second)
ktime         line 0, idur, idur
kruntime      line 0, (idur/6), 1
kexp          = ktime/(idur/6)-1
kpower        = powoftwo(kexp)
kfreq         = ifreq * kpower


;Envelope and Oscillator
kenv         linen 1, iatk, idur, irel
aout         oscil3 6000*kenv, kfreq, 1

; Output it
out         aout

endin

And now the score file:


; Sine
f 1 0 16385 9 1 1 0

t 0 60

; inst start duration reps
i 1      0       30       20

Most of this is pretty straightforward to someone with just a bit of experience with Csound. The "powoftwo()" function is a bit weird since it only takes arguments in the range -5 to +5, so a bit of finagling was needed there. The tricky bit is that the score file only plays the first tone. After it has increased in frequency one octave, it uses the "schedwhen" to create a dynamic score event that starts the next tone. The line opcode and the kruntime variable do the timing for when to kick off the next tone. There is a repetition count given in the original score event that is decremented every time a new tone kicks off, and when it decrements to zero, the code quits generating new tones, which eventually brings the whole thing to an orderly end. I originally had the test of kruntime at the end of the code block, and I found that I had timing problems -- with the arguments given, the next tone kicked off a few milliseconds either early or late, depending on what value I used for the kr rate. That created noticable beating between the tones because the intervals between them weren't exact octaves. Once I moved the kruntime test to the start of the block, the timing was dead on after that.

I had Csound render this to a file, which created two minutes' worth of audio. I then loaded it up into two channels in Metro and panned the two hard left and right. I offset the start time of the right channel. Then I ran both through the Expert Sleepers phase shifter plug-in. Then, to give them more presence, I ran them through an ambient (small room reverb) algorithm on my Lexicon MPX500 and re-recorded the result to a stereo channel. Here is a brief sample of the result.

I think the results are pretty decent. The cycling isn't too terribly obvious, once all six tones get kicked off and running. I'm going to experiment with it more Based on some Youtube videos I watched today, I'm not sure there is anything special about the octave interval (contrary to theory); I'm thinking many closely spaced tones would produce a better effect with less noticable cycling. But I'm going to use what I've got for the tune I'm working on.

Sysex strings for controlling JD-800 tones

2011-01-12T20:22:00.007-06:00

If you've spent time working with sysex on the Roland JD-800, you may have noticed that the tone layer/active buttons don't send sysex. This means, among other things, that if you are trying to turn tones on and off during a performance and record it with a sequencer, these actions will not be recorded.

There is, however, a sysex string that you can send to a JD-800 to turn tones on and off. Here is the string; it's expressed in bytes as two hexadecimal digits per byte (most software sequencers will allow you to copy and paste this into a sysex editing dialog box):

F0 41 10 3D 12 00 00 21 0x 5y F7

The x and y are variables. A byte-by-byte breakdown of the string:

F0 The MIDI standard command byte that indicate that a sysex follows.
41 Roland's manufacturer ID.
10 The JD-800's unit number. The value "10" represents the factory default unit number of 17. If you have set your JD-800 to some other unit number, see the note at the end of this post.
3D The model number for the JD-800.
12 The command ID that tells the JD-800 to receive the following data.
00 00 21 The address of the tone layer parameter in the single-mode edit buffer.
0x is the parameter that tells the JD-800 which combination of tones should be on. The table below tells you how to set this value.
5y is a checksum, which the synth uses to check to see that the sysex string was transmitted correctly; if the checksum value is incorrect, the JD-800 will ignore the sysex. The table below tells you how to set this too.

Set the x and y values according to this table:
[apologies for the white space problem below; I haven't been able to fix it]

Tones On	X	Y
A	1	E
A, B	3	C
A, B, C	7	8
A, B, C, D	F	0
A, B, D	B	4
A, C	5	A
A, C, D	D	2
A, D	9	6
B	2	D
B, C	6	9
B, C, D	E	1
B, D	A	5
C	4	B
C, D	C	3
D	8	7

Sending the string turns on and off the selected combination of tones in the currently loaded patch. It only effects the edit buffer; it's exactly the same as pushing the buttons on the panel. The changes are not saved unless you write the patch to memory.

If you've changed your JD-800's unit number, you need to change the unit code in the sysex string. The JD-800 allows its unit number to be set to any value in the range 17-32. For unit number 17, the sysex code is 10, as show above. For unit 18, the code is 11; for unit 19, the code is 12, and so on... up to unit number 26, for which the code is 19. For unit 27, the code is 1A; for unit 28, the code is 1B, and so on... up to unit number 32, for which the code is 1F.

Fear and loathing of tantalum capacitors

2010-12-08T21:02:00.000-06:00

Why is everyone so afraid of tantalum capacitors? Lately I see posts in various places from people who are anxious to go through all of their synths and rip out any tantalum caps that they find. Completely unnecessary, provided that the person who designed the circuit that the tantalum cap is in was designed by a competent engineer, and that it was installed properly. There's a few rules that you need to be aware if you are going to use a tantalum cap in a synth (or any other kind of circuit), but if you do it right, tantalums have significant advantages over electrolytic types.

Solid tantalum capacitors. The loose capacitors at left are 10 uF; the caps fastened to the tape at right are 1 uF. Note the penny for size comparison.

First, let's talk a bit about how tantalums work. Tantalum capacitors have something in common with electrolytics: in both cases, the capacitor dielectric consists of the oxide of the metal that one of the electrodes is made of. In the case of the electrolytic capacitor, it's a layer of aluminum oxide that forms when the electrolyte reacts with the aluminum anode under an electric charge. The electrolyte is necessary to maintain the oxide layer. When it eventually evaporates or leaks out of the capacitor's container, the oxide layer breaks down and the capacitor loses capacitance.

Most tantalums used to be made the same way; these were called "wet slug" types. The ones that fail inside '70s ARP synths are wet slugs. The problem with these is that, because tantalum is pretty unreactive compared to aluminum, strong acids have to be used as electrolytes. These eventually eat up the seals around where the leads penetrate the body, and the electrolyte gradually leaks out. However, at some point, someone figured out that they could use a process where the tantalum "slug" is dipped in an electrolyte at the factory before the capacitor is assembled, and connected to a voltage to form the tantalum oxide dielectric layer. Then, the slug is removed, dried out, and assembled into a capacitor; the electrolyte is used only at the factory. Because tantalum oxide is more stable under electric charge, the electrolyte isn't necessary to maintain it. These are the "solid tantalum" or "dry slug" types. There are several different designs used in solid tantalums, but they all use the same basic idea -- a tantalum slug forms the anode; a layer of tantalum oxide built up on it is the dielectric, and some material (often manganese dioxide) is bound onto the outside of the oxide layer to form the cathode. Thus there is no electrolyte inside the assembled capacitor; since there is nothing to leak out, the capacitor will not degrade or lose value.

Cutaway drawing of a surface-mount solid tantalum capacitor. From an NEC data sheet.

As it turns out, tantalum oxide has a very high dielectric constant. This means that a tantalum capacitor can use a very thin layer of oxide, only a few microns. The basic rule of capacitors is that the capacitance value is directly proportional to the surface area and inversely proportional to how far apart the electrodes are; since the electrodes in a tantalum cap are separated only by the thin oxide layer, a tantalum cap can pack a lot of capacitance into a small volume.

The problem with having an exceedingly thin dielectric layer is that it can't withstand high voltages; it doesn't have enough insulation value. Thus one of the first rules for dealing with tantalum caps: do not expose them to excessive voltages. Most references I've seen recommend that tantalums be derated to 50% of the rated voltage. Some say to go down to 30% when the tantalum is used in a low-impedence, high-power circuit. Inexpensive tantalums are usually rated for 25-50V, with higher voltage ratings becoming harder to obtain above 10 uF.

The second thing about tantalums is something that they are notorious for: exploding. Why does this happen? Two words: reverse voltage. The polarity of a tantalum cap must be respected. Any reverse voltage breaks down the dielectric layer. It reduces the oxide back to metallic tantalum, which then forms a short-circuit path; if the circuit the capacitor is in can supply a lot of current, the short path heats up rapidly and the capacitor goes boom. The third point is related: ripple current. Large amounts of ripple have a similar effect: they create localized heat spots in the dielectric, which breaks down at those spots and a short circuit results. So it is generally best to avoid using tantalum caps in applications where they will be exposed to high ripple current. (Nonetheless, some power supply makers use tantalums as filter caps in their supplies, and it works. How they get this to work, I don't know.)

So why use tantalums? For one thing, they are small and light, much smaller than electrolytic caps of the same rating. Second, solid tantalums can't leak because they contain no electrolyte. "So what", you might say. In a synthesizer, does it matter if the capacitors are a few grams lighter? Probably not. No electrolyte is an advantage, but many synths are loaded with electrolytic caps and they seldom leak, and even if they do, it's usually not that big a deal.

Well, there's another, very good reason to prefer a tantalum capacitor over an electrolytic. Consider:

This is an interface circuit that I built for a home automation system. It's basically a 7555-based monostable timer circuit that, when activated, holds a relay open for a fixed amount of time. The little yellow blob at bottom center, just above the red connector with the blue and orange wires going to it, is a .68 uF tantalum capacitor. When I prototyped this circuit, I used an electrolytic capacitor of the same rated value. When I fired it up to check it out, the actual value of the cap as determined by the circuit's time constant worked out to about .35 uF -- unsatisfactory, because the timer needed to hold for 600 milliseconds and I was only getting about half of that. Electrolytic capacitors are not noted for being precision devices; the .35 actual that I got from the cap marked at .68 is within the typical tolerance for electrolytic caps, which are often specified as being -50%, +100%. And, electrolytics will lose value over time as the electrolyte evaporates; it isn't unusual to pull one out of a circuit after several years and find that it is only operating at 10% of its marked value.

Standard tantalums, on the other hand, are specified at plus or minus 10%, and you can get tighter tolerances at a somewhat higher price point. And solid tantalums will retain their operating value over a long period of time. (Wet-slug tantalums are still made for special applications, but don't waste your time with them.) The advantages for synthesizer circuits should be obvious: when the calibration of, say, a VCO or a VCF depends on a capacitor circuit, tantalums will make the circuit closer to the center of the calibration range at aseembly, and will retain their value over time, reducing the need for recalibration. That's why I went with a tantalum when I built the board above; the .68 uF marked cap got me as close to the 600 millisecond time value as I could easily measure (50 ms or so) without trimming.

The one other thing about tantalums: there is no such thing as a non-polarized tantalum cap. So keeping in mind that you need to respect the polarity, a bit of care is required in design. Do that right, though, and you'll have a more stable and reliable circuit. After all, tantalums are considered reliable enough by the aerospace industry that they are heavily used in aviation and spacecraft. So don't be afraid of the tantalum.

Electricity for Synth-DIY'ers: Transistors

2010-10-01T22:34:00.000-05:00

This is where we start to get into the more complex of the basic electronics components. Transistors are really the basis of modern electronics. It was the transistor, and its subsequent micro-miniaturization into integrated circuits, that really kicked off the electronics revolution in the latter half of the 20th century.

There are a number of different basic types of transistors. In this installment I'm only going to cover the bipolar transistor, which is still the most commonly used type. You are also likely at some point to run across another type, called the field effect transistor (e.g., JFET, MOSFET). Those work on a different physical principle and they have different characteristics. I'll cover them in a future installment.

What Does a Transistor Do?

Basically, a transistor is a device that controls a flow of current. This is the first thing to understand: the bipolar transistor is not a voltage regulating device; it's a current regulating device. A transistor has three terminals: current flows from one terminal to the other, under the influence of the third terminal. A small current flowing into or out of the controlling terminal controls a much larger flowing between the other two terminals. This is what allows the transistor to act as an amplifier: a small current can control a much larger one.

The three pins of a bipolar transistor are referred to as the emitter, the base, and the collector. The base is the control terminal; a small current at the base controls a larger current flowing between the emitter and the collector. Which direction these current flows go in depends on which of the two flavors of bipolar transistor you are dealing with; the two are known as PNP and NPN, based on their construction. If you remember the discussion from the diode installment of this series, a diode is a junction between two types of semiconductor, known as P-type and N-type. Same thing here: the PNP and NPN designations indicate what types of semiconductor material the transistors are made from, and by implication, which direction current can flow through them.

At this point we will introduce the symbology for transistors. The two types are denoted as:

The pin coming out to the left side of the symbol, perpendicular to the little bar, indicates the base terminal. The little arrow indicates the emitter, and of course the remaining pin is the collector. Note that in the symbol for a bipolar transistor, the collector and emitter are always portrayed as meeting the little bar at a 45-degree angle. If they aren't, then the symbol is for some type of FET transistor, not a bipolar.

As you may have figured, the direction of the arrow indicates whether the transistor is a PNP or NPN type. But what does the direction actually mean? This is where we introduce the first principle of the bipolar transistor: the junction between the emitter and the base acts as a diode. And, in accordance with the base pin being the middle pin in the transistor symbol, the middle letter in "PNP" and "NPN" indicates the type of the base; either of the other two letters can be regarded as representing the emitter. So, in a PNP transistor, the emitter is a P-type and the base is an N-type. If you remember from the diode discussion, this indicates that current can only flow from the emitter to the base, not the other way around. Accordingly, the arrow on the PNP symbol is pointing inward, to indicate that current goes in through the emitter. The opposite is true for an NPN transistor; since the base is P-type, current can only flow from the base to the emitter. So the emitter has to be a current exit, and so its arrow points out.

Do not, at this point, start confusing the current-in or current-out state of the emitter with where you actually obtain the amplified output signal! In both the PNP and NPN types, you can tap off of either the emitter or the collector to get the output, depending on the circuit design. We'll cover this further down. The arrow only indicates which direction current is flowing through the transistor.

So what about the base-collector junction? Does it also act as a diode? Well, no. Why not? To be honest, I don't know enough about the physics to give you a complete answer, but here's a simplified explanation: The base region of the transistor is very, very thin compared to the collector and emitter regions. So when the electron is attracted to the base region due to an opposite charge leaving the base, when it hits this region, instead of taking a 90-degree turn and heading laterally towards the base pin, it's easier to just pass through it to the other side. The collector region is "doped" differently so that it has much less of a tendency to pull charges away from the junction area when it is reverse-biased. So, although some electrons or "holes" go get gathered up by the base, most of them just go on through. That's what makes the transistor an amplifier. This is far from being a satisfactory explanation, but for a better answer, I'll have to refer you to someone who knows more about semiconductor physics than I do.

Transistor Circuit Basics

So, let's see what we've got so far. We have three layers of semiconductor material, with two junctions between layers of opposite types, one of which acts like a diode and one of which doesn't. For purpose of discussion, let's assume for a while that we are talking about an NPN diode. Now let's apply some voltages to some of the terminals. We'll start by applying 12 volts to the collector, and grounding the base and the emitter.

What happens? Nothing. Why not? Remember, the P-N junction between the base and the emitter has all of the properties of a diode -- including diode drop. Because of the diode drop, and the fact that we're talking about silicon, no current can flow from the base to the emitter until the base is at least 0.6V higher than the emitter. So now let's apply, say, 3V from a battery to the base.

Now what happens? Probably, the transistor burns up! Why? Because we're not thinking in the transistor's terms. The transistor isn't a voltage controlling device; it's a current controlling device. When we connected the battery to the base, we gave it an almost infinite source of current. The input impedence of the base of a bipolar transistor is very low. So right away a large current went into the base; this in effect threw the transistor "wide open" and a whole bunch of current flowed from the collector to the emitter. Even running wide open, the transistor has a small amount of resistance, and like any resistive device, there is a limit to how much heat it can dissipate.

So, after clearing the smoke and installing another transistor, let's re-think this and try again. The voltage into the base, as long as it's at least 0.6V above the emitter, doesn't really matter. What matters is the current going into the base. So let's put a potentiometer in between the base battery and the transistor base terminal. We'll also put in an ammeter in between the pot and the transistor base, and we'll put another one on the transistor's collector. I've prepared three short videos to illustrate some of thes principles; they use the circuit below:

We turn on the power and start turning the pot slowly clockwise, as in this first video:

What do we see now? At some pretty small value of current going into the base, we'll also see current starting to flow into the collector. As we increase the base current, the collector current will increase in proportion (up to a point, which I'll describe in a moment). The transistor is amplifying the current: as we put a current N into the base, we get a larger current X*N going into the collector. What is the value of X? That is, by definition, the transistor's current gain. When discussing bipolar transistors, the current gain is called the beta. (You may also see the terminology "Hfe".) The beta is a property of the design and construction of the transistor, pertubated somewhat by manufacturing variation. For the typical commodity small-signal transistors that we commonly deal with, the beta will generally be in the 50-250 range depending on the specific type. Fancier transistors, particularly power types, have gains ranging from the hundreds up to about 1000. The transistor used in the demo video has a beta of 200. So, for instance, when we adjust our pot for 50 uA into the base, we see 10 mA going into the collector. What if we used a PNP transistor? Well, we'd have to swap the voltages at the collector and the emitter, and apply a negative voltage so as to draw current out of the base rather than putting current in. But the operating principles are exactly the same. Only the arithmetic signs change.

Question at this point: what is the current coming out of the emitter? Well, all current that goes into the transistor has to come out somewhere. We have current coming in at both the collector and the base, and the only place where current is coming out is at the emitter. So it stands to reason that the emitter current is the collector current plus the base current: 10 mA + 50 uA = 10.05 mA. And if we had a third (accurate) ammeter on the emitter, we would see that this is true. Much of the time, when you are figuring currents through a transistor, you can disregard the base current and figure that the collector current equals the emitter current, since the base current is always a small fraction (according to the transistor's beta). Next question: what is the voltage at the base? Answer: it depends on the voltage at the emitter. As long as the transistor is flowing current, and isn't saturated ( in a moment), it will always maintain the diode-drop relationship between the base and the emitter, so if we are interested in the base voltage, we can always look at the emitter voltage, and the base voltage will be about 0.6V more than that. But the transistor is not really sensitive to the voltage at the base -- it's the current at the base that matters. You will often have to figure out the value of a resistor to put in series with the base.

Which brings up a point: you must always ensure that both the collector-emitter and the base-emitter currents are limited somehow. As we discussed earlier, the transistor will willingly pass enough current to blow up both itself and components connected to it. One way to limit the collector current, as shown by our example circuit, is to put the load (in this case, just a resistor) in series with the collector. However, you can also put the load in series with the emitter -- but the circuit will behave somewhat differently. Check this second video, in which we add resistance to the emitter load; as the resistance goes up, the base voltage goes up. This occurs because the pot plus the current-limiting resistance at the collector effectively puts the transistor in the middle of a voltage divider. As the voltage at the emitter goes up, the base voltage has to go up in order to maintain the diode-drop relationship.

Transistor Performance and Characteristics

Transistor performance is often shown in the form of a characteristic curve which plots collector current vs. collector-emitter voltage, for various values of base current. I don't find these to be terribly useful; the main thing they illustrate is that as long as the collector-emitter voltage is sufficiently large (typically 1-3V), the collector current depends only on the base current -- it doesn't depend on the collector-emitter voltage. But we knew that already. What we really want to see, for educational purposes, is a chart that details base current vs. emitter current. Here's a rough one that I assembled from taking measurements on the transistor I used in the video above. The green area indicates where the transistor is behaving linearly (constant beta), and the red areas indicate where it is non-linear. Bear in mind that the upper red area, in particular, depends considerably on the circuit that the transistor is installed in.

This illustrates a couple of things. The red "corner" at the bottom left is where the transistor is approaching cutoff; as in the case of the diode, this is where the base-emitter junction voltage is less than the diode drop and it is very non-linear. When the base voltage drops to or below the emitter voltage, the transistor will completely stop carrying current, except for a small leakage current which, for smaller transistors, is usually on the order of a few nanoamps. The other interesting red bit is the flattening out of curve as the collector current approaches 20 mA. This occurs as the transistor is approaching saturation -- a condition where it cannot carry any more current because there are no more charge carriers available at the junctions. Why are there no more charge carriers available? Because, in the case of the video above, the circuit that the transistor is in can't source any more current. Note that this is not a limitation of the transistor itself; it's a limitation of the circuit that the transistor is in, and specifically the resistance that I put in series with the collector. The 500 ohms resistance between the battery positive and the collector limits the maximum possible current to about 20 mA.

In between the cutoff and saturation regions is the active region, shown by the green portion of the graph. In the active region, the beta remains nearly constant, and so the transistor's response to base current is nicely linear, as shown in the center portion of the above graph. When you are using the transistor as an amplifier, this is what you want. On the other hand, the cutoff and saturation regions have their uses too.

Transistor Applications: Switching vs. Amplification

The two most common uses for a transistor are switching and amplification. Let's cover the switching case first since it is easier to understand. In a switching application, we want the transistor to, at any given time, be in one of two states. The "off" state obviously corresponds to the transistor's cutoff region: we kill the base current, and the collector current goes to near zero. The "on" state can be a little trickier. It's common to operate a switching transistor in saturation; in this state, it has a very low resistance (provided that the collector current isn't excessive). A common rule of thumb for switching circuit design is to figure out what base current is needed to achieve the necessary collector current, and then double it. However, in doing so, you have to make sure that the transistor's limits for base current and power dissipation aren't exceeded. Here's an example of a common use for a transistor circuit: the transistor is being used to drive a relay. (The relay could, for example, be switching a high-voltage circuit on or off. A typical example in the audio world would be that the relay might be controlling the high-voltage plate supply in a tube amplifier.) Assume the following:

The relay coil has an impedence of 250 ohms
It takes 20 mA of current through the coil to make the relay close
The transistor has a beta of 100
The supply voltage is 12V

When the switch is closed, the base resistor feeds current into the base of the transistor. This is where the fun starts: how do you analyze this to know how much base current you need? Start by doing an Ohm's Law calculation: in order to to push 20 mA through the 250-ohm coil, you need to apply at least 5V to it. So we know it's doable, provided that our 12V power supply can source at least 20 mA of current without drooping. Now, in order to pass 20 mA of current through the transistor, how much base current do we need? Dividing that current by the beta gives us 0.2 mA, or 200 µA, of base current. If the switch is also connected to the 12V supply, then in order to put 0.2 mA into the base, we need a 60K ohms resistor in series with the base. (Note in this case that because of the base-emitter diode drop, the voltage at the base will need to be at least 5.6V if the relay coil is in series with the emitter. If we only had 5V being supplied to the switch, that won't work; the transistor will remain cut off. If this is the case, you have two options: (1) supply a higher voltage to the switch, or (2) put the relay coil in series with the collector and ground the emitter.)

Now, the thing about transistors is, the beta of specific units varies a lot from unit to unit across a given part number. So if our transistor is spec'ed for a nominal beta of 100, the beta of the particular unit we have might only be 70 or so. There are two approaches to this. One is that we can test the individual transistor that we put into the circuit and make sure its beta is at least 100. However, if we are manufacturing in quantity, it will take extra time to do the testing, and cost more because we'll have to discard some percentage of the transistors that we buy. The other approach is to provide for the minimum expected beta of a particular unit. In a switching application, this is pretty easy; if we reduce the base resistor to 30K, then the transistor beta can be as low as 50 and it will still work, but the base current is still low enough to not cause damage.

Well, assuming that your transistor is operating in the active region, it's pretty easy: choose the resistor that results in 2 mA of base current, and you'll get 200 mA out. But how do you know what resistor value gets you 2 mA of base current? And how do you know for sure that you are going to be operating in the active region? After all, that depends on the collector-emitter current.

Historically, though, amplification has been the most common use for transistors. This is not quite so true in audio applications today; opamps have taken over most of the functions of amplification within audio devices because they are easier to design with. In audio-frequency applications, discrete transistors remain mostly in power amplifiers. Nonetheless, we'll look at two methods of using transistors as amplifiers.

The first type of circuit uses a single transistor. As you might guess, the first problem that one faces when using a transistor to amplify alternating signals is that the transistor can only flow current in one direction, which means that if you just feed an AC signal into the base, at best only half of the signal will get amplified -- the other half will be absent from the output, because the opposite-polarity voltage applied to the base drives the transistor into cutoff. The way to solve this problem is to bias the input signal -- that is, add an offset current to it so that the transistor stays within the active region throughout the input signal range. This drawing illustrates a common way of doing this:

This is an example of a single-transistor amplifier driving a loudspeaker. The resistors obviously constitute a voltage divider, but it can also be thought of as a current divider. It sets the "operating point" of the transistor: the amount of current that enters the base when the input signal is quiescent, which is proportional to the voltage drop in the middle of the divider. Assuming that the input signal will on average be symmetric, the operating point needs to be at the midpoint of the transistor's active region. The input also needs to be scaled, per the transistor's approximate beta, so that a maximum input signal will not drive the transistor into saturation, and a maximum negative signal won't drive it into cutoff. Since the output will have a DC offset, the usual practice is to couple it into the following circuit via a DC-blocking capacitor (not shown here).

This single-transistor circuit is pretty common in circuits which handle small currents, but for large-current circuits, there are better methods. The two-transistor "totem pole" is a commonly used circuit; it is nearly universal in audio power amplifiers. In this configuration, a PNP and an NPN transistor are used together, with either their collectors or their emitters tied together, and the output taken from that node point between the two. One transistor amplifies the positive half of the signal; the other transistor amplifies the negative half. In Class A operation, the transistors have their operating points set so that neither of them ever goes into cutoff. A given level of input signal causes one transistor to conduct more current while the the other one conducts less; the difference is what appears as the output signal. When the input signal is quiescent, an equal current flows through both transistors. Well designed Class A amps are noted for their low-distortion operation, but they are very inefficient at low signal levels because a lot of "wasted" current is passing through the two transistors.

In Class B operation, the transistors have their operating points set so that they both enter the cutoff region precisely at the point where the input signal is zero. Note that some bias is still required in order to overcome the base-emitter diode drop; otherwise, there would be a "hole" in the response at very low input levels. It's harder to design a Class B amplifier for low distortion due to the fact that the transistors are approaching the non-linear cutoff region, but a Class B amp is far more efficient than a Class A -- at a zero input level, the Class B amp draws little current. There exist designs that are hybrid of these two concepts, known as "Class AB"; they set the operating points so that one transistor cuts off when the other is well into its active region. These are a compromise between distortion and efficiency.

Emitter vs. Collector Output

There are two basic ways of obtaining the output signal from the transistor, which we've already touched on, but they need to be described more explicitly. One is the common emitter configuration; in this configuration the emitter is connected to ground or to a power supply, and the output signal is obtained at the collector. If the output signal is something that requires a lot of current and has some resistance, such as a relay coil, the easiest way to do this is to put the load in series with the collector.

However, this configuration can also be used to derive a voltage signal from the input, provided that the load doesn't draw much current. The way to do this is to put a resistor in series with the collector (which you often will do anyway, to limit the maximum current) and then tap a point between the resistor and the collector. This effectively makes the transistor behave as a variable resistor in a voltage divider, and the output comes from the middle of the divider, as such:

Note a couple of things about this. The first is that the signal output is inverted from the input; the output voltage will be at its maximum when the base current is zero, and vice versa. The second thing to note is that the swing of the output voltage is determined by the maximum collector-emitter voltage, and so it can be of higher voltage than the input to the base. So in this mode, the transistor is capable of voltage gain in addition to current gain. This is illustrated by the video below:

The other configuration is usually called the emitter follower. In this configuration, the signal is taken from the emitter:

You can, if you are careful, design an emitter follower circuit with no current-limiting resistors. This configuration provides the maximum possible power within the limits of the transistor, which is why it is frequently used in power amplifiers.

Darlington Pair

You can use a pair (or more) of transistors to obtain higher gain, by driving the base of a second transmitter from the emitter of the first one, like so:

This is known as a Darlington pair. Where the name comes from, I don't know; I've often wondered if it has anything to do with the notorious speedway in South Carolina. The total beta for the Darlington pair is B1 x B2, where B1 and B1 are the betas of the individual transistors. As you can see, you can obtain quite high gain levels this way. Darlington pairs can be purchased as pre-made parts, or you can of course make your own from individual transitors.

Other Useful Transistor Circuits

You may recall that back in the diodes chapter, we presented a simple regulated power supply circuit controlled by a zener diode. The problem with that circuit is that the zener itself has to dissipate all of the power not drawn by the load at any moment. Here's an improved version:

In this circuit, the zener only has to dissipate the very small base current. The regulated voltage will always be 0.6V less then the zener's voltage rating, thanks to the B-E diode drop. The transistor has to pass all of the power supply current, but you can get power transistors that can handle substantial current.

The next circuit is called a current mirror, and it has a number of uses:

The way it works is that the resistor on the left allows a specific amount of collector current to flow through the transistor on the left, and its connection to both of the bases will cause the same amount of current to flow through the transistor on the right. This means that a fixed amount of current will flow through the load (within the limts of the power supply). The two transistors must be of the same part number and be matched -- that is, tested to make sure they have the same beta value. The circuit is useful for any situation where a constant-current supply is needed. Replacing the resistor on the left with a varying load will cause the two loads to always see the same amount of current.

A similar-looking circuit is the differential amplifier:

This circuit amplifies the difference between the inverting and the non-inverting input. If both inputs change by the same amount, the output does not change. This is frequently used in pro audio applications where "balanced" lines are used, e.g., to send microphone signals long distances to a mixing console. Noise that tries to enter the line will enter both inputs in equal amounts (called "common mode" noise) and the amp will reject it. It is also frequently used in sophisticated amplifier circuits where negative feedback is used to stabilize the circuit.

Transistor Numbering, and Packaging

Three main packaging styles are used for transistors these days. "Small signal" transistors are generally found in what is know as a "TO-92" package:

It's a small cylindrical plastic package, nearly always black. One side is flat for orientation, and circuit board designers often use a flat-sided circle on the board silkscreen to indicate which way the transistor is to be inserted. Generally, as you look at the flat side, the order of the pins is emitter/base/collector, but not always. So be sure to check the data sheet for the specific type that you have. Usually, the part number will be printed on the flat side. Not pictured is the TO-18 metal can, which is the same size as the TO-92 but does not have the flat side. A small piece of metal sticks out from the edge of the can to identify the emitter pin. The TO-18 version is not much used anymore, but very common in electronic devices made prior to 1985 or so.

Somewhat higher-power transistors will usually be found in a "TO-220" package:

Photo courtesy of the University of Michigan Smartsurfaces Project

As in the case of the TO-92 package, the order of the pins may vary. The metal tab is intended to be mechanically fastened to a heat sink, to increase the current carrying capacity of the transistor. The tab is often electrically connected to one of the pins, usually the collector, so electrical insulation may be required between the tab and whatever it is fastened to. As usual, check the data sheet.

Serious power transistors come in a "TO-3" case:

Photo from Wikipedia Commons

These are most commonly found as output transistors in audio and RF power amplifiers, and switching transistors in solid-state power controls. Not visible in the photo are two pins that stick out from the bottom side, which are the base and emitter pins; the case itself is the collector. These are made to be inserted into a special socket, and they are usually in contact with a live heat sink (insulated from ground) which is also part of the collector circuit. Transitors this large have a number of non-ideal properties, such as high leakage current and parasitic capacitance.

There are three main systems for part numbering of transistors. As in the case of diodes, there is a JEDEC system for transistor numbering. While diode JEDEC numbers all start with "1N", transistors start with "2N". This will be followed a a three- or four-digit number, and possibly additional letters to identify variants. In general, higher numbers are more recent designs.

Many transistors made in the Far East use the Japanese Industrial Standard (JIS) system. In this system, numbers for PNP types start with "2SA" or "2SB", and NPN types start with "2SC" or "2SD". A European system called Pro-Electron also exists. In this system, the first character is "B" to indicate a silicon-based device. A second letter of "C" or "D" indicates an audio-frequency transistor; "F" or "L" indicates a radio-frequency transistor, and "U" indicates a power switching transistor. Here is a good Web page that summarizes these systems.

Transistor Conventions and Standard Parts

You may have noticed that all of the circuits I've presented in this post have all used NPN type transistors. There is some history for that: in the 1960s when transistors first came into use, for reasons not clear to me, NPN types were a lot easier to make -- and therefore a lot cheaper -- than PNP types with equivalent characteristics. For that reason, designers figured out how to build many common circuits using only NPN types. This persists somewhat in electrical engineering; in situations when the choice between designing a circuit to use NPN or PNP transistors is an arbitrary choice, designers will usually go the NPN route. A side effect of this is that when engineers discuss the characteristics of various transistors, they will generally talk about the NPN type as being the standard, and then just note whether or not an equivalent PNP type exists. PNP types aren't often discussed for their own sake.

In synthesis circuits, we don't typically deal with large currents, so we usually stick to small-signal transistors. (This will change if you get into building amplifiers.) Doing a quick survey of small-signal transistors used in the most common published synth-DIY circuits, I note these three commonly used part numbers:

* BC547. A very useful and inexpensive NPN small-signal transistor, with a maximum collector-emitter voltage of 45V and maximum collector current of 100 mA. A Fairchild data sheet specifies beta as being in the range 110-220 for the "A" version; there are "B" and "C" versions with higher beta ranges at somewhat higher prices. As of this writing, Mouser has the "A" version listed for a whopping five cents USD apiece, quantity 10. They come in the TO-92 plastic package. The BC557 is a complementary PNP type.

* 2N3904. A similar part to the BC547, with max collector-emitter voltage of 40V and a maximum collector current of 100-200 mA depending on which version you get. Beta is in the 100-300 range; here's a Fairchild data sheet. It comes in a TO-92 package and a variety of surface-mount packages. It's a little faster than the BC547, which generally won't matter in synth applications. The main thing you have to watch is the max emitter-base reverse breakdown voltage of 6V. Mouser is quoting them at five cents USD apiece, quantity 25. The 2N3906 is a complementary PNP type.

* 2N2222. The easy-to-remember "all twos" NPN transistor has some virtues over the above, mainly that it can handle more collector current: 500-600 mA for most versions. It is still available in the TO-18 metal can as well as the TO-92 package; the former is good for applications where you need to thermally couple it to something (e.g., the expo converter circuit in a VCO/VCF), and some people claim that the extra parasitic capacitance of the TO-18 can makes it sound a bit better in audio applications. Here's an ST Microelectronics data sheet; collector-emitter voltage is 40V and beta is in the 100-300 range. As in the case of the 2N3904, you have to watch the 6V base-emitter reverse breakdown voltage. Main disadvantages are the somewhat higher cost (cheapest one I see at Mouser is $0.54 USD, quantity 25), and the fact that there is no exactly complementary PNP type.

On To the Next

As you can see from the date of my last post, it's taken me a long time to get this one together, for various reasons. This is the end of the series on discrete parts; next, we'll swing into integrated circuits with a discussion of operational amplifiers -- op amps.

Review: Encore Electronics Frequency Shifter

2010-08-05T21:15:00.000-05:00

One of my recent acquisitions for the Discombobulator, as documented in the previous post, is an Encore MFS01 Frequency Shifter, in MOTM format. Encore also offers this in Frac and Euro formats; the MOTM-format unit had been out of stock for some time, but this year Encore has been doing new runs of its MOTM-format modules. I received mine a couple of months ago, but I didn't have a place to install it until I built the new block that I documented in my previous post. So I'm just now getting to play with it.

A Frequency Shifter is Not the Same as a Pitch Shifter

So what's a frequency shifter? Well, I had a bunch of material that I wrote to address all that, but I've decided to save it for a follow-up post. To keep it short, a frequency shifter is sort of like a pitch shifter, but it does not maintain the harmonic or musical relationships between the various tones and sounds that make the input. What's that good for? Well, for one thing, it's great for bell and chime sounds, and it behaves a lot more predictably than a ring modulator in doing that job. It can do flanging-like effects that range from subtle to startling. It can do creation of sounds that play in unusual intervals and scales. But if you want to completely brutalize a sound, rip it apart and then glue the pieces back together like a ransom note, a frequency shifter is what you want.

Why Frequency Shifters are Usually Expensive

The classic frequency shifter is the one developed by Harald Bode in the early '70s and licensed to Moog. They were sold both as modules for Moog modulars, and as stand-alone devices. They were as renown for their sound as they were notorious for their price tag -- they sold for about $1000 in 1975 dollars. Obviously that put them out of reach of most musicians, and so not that many were made. They are quite rare and valuable now.

One of the reasons the Bode frequency shifter was so expensive was the sheer amount of circuitry they contained. There are several hard problems that a frequency shifter design has to address. One of these is the problem of generating two sine-wave carrier signals "in quadrature", that is, identical in frequency but separated in phase by 90 degrees. The Bode design was noted for its ingenious approach to this, which produced a very clean pair of carrier signals, but it was costly to implement.

With the benefit of three decades of subsequent technology development, Encore was able to take a different approach to this problem, using an option that wasn't available to Bode: go digital for the carrier generation portion. Encore incorporated a microprocessor that generates the quadrature sine waves from (I presume) an internal look-up table. This makes it easy to maintain the phase relationship while responding to panel controls and control voltage, and it takes a lot less circuitry than the Bode design. Thanks to this, Encore is able to offer their frequency shifter at a lower cost ($399 USD) than competing models from Modcan (the Modcan 39B sells for about $1050 as I write this) and Club of the Knobs (the COTK 1630 lists for E950, about $1250 at the moment). Note that the audio signal path is still all analog -- only the generation of the carriers is digital.

(In fairness, it should be noted that Modcan has two models, the all-analog 39A/B, and the all-digital 65B, which is a dual unit. The 65B currently list for $770.)

The Panel and Controls

With that, let's take a detailed look at the panel. Here's the top portion:

The large Initial Shift and the smaller Fine Shift knobs together set the amount of frequency shift. The Fine Shift knob has a range of about 150 Hz in either direction. The Initial Shift control has a range of approximately 3500 Hz; on the unit I have, all of the action takes place between, roughly, the -4 and +4 positions of the knob -- the 4-to-5 areas have no effect. The Initial Shift knob has a deadband at the zero mark that is useful in using the Fine Shift knob to achieve small frequency shift settings. However, the deadband is also rather disconcerting because there is a jump of about 100 Hz when the knob is moved off of the deadband. To do shifts of less than 100 Hz, you must center the Initial Shift knob on the deadband and then use the Fine Shift.

The Input Gain control attenuates the input signal to the frequency shifting circuits. Note that said signal consists of a mix of three things: the signal from the input jack, and the signals being fed back to the shifting circuitry by the Up Feedback and Down Feedback controls. When using the feedback, you'll find that you have to turn this down some to avoid overloading the input. The red LED next to the knob indicates clipping. The Frequency CV attenuates the signal being fed into the frequency control voltage jack.

Below is the lower half of the panel:

We'll start with the jacks. The Input jack is obviously the input for the audio signal to be processed. The CV jack accepts a control voltage (range +/- 5V) which controls the amount of frequency shift, along with the Initial Shift and Fine Shift knobs. Note that response to control voltage is linear, at a rather measly 100 Hz/volt with the Frequency CV control on 10, so you can't do huge sweeps with the CV. The Up Out and Down (DN) Out jacks are the two outputs from the shifter. The Down Out output responds to the reverse of the shift controls and the CV; in other words, when the Initial Shift knob is turned to the right, the output at the Up Out jack increases in frequency, but the output at the Down Out jack decreases in frequency. Note that the signal present at both of these jacks is 100% "wet"; there is no provision for mixing in any of the unprocessed dry signal. If you want that, you have to use an external mixer.

The Up Feedback and Down Feedback knobs feed some of the output of the Up Out or Down Out jacks, respectively, back to the input. Be careful with the Down Feedback since it can create a "ping-pong" resonance in the circuit which can result in runaway feedback if the input signal hits a resonant frequency. If this occurs, you have to turn the input trim control down to 0 to clear it, which could be embarrassing in live performance.

The device makes available the two sine-wave carrier signals at the Sine Out and Cos Out jacks. The two knobs control the signal level present at these jacks. These two signals will always be at the same frequency, which is determined by the amount of frequency shift (which means their frequency is also effected by the frequency shift CV). The cosine signal leads the sine signal by 90 degrees when the frequency shift is positive; the opposite is true when the frequency shift is negative. The two lights next to the knobs light when each signal is near its positive peak; at low frequencies, they provide visual indication of both the frequency and the direction of the shift.

Testing and Demonstration

I started off doing some tests with some simple waveforms; here is a clip, which I'll refer to as we go through the description. (The clip is an uncompressed WAV file, to avoid any MP3 artifacts that might be triggered by the unusual timbres.) First, I fed a sine wave into the input and tried various frequency shift settings, starting with the fine shift. The pitch of the sine wave goes up like you would expect it to, within the range of the fine shift control; this is at 0:07-0:25 on the clip. Then, I advance the Initial Shift as far up as it will go. This is at 0:33-0:56.

Next, I move the Initial Shift into the downward range. Going the other way, something interesting happens which is characteristic to frequency shifters. As you turn the knob left of the zero mark, the pitch gets lower and lower, goes through bass and subsonic -- and then starts going up again. What's happening is that the shifter has actually taken the input through zero Hz, and it is now outputting a "negative" frequency. As it happens, a negative frequency sounds the same as the corresponding positive frequency; the output signal is inverted, but you can't hear that in isolation. But you notice the difference as the source goes up and down: when the frequency of the source decreases, the frequency of the output increases, and vice versa! It's called frequency reversal, and it makes sense when you look at the math: if the source is at 400 Hz, and the frequency shifter takes it 1000 Hz in the negative direction, the resulting frequency is -600 Hz (which sounds the same as positive 600 Hz). If you lower the frequency of the input, it's going towards zero and the subtraction is moving further away from zero, taking the absolute value of the output higher. If you lower the input to 100 Hz, now the output is at -900 Hz. This is the much-talked-about through-zero operation, and it's something that conventional pitch shifters can't do. In the clip, it starts at 0:56; it goes to subsonic and starts into the negative frequency range at 1:00. From 1:10 to 1:18, I twiddle the frequency knob on the source VCO, and the frequency coming out of the shifter responds opposite to it; this is frequency reversal. If you do this with a signal containing a mix of tones, the higher tones will be lower than the lower tones in the frequency-reversal region, which can do some truly bizarre things to natural sounds like voices and animal noises.

In the clip, you can hear a subtle change in the timbre of the sine wave as I manipulate the frequency shift. Now, a sine wave should not have its harmonic content effected by the frequency shifter since it theoretically doesn't have any harmonic content; it should just go up and down in pitch, with no noticable change in timbre. That isn't quite what happens: as I went up in frequency, some odd sub-harmonic tones started to appear. They weren't very loud, but they were audible. I'm not sure where they are coming from. These also appeared as overtones when I went into the negative frequency range. Perhaps the module is picking up electrical noise from something else in the room. Another possibility is that the module's carrier suppression is not quite perfect, and the carrier or some overtone of it is leaking into the output. It could also be that the sine-wave output on the VCO that I used as the source (a Dotcom Q106) is not absolutely pure; that would not be unexpected, since producing pure sines is a very difficult thing for a VCO to do. (I did also try the fixed 440 Hz generator from a Q125 standards module. Although the sine wave from that is noticeably more pure, the artifacts were much the same. So I'm thinking that some of the artifacts are carrier leakage.)

From 1:24 to 1:45, I turn up the Initial Shift again, and then bring in some Up Feedback, and then from there until 2:09 is the Down Feedback. You can hear the effects of these. Be careful when you listen to the latter part; it hits a couple of of the aforementioned ping-pong resonances that cause level spikes.

Next, I ran a sawtooth wave through the frequency shifter, with far more dramatic and impressive results. Shifting the sawtooth wave by small amounts in either directing results in beating and interference patterns as the harmonics are moved off of the integer relationships; the waveform starts to intermodulate with itself. You can hear this in the Fine Shift demonstration from 2:29-2:48, A large shift upwards transforms the sawtooth wave into something distorted-sounding, eventually ending up sounding a bit like it's overdriving a filter with the resonance turned way up. This is in the clip at 2:50-3:10.

Negative frequency shift is even weirder; as the frequency goes through zero, first the fundemental and then the harmonics all go through zero and get inverted, and you can hear them doing it one at a time as the Initial Shift knob is gradually turned more to the left. And the timbre is just strange. This is in the clip at 3:11-3:35. The really interesting bit here is when the source frequency is varied while in the negative-frequency range. The effect is hard to describe; some harmonics go up in frequency, some go down, and the only words I can come up with is that it turns the sound "inside out". That makes no sense, so you just have to hear it, at 3L36-3:46. The up and down feedback controls were very successful at destroying the waveform, creating a bunch of noises that sounded like various forms of exotic radio broadcasts, or perhaps modem noises. Up feedback is demonstrated at 3:54-4:12, and down feedback is at 4:18-4:35.

Interestingly, through most of the things that I did with the sawtooth, there seemed to be a bit of the unaltered input waveform leaking through. I wondered about that, since I didn't hear it with the sine wave. I think it might be this: a sawtooth wave has a portion of the waveform that is basically an impulse -- on the scope it's nearly vertical. Fourier analysis tells us that an impulse or spike is a waveform of indefinite frequency; in theory, a perfectly vertical spike (impossible to achieve with finite bandwidth) contains every possible frequency. For this reason, I think the frequency shifter simply couldn't do anything with that part of the waveform; the math breaks down. The result is that it sounds like an extremely narrow pulse wave, at the pitch of the unaltered input, is riding through the circuit. That would explain why I did not hear it with the sine wave; a sine wave does not have a steep slope anywhere in its waveform.

Some Usage Ideas

So what can you do with the Encore frequency shifter? Here are a few examples. One use is to create phasing/flanging/stereo simulation effects. Connect a mono signal to the input. Center the Initial Shift knob in the deadband and set both of the feedback knobs on zero. Take a stereo out using the up out as one channel and the down out as the other. Or, for a better enhanced effect, run the up out and down out to an external mixer. Mix the up out with some of the dry signal and pan that hard left; then mix the down out with some of the dry signal and pan that hard right.

A frequency shifter is of course the bees' knees at creating bell, chime, and other clangorous timbres. You'll have the best luck with sounds that don't contain a lot of closely spaced harmonics. I actually had good results feeding FM noises into it. Treat with the appropriate envelopes, and you've got the bells of doom. The nice thing about it is that the apparent pitch of the resulting sound is pretty predictable, much more so than if you use a ring modulator to create bell sounds. So you can make it sorta kinda play in scale. I managed to do some things that played more or less in scale over an octave. Precision tuning it won't do, but that isn't what you get a device for. If you want the strange sound but you also want it to play in tune, sample it; the circuits are very stable and will hold a specific timbre while you set it up for sampling.

However, I found that what this unit does best is brutalize sounds. There are a million ways to make it distort a sound -- but in a completely different way from what a clipping or fuzz circuit does. It excels at creating sounds that have a lot of closely spaced overtones with weird quasi-random beating and pulsing going on. And when you get into the frequency inversion regime, the results are indescribably weird.

The sine and cosine carrier outputs can be used usefully as control voltages for various purposes. For example, you can do a simple rotary-speaker effect by feeding each one to the CV input of a VCA. Feed the same audio into both VCAs, and then take their outputs to a stereo mixer panned left and right. The shift controls will control the speed of the apparent rotation, in either direction.

In short, this isn't a pitch shifter. You won't use it to correct clam notes in a track. You will use it to make bizarre and other-worldly sounds. Getting it to behave predictably may be a bit of a struggle. But if all of life was predictable, what fun would that be? The opportunities for serendipitous discovery are huge here. Run stuff through it, turn the knobs, and see what happens. Then, build a patch that uses that. You may surprise yourself.

Discombobulator Block 5: Tethys

2010-07-27T13:38:00.000-05:00

Presented with appropriate mood lighting: the long-awaited Block 5 of the Discombobulator, Tethys!

The currently installed complement consists of:

Three Synth Tech MOTM-310 micro VCOs
STG/Soundlabs Mankato Filter
Encore Electronics MFS01 Frequency Shifter
Cynthia StereOSpace
Synth Tech MOTM-890 micro mixer

Here's the usual sequence of build photos. I finally used up all the scrap 3/8" plywood (left over from a years-ago repair job at our previous house) that I've been using to build the bases of these things. I decided I wanted something a bit sturdier anyway, so I got some 5/8" plywood. Here, the rail pieces have been cut and are laying on the base, and the front rail has been glued in position and is clamped on while the glue sets.

All of the rail pieces are in place except for the top rail. The two rear posts (to hold up the top) were cut from a piece of dowel that I had laying around from some long-ago job.

After building the last block (Iapetus), I swore that I'd use its power supply configuration from now on. So of course I changed it again. However, the reason I did so was because I acquired a used Power One triple-voltage supply from a regular poster at Muff's for a very reasonable price. The Power One supplies are functionally and physically interchangeable with the Condor supplies that I usually use. Both are considered good brands; I usually use Condor because that's the brand that Mouser carries.

Here's the supply and the MOTM-990 power distribution board, in place but not yet mounted on the base. It was a little tight, and I wanted to get all the wiring connected before I screwed them down. The supply came with leads already soldered on the +/-15V side, and a power cord installed on the transformer. Unfortunately, someone had used a shielded cable for the power cord. Never ever use shielded cable for power! I had to remove that and make a new one out of zip cord.

Here is the completed power supply installation. The +/-15V comes from the part of the supply to the right of the transformer; the last owner had configured it for 12V, and I had to track down and remove two soldered jumpers to convert it to 15V. Fortunately, the instructions for how to do so are printed on the back of the chassis. The +5V comes from the smaller board to the left of the transformer. Everything is wired to the MOTM-995 distribution board, which has connectors for both MOTM 4-pin and 6-pin standards. The 6-pin connectors have the +5V and can be used to power a Dotcom or similar module, by constructing an adaptor cable. The line cord comes in at the bottom left of the picture and has an in-line AGC standard fuse holder on the hot side. I fuse these at 1A.

Some close-ups of the installed modules. First, the bank of MOTM-310 VCOs. I bought these as a package; I intend to use them for FM experimenting.

The Mankato Filter. This is an unusual filter, designed originally by Thomas Henry, that produces different response characteristics depending on which of the output jacks in the big circle you plug into. When the resonance is turned up to self-oscillation, it becomes an 8-phase sine wave VCO; each output jack is 45 degrees advanced from the previous one.

The Encore Frequency Shifter. Unlike a pitch shifter, a frequency shifter shifts each partial in the input signal by the same amount, in terms of Hz. This means that, unlike the pitch shifter, it does not maintain the harmonic relationships that are present in the input signal. Frequency shifters are complex circuits and are usually very expensive; Encore figured out how to build a less expensive one by using a microprocessor to generate internal control signals. It is capable of doing quite brutal things to a signal! The MOTM-formatted version was out of production for a long time, but when Encore announced they were doing another run of them early this year, I jumped on it.

I'll have reviews and sound samples of these coming up over the next two weeks.

Statescape Mississippi

2010-07-16T23:33:00.002-05:00

Jeez, have I actually not posted anything since April? Anyway, there is a new Statescape, Mississippi, here. There's some info about it on the Web site. I'm building the long-delayed block 5 of the modular; I'll try to get a post up about it this weekend.

Function Generators

2010-04-29T22:09:00.001-05:00

The idea for this post started from a thread on VSE, talking about the use of test equipment as electronic music instruments. There is, of course, a long history of such, actually pre-dating the invention of synthesizers per se. Significant parts of early electronic themes and soundtracks, such as the "Doctor Who" theme and the soundtrack to Forbidden Planet, were realized using laboratory electronic test equipment such as oscillators, RF modulators, and sweep generators.

The function generator I have is a Hewlett-Packard 3312A, one of the company's last analog models. (Note that the former Hewlett-Packard test instrumentation division is now owned by Aglient.) Its controls and capabilities are pretty typical for its era; I think the one I have was built around 1980. It is a solid-state, purely analog device. Here is a look at the front panel:

The front panel is divided into two sections: the main generator section, and a modulation generator section. The left two-thirds of the panel contains the controls for the main generator:

The big knob on the left is the frequency control, which moves through a range of 0.1 to 1 to 13. The big plastic dial is part of the knob and rotates with it. The frequencies available are chosen by pressing one of the nine decade buttons across the top, which provide ranges (with the frequency knob set to 1) from 0.1 Hz to 1 MHz. The knob multiplies the decade setting, so for example, if the knob is at the 4 position and the 1 KHz button is pressed, the output frequency is 4 KHz.

The signal generator can produce one of three basic waveforms: sine, square, and triangle. The FUNCTION buttons at the top right select the waveform. There are several controls that can modify the waveform. The first and most basic is the amplitude control. This consists of two nested knobs; the inner knob is a rotary switch that selects a peak-to-peak voltage range of 10 mV, 100 mV, 1V, or 10V. The outer knob is a vernier control that varies from 0 to the selection of the inner knob.

The symmetry control, to the right of the amplitude control, varies the generator's time base such that other derived waveforms can be produced. The knob is only effective when the blue "CAL" button in the middle of the knob is out; when it is pressed in, the waveform is symmetrical and the knob's position is ignored. It effects all three waveforms. When square wave is selected, the symmetry knob serves as a pulse width control. For the triangle wave, the knob at its extremes produces ramp and sawtooth waves. Here's an example of the symmetry being varied from triangle to ramp:

Audio -- click here

With sine selected, it produces various distortions of the sine wave; turning it all the way to either extreme produces a waveform that moves through half of a sine wave and then jumps back to the starting point. Example:

Audio -- click here

I noted the pitch variations caused by the use of the symmetry knob. As I understand the circuit (I still need to look at it some more), the generator core is a triangle-core VCO, with separate integrating capacitors for the positive-going and negative-going halves of the waveform. The symmetry control works by varying the charging current for the two capacitors; it increases one to speed up that half of the waveform, and decreases the other to compensate so that the overall wavelength, in theory, remains the same. Apparently that process isn't perfect, or perhaps my unit is just in need of calibration.

The offset knob adds a negative or positive DC offset, when the blue CAL button is out. Offset can range up to +/- 10V, but the manual cautions that if it causes the waveform to exceed 10V in either direction, clipping and possible damage to the output circuit will result. Pressing the CAL button in cancels the offset. The phase knob is used to produce non-continuous waveform bursts. I'll get to that in a minute; for now, note that it has to be in the "free run" position in order for the generator to run.

The waveform emerges from the jack under the amplitude control. Note that all of the jacks on this unit are BNC (bayonet) jacks, which are common in test equipment but not generally used in synths or audio production, so you'll need an adaptor. Also note that the settings on the amplitude control are rated for connecting the output to a 50-ohm load. Most synths and audio equipment have a much higher input impedance than this, which means that the peak-to-peak output voltage will be higher than indicated by the knob position. You'll have to tweak the amplitude control to get the output level than you want, and avoid clipping. The jack to the left, marked "sync", outputs a square wave whose trailing edge is at the positive-going zero crossing of the main output waveform. This can be useful for syncing other oscillators or sequencers.

The area occupying the right 1/3 of the panel is the modulation section:

The modulation section contains all of the circuitry needed to do amplitude modulation (AM) or frequency modulation (FM) on the main generator's waveform. It also contains its own low frequency oscillator to use as a modulation source. The three buttons at the upper left allow the selection of AM, FM, or sweep (labeled SWP) generation. These are not radio buttons; they are individual on-off switches, and any combination can be selected. To the right of those buttons, the next three allow selection of the LFO waveform: sine, square, or triangle. A concentric control underneath the waveform selection buttons controls the LFO frequency: the outer knob is a four-position rotary switch that can select a range of up to 1 Hz, 100 Hz, or 10 KHz. (The 0 position is for setting up the sweep function; it "freezes" the LFO at a zero crossing point.) The inner knob is a vernier that allows selection of the desired frequency within the selected range.

The knob directly underneath the modulation selection buttons (the one with the little arrow) controls the amount of modulation. The modulation generator has its own symmetry knob, whose CAL position is a click-stop at full counterclockwise rotation. The BNC jack in this section is both an input and an output; when the internal LFO is in use, its waveform is output from this jack. When external modulation is selected (by partially depressing and then releasing a waveform selection button, so that all three of the sine, square, and triangle buttons are out), it is input via this jack.

The AM is powerful and useful; both the modulation generator and external modulation can achieve 100% modulation of the generator output (from 0V output to maximum output). AM is a very useful capability which is seldom found on commercial synths for some reason. Here's an example of a triangle wave being AM modulated with a square wave:

Audio -- click here

The FM mode, on the other hand, is limited to +/- 5% of the carrier frequency. Audio FM synthesis usually requires far more modulation than that, so this is not very useful from a musical perspective. (There is another way to do it, which is described further down.) Example of a sine wave, FM'ed with a square wave (the photo is a multiple-trace exposure):

Audio -- click here

The "sweep" modulation mode is one of those things that is common and useful for the function generator's intended purpose, but rather peculiar when viewed from a musical standpoint. In this mode, basically the modulation generator creates a ramp wave which can FM the main generator from the minimum to the maximum frequency for the frequency range selected. The starting frequency, ending frequency, ramp rate, and time between sweeps are all controllable. There are several controls which have alternate purposes in the sweep mode (which unfortunately are not labeled on the panel). First, the modulation frequency control's range switch must be set to the 0 position. Once this is done, the modulation percentage knob sets the starting frequency of the sweep. The modulation symmetry control sets the rate of the sweep, and the main frequency control sets the ending frequency. The modulation frequency vernier sets the repetition rate. The controls interact to an extent, and I had to do a fair amount of experimenting to get something that would be illustrative when viewed on the scope. I finally settled on a high frequency triangle wave with a fairly slow repetition rate for the scope photo. The audio sample is a mix of different settings.

Audio -- click here

The one other interesting mode of operation is the "burst" mode. In this mode, the function generator can be made to produce either single-cycle waveforms, or short bursts of waveform separated by periods of 0V output. The burst mode is activated by turning the main generator's trigger phase knob (below the symmetry knob) away from the "free run" position.

Two switches and a jack on the rear panel come into play: the slide switch at the upper left of the rear panel selects the single-cycle or multi-cycle burst mode. The switch below it selects internal or external trigger. With single cycle and internal trigger selected, the main generator will output one complete cycle of whatever waveform is selected, repeating at a rate determined by the LFO in the modulation section. The trigger phase control determines the starting and ending phase of the cycle. With the single/multi slide switch set to multi and internal trigger selected, the square wave of the modulation LFO gates the main generator on and off. When the LFO square wave goes high, the main generator starts at the phase selected by the trigger phase knob and continues to output until the LFO square wave goes low. When it does so, the main generator completes the cycle it is on and then stops. In either single or multi mode, when the trigger switch is set to external, gating of the main generator is controlled by the signal input at the rear panel EXT jack. This signal needs to be at "TTL" levels; that is, 5V for the high state, and 0V for the low state. Here's an example of the burst function at work, with the triangle waveform, and with the phase control and burst interval being varied. The first two-thirds is multi-burst mode; the last portion is single cycle burst mode.

Audio -- click here

Same as above, but with a different starting phase setting:

The rear panel VCO jack provides for a control voltage to control the main generator frequency. Unfortunately, the signal format is not even slightly compatible with the 1V/octave standard used in most synths. Basically, with the frequency control set at its minimum position, the VCO input does what the frequency control does; that is, it varies the frequency from the minimum to the maximum for the frequency range selected. All of the ranges constitute 10:1 ratios between maximum and minimum frequency, so if you do the math, that's slightly over three octaves. However, the VCO input is linear, that is, it is a V/hertz input. The scaling is about 0.2V per one-tenth of the frequency range (so, for example, if the 10 KHz range is selected, it's 0.2V per 1 KHz.) It also uses negative voltages; the minimum frequency is at 0V and the maximum is at -2V.

To wrap this up: Just for fun, here's a mix of the six sample waveforms: Audio -- click here

New Large-Format Modules and Manufacturers

2010-04-21T16:10:00.000-05:00

A while back in this blog, I wrote a post lamenting the lack of modular manufacturers getting involved in large-format (specifically, the "5U" formats -- MOTM, Dotcom, and Modcan-A) modules. All of the action seemed to be happening in the Euro format, and I was getting a bit concerned for the future of large format.

But since then, there have been quite a few new makers jump into the "large market": Curetronic, Rob Hordijk, Moon Modular, MOS-Lab, and STG Soundlabs are all building to the 5U format. And what's encouraging is that most of them are not just building "me too" modules, but are actually applying original thinking to the format. Although, so far I'm not seeing the variety of out-there designs that one sees in Euro... but that might not be a bad thing. At least there haven't been any major scandals involving any 5U makers. (And guys, let's try to keep it that way, please?)

I'll take a look here at two of the most recent entrants. The first of the new contenders is Megaohm Audio, and first up is their Delta VCF. Megaohm advertises this as being a new design and not a clone of an existing filter, which is certainly welcome -- I think the synth world has just about all of the Moog-ladder clones it needs. From the block diagram and the board layout, it appears that this is a 4-pole OTA-based design. Also on board is an auxiliary VCA, going along with a recent trend to include VCAs on modules whose primary function is something else. It often seems as if modulars never have enough VCAs when doing complex routing, so I don't think anyone will complain.

Delta VCA -- photo from Megaohm Audio's Web site

The module is packaged in a 2U width Dotcom format. The panel layout is attractive and fairly well organized, if somewhat busy. The filter has a two-input mixer for audio in, and the input level controls, instead of the usual setup of unity gain at full clockwise, have unity gain at the 2 o'clock position and full CW is a gain of 2. That can be a useful feature when you are doing severe filtering and you are having problems with low output level, or if you want to overdrive the filter. The FM 1 input jack has an attenuator that can select inverted or non-inverted input. The FM 2 input is designed expressly for audio frequency modulation of the cutoff; it is AC coupled and switchable for linear response. The filter also has a 1 V/octave output; the manual notes that it is calibrated over about four octaves.

There's an interesting normalling loop. The filter's output is normalled to the VCA input. The VCA's output, in turn, is normalled to the FM 2 input. This means that if you feed the VCA a control voltage, and nothing is plugged into the normalled jacks, the filter's output is fed back to the FM 2 input. You can of course use the VCA for other purposes, although the normalling creates the oddity that if you are doing so, and you aren't using the FM 2 input, you need to insert a dummy plug into the FM 2 jack.

The filter is switchable between lowpass and bandpass response. There is no voltage control over resonance. However, you could use the VCA to create it: patch the VCA's output back into one of the filter inputs, and then apply control voltage to the filter to control the feedback.

It appears that quite a few large-format users are building Frankensynths these days, accommodating more than one format. Accordingly, you find a variety of power distribution schemes. Megaohm realized this and provides both Dotcom-style and MOTM-style power connectors. Very perceptive on their part! They ship the module with a cable that connects to a Dotcom octopus cable, but for a few bucks extra, they will provide a cable that connects to a MOTM 4-pin power distribution board.

The other module Megaohm is currently offering is the LFO Two. This is a slightly odd combination of two dissimilar LFOs in a 1U-width Dotcom format module. Like the Delta VCF, it has both Dotcom and MOTM power connectors.

LFO Two -- photo from Megaohm Audio's Web site

The top LFO is based on a Korg MS20 design. It outputs square/pulse and triangle/ramp signals, through separate jacks (there are actually two jacks for the tri/ramp output). A shape control affects both the pulse width of the pulse output, and the shift between up ramp, triange, and down ramp on the tri/ramp outputs. Shape is not voltage controllable. This LFO is designed to run slowly; in the "hi" position of the three-range switch, range is from "just below audio" (presumably around 20 Hz) to one cycle every 30 seconds. There isn't much apparent difference between the "medium" and "low" range positions; both provide maximum cycle times of a few minutes.

The top LFO is hard-syncable via a reset jack. The bottom LFO can also be synced to this jack, if a circuit board jumper is moved; more about this in a moment. The bottom LFO is said to be based of that of the ARP Odyssey. Its outputs are square and, oddly, sine -- no triangle or ramp. There is no shape or pulse width control for this LFO.

An interesting feature of the LFO Two is the extensive set of options controlled by moveable jumpers on the circuit board. By moving them, one can, for example, select the peak-to-peak output voltages, what phase the top LFO advances to on reset, and whether or not the bottom LFO responds to reset. One useful option makes the pulse output of the top LFO a positive-going-only signal, which is what you want when you are using it as a trigger, gate, or clock. There's also a range selection for the bottom LFO. It would be a fairly simple matter to DIY an auxiliary panel and patch many of these options out.

The other recent arrival on the large-format scene that I want to discuss is Grove Audio. This is a company that has been known in the pro audio area for a while, but they are now taking the plunge into modular synthesis. Their two current offerings are the GMS-782 Dual LFO/VCA and the GMS-725 four-channel mixer. Both are packaged as 2U width Dotcom format modules.

GMS-782 -- photo from Grove Audio's Web site

The GMS-782 has two halves, each consisting of an LFO and a VCA. Both halves are identical. The panel layout is visually pretty clean, considering how much is going on. The LFOs are voltage controllable, and have sine, square, and triangle outputs on separate jacks. There is a high range and a low range; with the switch in high, frequency range is from 100 Hz to about one cycle per minute. The low range divides the rate setting by 10, which didn't seem that useful until I thought about it some; that's a low end of about one cycle per 10 minutes! And on the high end, I admit to having a preference for LFOs that can go at least a little ways into the audio range; I like to use these to drive VCAs for AM effects. However, there is no pulse width or shape control; hence no ramps. Each half has a trigger (reset) jack for hard sync.

The VCAs can be used to have voltage control over the level of the LFO output. Each LFO has its output wired into a VCA; a three-position rotary switch selects which waveform is fed to the VCA. The problem here is that the VCA input is not patchable, so the VCA is not usable for other purposes, which decreases the utility of this module some. The rotary switch takes up a lot of room on the panel, and if I were DIY hacking this, I'd take it out, put in a normalled VCA input jack, and put in a three-position toggle switch for selecting the waveform normalled to the VCA input.

The other Grove Audio module is the GMS-725 mixer. This is pretty straightforward; it contains a four-input mixer with each input having its own attenuator.

GMS-725 -- photo from Grove Audio's Web site

The nice thing here is that there is also a master gain control. Also, there is a DC offset control that can add or subtract up to 5 volts to/from the output. And there are inverted and non-inverted output jacks. Very neat. Further, there is a separate attenuator with its own input and output, and its own DC offset control. This looks very useful as a control voltage adder/mixer, and the default configuration comes with linear pots on the attenuators for that purpose, although you can optionally order it with audio-taper pots.

It's good to see new manufacturers entering the large-format arena with fresh ideas. It will be even better if these fresh ideas can be implemented while maintaining the quality standards that have become expected in the large-format community. These appear to be a good start.

Q106 Calibration

2010-03-28T16:25:00.000-05:00

While doing the tests and experiments for the MOTM-650 posts, I noticed that one of my Synthesizers.com Q106 VCOs was not scaling properly; high notes were playing sharp. This particular Q106 was one of my first modules; I've had it for five years and it hasn't been calibrated since it left the factory, so it was due. I have a second Q016 and I have calibrated that one (I bought that one used), so I know it can be a bit of a contest of wills, like most VCO calibrations are.

The Q016 has three adjustments, as pictured in the photo below:

The three blue pots right of center are the three basic tuning and scaling adjustments. The four at the upper left are associated with the Q106-CRS calibrated range switch; they allow you to trim scaling for each range (except the 2' and LFO ranges). The three pots at the right, from top to bottom, are the coarse tuning, scaling, and high-frequency compensation.

Dotcom's suggested calibration procedure has never worked for me, for some reason. Their procedure starts with setting the coarse tuning at C0 (32 Hz), going up one octave and adjusting the scaling there, and then going to high C and setting the high frequency compensation there. I always wind up in an endless loop of tuning the low C and then adjusting the scaling, and after I get done with that, it's still not in scale in higher octaves. Here's what I do instead:

Set the range switch to 32' and center the frequency knob.
Center the high frequency compensation pot.
Center the 32' pot on the CRS. (If you don't have a CRS, you can skip this step.)
Tune middle C, turning the coarse tune trimpot until it's in tune. If I hit the limits of the trim pot, I give the high frequency compensation pot about 1/4 turn clockwise.
Check low C, adjusting the scaling pot until it's in tune. If I hit the limit of the scale pot, I adjust the 32' pot on the CRS until I have some more leeway on the scale pot.
Re-check middle C and repeat steps 3 & 4 if necessary.
Check high C. If necessary, adjust the high frequency compensation pot and go back to step 4.
If you don't have a CRS, you're done. If you do, set the range switch to 16', play middle C, and adjust the CRS 16' pot until it is in tune.
Repeat for 8' and 4'. I'm not usually able to get 4' to be exactly in tune; it usually winds up a bit sharp. I don't mind because I rarely use the 4' or 2' settings.

Still More on the MOTM-650 -- Arpeggiators

2010-03-21T23:18:00.000-05:00

I've now had some time to play with the arpeggiator functions on the MOTM-650, and here's what I have found out so far. First, in answer to a question I saw posted elsewhere: where is the MOTM-650 manual on Synth Tech's Web site? There doesn't seem to be a link to it from the 650's page, but the manual and menu navigation charts are here (PDF). However: the manual is unfinished, and none of the stuff that I'm about to cover is in it. So keep reading.

We discussed in the last installment how the 650 contains software constructs called "voice groups" which have physical output channels assigned to them. The arpeggiators are also software constructs. There are two of them, and each can be assigned to a voice group, or turned off. The arpeggiator can be thought of as a mischievous little sprite that looks inside the voice group to see what notes it has received, and then it arranges those notes into arpeggiations which it compels the voice group to play. When an arpeggiator is assigned to a voice group, it takes control of that group and arpeggiates all notes that that group receives. It is quite possible to assign an arpeggiator to one group and have it outputtting arpeggiations, while other voice groups continue to respond normally to incoming notes.

The arpeggiators get their clock source from the source selected in the global parameters. I'm going to write this tutorial-style, using a setup of the 650 and a modular that I described in the previous installment, so if you want to play along, go back and read that and set up your modular as described there.

Arpeggiator Clocking

Two parameters in the global options control how the arpeggiators are clocked. Press ESC as many times as you need to in order to get back to the top level screen. Then press ENTER and you will see:

Global

Options

Press ENTER to go into the global options. Then press INC as many times as needed until you see:

Clock

Internal (or something else)

This parameter selects the clock source for both of the arpeggiators. The available choices are:

Internal. This selects a clock source that is built into the 650.
MIDI Clk. In this mode, the 650 expects to receive MIDI Clock messages, which would typically come from a sequencer or DAW connected to the 650's MIDI In.
Ext. Reg. This mode accepts a clock signal through the EXT CLK jack on the panel.
Ext. Irr. This acts like the above, except that it appears to apply some sort of anti-noise algorithm to the clock signal. I'll explain what I found out about this further down.

For purposes of this tutorial, for right now we want it on Internal. So if it isn't, press ENTER, then INC/DEC until it says Internal, then ENTER again. Now, press INC to move to the next parameter, which is the beats per minute (BPM) setting for the internal clock. (If the clock source isn't set to Internal, you can't access this parameter.) You will see:

IClk BPM

100.0 (or some other value)

Press ENTER, and then use the INC/DEC buttons to select the value you want. If you hold a button down, the value advances rapidly. The allowable range is from 60 to 238 BPM. The bit that is a bit strange is the resolution (step size) by which you can raise or lower the value, because it depends on what BPM range you are currently in. Here's a handy table:

Range	Step Size
60.0 -- 90.0	2.0
90.0 -- 105.0	1.0
105.0 -- 135.0	0.5
135.0 -- 150.0	1.0
150.0 -- 238.0	4.0

The highest resolution steps are concentrated in the range that is typical of club dance music. If you're more interested in doing, say, ambient/space rock, or ballroom music, these might be a bit fast. However, there's a way around that, which I'll get to in a minute.

At this point, if you have entered the value you want (i.e., the parameter isn't flashing), you can press INC and see the other clock-associated parameter:

MIDI Clk

Off (or Transmit)

In any clock source mode except the MIDI Clk mode, this parameter is available. It tells the 650 whether or not to transmit MIDI Clock on its MIDI OUT jack. This is an incredibly useful function -- consider: if you set the 650 to an external clock mode and then drive it with an LFO, you can use the MIDI Out to sync your drum machine to the LFO. If you have a voltage controlled LFO, you can drive the tempo of your whole song with a control voltage!

Assigning and Activating an Arpeggiator

Now let's go look at the parameters of the arpeggiators themselves. Press ESC until you get out to the top level screen, then press ENTER. Now press INC until you see:

Arpgtr 1

Settings

This is where, among other things, you assign the arpeggiator to a voice group. In fact, let's do that now. Press ENTER and you will see:

A1Assign

Off (probably)

Press ENTER, INC, ENTER. It will now say:

A1Assign

VoxGrp1

As of now the arpeggiator is active. If you left the MIDI clock source on Internal back in the global settings, the yellow BEAT LED will now be flashing at the tempo you set in the IClk BPM screen. This is the 650's way of telling you that you have an arpeggiator assigned to a voice group and the clock source is active. (If you left the clock source on MIDK Clk, you won't see the light flash until the 650 starts receiving MIDI Clock messages. Most sequencers and DAWs don't send MIDI Clock when they are stopped. If you left it on one of the external clock settings, it won't start flashing until you connect a clock source to the EXT CLK jack.) For now, go back and set it to Internal if you left it on something else, then come back to this screen. At this point, if you play some chords, you should hear the arpeggiator working!

Now press INC and you will see the next parameter:

Arp1Mode

Normal (or something else)

Press INC again and you will see:

A1 Order

Up (or something else)

These two parameters together determine in what order the notes will be arpeggiated. If the mode is Normal, then the Order parameter determines in what order the notes will be sounded:

Up: Notes are sounded from lowest to highest.
Down: Notes are sounded from highest to lowest.
Up/Down: Notes are sounded from lowest to highest, then back in the other direction. Note when counting beats that the bottom and top notes are sounded only once each in the progression, while the in-between notes are sounded twice.
Down/Up: Like above except starts with the highest note, going down.

The other three Arp1Mode settings behave independently of the Order parameter, and are as follows:

Ordered sequences the notes in the order in which they are pressed.
PingPong is kind of hard to describe. Basically, it takes the notes that are pressed, and divides them into "early" and "late" groups according to the order in which the notes are pressed. Then, it plays a note from the "early" group, a note from the "late" group, then the next note from the "early" group, and so on. The best way to hear this in action is to try this: play and hold three notes with your left hand, at the bass end of the keyboard. Then play and hold three notes with your right hand, at the treble end. What you will hear is a sequence of alternating left-hand and right-hand notes.
Random does what you expect: it plays the notes in random order. Note that this is a totally random choice for each note sounded, which means that the same note may be played twice or more in succession.

There's one more arpeggiator parmeter, and this plays into the clock selection, as we mentioned a while ago. Press INC until you see:

A1 ClkDv

X1 (or something else)

This allows you to set a factor that multiplies or divides the clock rate, for this arpeggiator. The setting X1 makes the arpeggiator play at the clock rate. The settings X1.5, X2 and X4 make the arpeggiator play faster than the clock rate, by the factor indicated. The settings /4, /3, /2, and /1.5 make the arpeggiator play slower than the clock rate, by the factor indicated.

Using the Arpeggiators and Stress Testing

With my 650 configured in the 2/2 voice grouping mode, I assigned arpeggiator 1 to voice group 1, which gave me arpeggiation acting on output channels 1 & 2. I did a lot of playing around with the various modes and parameters and observed how the arpeggiator performs. With the clock source set to Internal in the global settings, I played some chords. As I did this, I went through the different output channel allocation settings in the voice group. Concerning that, here's what I found out about how the how the output channel allocation effects the arpeggiations:

When the allocation mode is set to Solo, all notes output by the arpeggiator are output on the first output channel assigned to the group; any other channels owned by the group are not used.
When the allocation mode is set to Solo Uni, all of the output channels assigned to the voice group play the arpeggiation in unison.
Setting the allocation to Solo Rot is the most interesting mode. In this mode, the arpeggiator outputs successive notes on different output channels. For example, in the 2/2 voice group mode, with the arpeggiator assigned to voice group 1, the first note of the arpeggiation is output by output channel 1. The second note is on output channel 2, the third note is on channel 1, etc. In the four channel voice group mode, where voice group 1 owns all four output channels, the first note goes out of channel 1, the second note out of channel 2, the third out of channel 3, the fourth out of channel 4, and the fifth out of channel 1. And so on.
The poly 1 modes act like Solo Rot. The poly 2 modes act like Solo. The unison modes act like Solo Uni.

This held true for all settings of the Arp1Mode parameter. I got some really unique effects using a 2/2 voice group setup with an arpeggiator on voice group 1, and each of the two output channels patched into a different string of the modular with different timbres, and the voice group's allocation mode set to Solo Rot. With this setup, if you play a chord with an odd number of notes, the notes swap back and forth betw een the two "voices" every other time through. It can create some very complex-sounding sequences with little effort.

There are a couple of "funnies" about the way that the arpeggiator handles notes being added or removed while a chord is arpeggiating. I haven't yet worked my way through all of the combinations, but one thing I noticed with an Arp1Mode setting of Normal and an A1 Order setting of Up: if you are adding or removing a note from the middle of the chord while the arpeggiator is running, you need to do it between the first now (lowest note) in the sequence, and when the arpeggiator gets to where the notes are being added or removed. If you press a note after the arpeggiator has "passed" that note, it will make a higher note in the sequence be double-triggered that time through. Similarly, if you let up a note after the sequence has "passed" it, it may make a higher note be skipped. It almost seems like at some point during the sequence the software is taking a count of the number of notes being held, and it will play that many notes regardless of how many are being held. However, with a bit of practice, you will get the feel of how and where in the sequence to change your chords without getting skipped or doubled notes; I was able to do it pretty consistently at a 120 BPM tempo after a few minutes of playing with it. I'm not sure if I'd consider this a software bug or not -- you just have to learn how to work with it, and if I were spec'ing out the software, I'm not sure what I would have it do differently.

I wanted to see how fast the arpeggiator could actually play, so I set the internal clock to its maximum rate of 238 BPM, and then I set the A1 ClkDv to X4, giving a note rate of 952 notes/minute, or about 16 notes per second. That seemed to work, although I couldn't swear that the sequence was actually sounding every single note; it was just too fast to hear. At some point I'll have to get the scope out and make sure that I'm seeing all of the pitch CV changes and gates -- I think the gate rate was too fast for any of the envelope generators I have in my modular, as it all kind of ran together even with all of the EG knobs on zero. At one point I pressed my sustain pedal (the 650 recognizes MIDI Sustain Pedal and does what you expect it to do) and glissed the whole keyboard, 49 notes. It seemed to be playing every note, and no notes remained hung when I let the sustain pedal up.

The MIDI Clock mode worked and it behaved like I expected it to. I was able to send it MIDI Clock from my DAW (once I found the clock routing parameters in MOTU Clockworks that I'd forgotten about, so that my MIDI Express would actually route it), and the arpeggiator followed tempo changes flawlessly. There is a setting in the voice group parameters which will provide a clock out signal from the aux jack when the clock source is internal or MIDI Clock, which I have not tried yet. At one point when I was trying to sort out the interrelationship between Metro (my DAW software) clocking parameters and the MIDI Express, I somehow managed to get a stuck note in the arpeggiator. Pressing the PANIC button on the MIDI Express, which sends All Notes Off to all outputs, failed to quiet it. However, starting and stopping Metro playback for a moment cleared it. I know that Metro sends a series of MIDI messages intended to quiet stuck notes when it starts and stops. I'll have to look it with an analyzer and see what's different about it that makes it work when just sending All Notes Off doesn't work.

I put the 650 in the Ext. Reg clock mode and took a pulse output from an LFO and put that into the EXT CLK jack. I was able to adjust the LFO rate and the 650's arpeggiation followed the LFO very nicely. I then set the clock mode to Ext. Irr. With the LFO as a clock source, I didn't notice any difference -- it seemed to track the clock exactly the same. I guessed that this mode perhaps was intended to apply some kind of smoothing to the clock rate, so in an effort to see the difference, I connected a white noise source to the EXT CLK jack.

Big mistake! The 650 went nuts; the screen became garbled and the LCD backlight flashed on and off irregularly. I disconnected that, and after pondering it for a moment, I connected the noise source to a lag processor and then connected that to the 650. While I was in the process of checking this out, the 650 stopped responding, and then a few seconds later I got the infamous "HOSED Q2" message. I cycled power and the 650 came back, with all the parameters set as before. So, I re-connected the output of the lag processor, with the slew rate turned all the way up (slowest). As I expected, the 650 saw no clock signal at this setting. So I gradually turned the rate down (faster) until the arpeggiator started triggering.

Here's where I saw a difference between Ext. Reg and Ext. Irr. With the Ext. Reg mode, as I turned the knob on the lag processor down, at a setting of about 3, the arpeggiator started triggering notes at random intervals, which was what I expected. However, when I switched to Ext. Irr mode, it stopped, and I had to turn the slew rate knob on the lag processor down to about 1 before I saw any action. Once it did start triggering, it looked about the same as in the other mode with the same slew rate setting. So I think my initial guess about what the Ext. Irr mode does was wrong. It doesn't average the clock rate. Instead, it appears to be a function that tries to clean up "ratty" clock signals. My guess is that it routes the clock through something like a Schmitt trigger.

There's been a lot of complaints from certain corners of the Internet about firmware bugs in the 650. So far from my experiments with it, I can say that a lot of the rumors are unfounded. Except for the two incidents described above -- both of which were cases where I presented "abnormal" inputs to the unit -- the 650's firmware performed as designed. Why did it crash when I routed the white noise to the EXT CLK jack? I'm not sure, but my guess is that the 650's CPU handles external clock input using the "interrupt" capability built into the CPU (that would be one way of assuring minimum latency in responding to the clock signal). White noise is broad-spectrum; my guess is that the white output of the source I used (MOTM-101 sample and hold) goes up to at least 15 KHz, which is a heck of a lot faster than anything you'd ever want to use as an arpeggiator clock. I ran into a problem on a work project, some time ago, where a small CPU like the one in the 650 was getting bombarded with interrupt signals at a far higher rate than it was intended to handle. Because the firmware was unable to dispatch the interrupts as fast as they came in, eventually it ran into a condition called "stack overflow" which caused a crash. My guess is that the 650 had something similar happen to it. I wonder: is the interrupt for that EXT CLK jack enabled all the time? And if so, does the jack normal to ground when there's nothing plugged in? Possibly, in dry weather, the jack picks up static or RF from the air. Here in Alabama, it's pretty humid most of the time, so we don't have too many problems with static or corona discharge. I should get out a function generator and see how fast I can clock it before it crashes. Sounds like a science project for next week.

As for "normal" use of the 650, though, I've already noted a number of interesting possibilities just going through the experiments that I used to compose these articles. Yes, it's definitely a keeper, and it's going to be used in my setup a lot.

More on the MOTM-650 -- Channels and Voice Groups

2010-03-19T12:30:00.001-05:00

So now that I've had a bit of time to play with it, here is some more information on the MOTM-650 MIDI interface. In this post, I'm going to concentrate mostly on how the the voice groups work and how MIDI note information is routed to the output jacks. In my next post, I'll cover the arpeggiator functions.

Output Channels and Voice Groups

The first thing I had to understand about the 650 is the difference between the four physical output channels and the voice groups. Notice that the 650 has four columns of jacks; each column is an output channel. (The MOTM documentation refers to these as "voices", but I think that's a bit misleading, since the 650 produces no audio output by itself. I'm going to continue referring to them as "output channels" even though that's a bit wordy. And yes, to avoid confusing these with MIDI channels, if I mean "MIDI channel", I'll spell that out.) Each output channel contains (from bottom to top) a pitch CV jack, a gate jack, a velocity CV jack, and an aux CV jack. (The EXT CLOCK jack at the upper right doesn't go with any output channel; it's an independent entity. More about it in the next installment.) Each output channel also has a red LED underneath the LCD display that lights up when that output channel's gate is in the "on" or "high" state; these are labeled V1 through V4 on the panel.

The voice group, on the other hand, is a software construct. Each voice group "owns" one or more output channels, depending on how many voice groups the 650 is configured to use. Each voice group listens to incoming MIDI data on one MIDI channel (which is configurable in the voice group's parameters). When the 650 receives MIDI note messages, it directs the MIDI events to a voice group according to what MIDI channel the data came in on. The voice group then transforms the MIDI note information into control voltage and gate signals, and directs these to be output on one or more of the output channels it owns. This is done according to the allocation mode, which is also a voice group parameter. The neat thing is that if the voice group owns more than one output channel, it can operate them polyphonically or in unison, depending on the chosen allocation mode.

Global Modes

To sort all this out, we need to being by introducing the 650's global mode settings. If you are at the 650 and you haven't pressed any of the buttons, the 650 LCD screen will be at its top-level display. (If you've been messing with it, press the ESC button 4 or 5 times, until the display quits changing.) The display will look something like what is pictured below. (In this article, I will show text that you should see on the 650's screen in bold.)

G1 CH=1
Solo

This summary display shows you the basic status of the voice groups. The above shows that you are looking at voice group 1, and it is set to MIDI channel 1 and is in SOLO mode. More on that in a minute. First, let's go through the global settings. Press ENTER and you will see:

Global Options

Press ENTER again and you will see something like:

VGrpType
1/1/1/1

This lets you select the division of output channels into voice groups. We'll come back to this in a minute. At this point, pressing INC or DEC will scroll through the available global parameters. I'll save a complete recap of the global parameters for the end of this post. For right now, there are two things you may want to change; one is the backlight setting. Press INC or DEC until you see something like:

Backlight
Auto

On this setting, the backlight comes on whenever you press a panel button, and goes off 5 seconds later. You generally want to keep the backlight off as much as possible because it draws a lot of current from your modular's power supply. But for right now, you may find the auto backlight annoying. Press ENTER, and the bottom line of the display will begin flashing. This indicates that you can now use the INC and DEC buttons to change the value. You can change the setting to OFF, AUTO, DIM, MED, or BRIGHT, and as you change it, you will immediately see the change. Press ENTER again to store the change, or ESC if you decide you really didn't want to change it after all. (This will be the case for all of the parameter-changing screens.) Either way, the bottom line of the display will stop flashing, indicating that you are no longer in the parameter change mode.

Now, press INC some more until you see:

Priority
LastNote (or something else)

The three possible values are Last Note, Low Note, and High Note. Most of the time you will want this on Last Note -- I'll explain why in a bit. If it's not on Last Note now, press ENTER so that it flashes, and then use INC/DEC until it says Last Note, and then press ENTER to lock that in.

At this point, pressing INC/DEC will show other global parameters. Pressing ESC will take you back to Global Options, and pressing ESC again from there will take you back to the top level screen.

Go to the Global Options again and press ENTER. You will be at the VGrpType screen. Here you will select the number of voice groups and how they will divide up the four output channels. The 650 can be configured to have (text in brackets shows what the second line of the display shows):

One voice group which uses all four output channels [4].
Two voice groups, each of which uses two output channels [2/2]. In this mode, voice group 1 uses the two leftmost output channels, and voice group 2 uses the two right most.
Four voice groups, each of which uses one output channel, counting from left to right [1/1/1/1].

For the purpose of following this discussion, use INC and DEC until 2/2 is shown, and then press ENTER to lock it in.

Voice Group Parameters

Each voice group has a set of parameters that tell it how to interpret MIDI data, and what to do when the voice group receives more MIDI notes than it has output channels available. To get to the settings for each voice group, press ESC as many times as you need to to get back to the top level screen. Now, press ENTER and then INC, and you will see

Voxgrp1
Settings

You are now in the parameter settings for voice group 1. Let's step through these parameters. Press ENTER and you will see:

VG1 MIDI
Chan=1

This says that voice group 1 expects to receive MIDI data on channel 1. Presuming that you have either a MIDI controller or a computer with a MIDI interface connected to the 650, check which MIDI channel your controller/computer is transmitting on. If it isn't channel 1, go through your steps for changing a parameter's value on the 650: press ENTER (bottom line of screen starts to flash), use INC/DEC to change the value, and then ENTER again to lock it in. Press INC again to see something like:

VG1 Allo
Solo (may be a different word)

This is the poly mode / output channel allocation selection. We'll come back to this one. Press INC again, and you will see:

VG1Glide
Off

This is your glide, or portamento setting. When on, it causes pitch control voltages output by this voice group to move smoothly between note values. There are constant-time and constant-rate settings. The constant-time setting can, incredibly, be set in millseconds from 1 ms to 65.536 seconds. The constant-rate setting is in arbitrary values from 1 to 127. Experimentally, setting 127 results in portamento that moves at about one octave in 100 ms; a setting of 1 causes it to take around 22 seconds to transition one octave. Leave this alone for the time being and press INC again, and you will see:

VG1 PBend
2 (or some other value)

This sets the pitch bend response in semitones. When the voice group receives MIDI pitch wheel messages, it will increase or decrease the pitch CVs being output according to the pitch bend. (You can also get the isolated pitch bend voltage as a separate output signal from an aux jack.) This parameter sets how much the full-scale bend is, in semitones. Allowed values range from 0 (no bend) to 24 (two octaves in either direction). If you want to change it, press ENTER, use the INC/DEC buttons to change the value, and then press ENTER again. Now press INC to see:

VG1GType
Normal

This parameter allows you to change the function of the gate jacks in the output channels assigned to this voice group. The available setting other than Normal is S-Trig. This is a requirement for interfacing to some old Moog gear; if you have any such, you probably already know about S-triggers. Otherwise, leave this parameter alone.

The next screen is:

VG1 V/T
Velocity

The Velocity jack in an output channel normally outputs a control voltage that represents the MIDI velocity of the note being played. This parameter allows you to change the jack's function to output a trigger instead. I'm sure this has a purpose, but I'm not sure what. Leave it alone and go to the next screen:

VG1 Aux
CC1 ModW (or something else)

This allows you to select what signal will be output by the Aux jacks. The available choices are Velocity (outputs the MIDI velocity), PitchBnd (outputs the pitch wheel value independent of the pitch CV), ChAftTch (aftertouch), ClkPulse (will be covered later), Disabled, or any MIDI Continuous Contoller (CC) from 0 to 31. Use the ENTER and INC/DEC buttons if you want to change it. Then press INC to go to the next screen:

VG1AUXSc
4 Volts

This sets the scaling for the aux jack; it will output the indicated voltage when the selected control signal is at its maximum value. It can be set to 1, 2, 4, or 8 volts. Depending on what you have the signal coming out of the aux jack routed to, it can be handy to be able to change this, but most of the time you will probably want to leave it on 4 volts.

Voice Group Allocation Modes

Now, press INC several more times until you cycle back around to the output channel allocation parameter:

VG1 Allo
Solo (may be a different word)

The most complex aspect of the voice groups is the output channel allocation, and they interact with the allocation modes. The complexity comes in when the "4" or "2/2" voice group mode is selected in the global parameters, so that the voice group owns more than one output channel -- what does the group do with each of the outputs? It's somewhat like the process of allocating voices on a polyphonic synth.

To illustrate the modes, we will now do a little tutorial. If you have sufficient resources in your modular, patch up a little "two-voice" demonstration patch. You'll need two VCOs, two EGs, two VCAs, and a mixer. Note that filters aren't necessary for this purpose; you just want to be able to hear notes play. Now patch it up like this:

The first output channel on the 650 has its pitch CV patched into VCO #1. The gate is patched into EG #1, and the VCO and EG outputs are patched into the signal and control inputs of VCA #1, respectively.
The first output channel on the 650 has its pitch CV patched into VCO #2. The gate is patched into EG #2, and the VCO and EG outputs are patched into the signal and control inputs of VCA #3, respectively.
The signal outputs of VCAs #1 and #2 both feed into a mixer.

If, back when we discussed the global settings, you set the voice group mode to 2/2, you are now good to go. Otherwise, press ESC until you get back to the main screen; press ENTER to get the global options screen, use INC to get to the VGrpType parameter, change it to 2/2, and lock that in. Now press ESC until you get back to the main screen, press ENTER to see the global options, press INC to get to the voice group 1 options, and press ENTER. Then, press INC until you see the VG1 Allo screen.

Press ENTER to change the parameter, and press INC or DEC until you see:

VG1 Allo
Solo

and then press ENTER to lock it in. This is the easiest of the alloction modes to understand. In this mode, the voice group plays in strictly monophonic fashion, and it only uses the first of whatever output channels are allocated to it. So as you play, you will see only the V1 light lighting. If you have set up the demonstration patch, you will only ever hear VCO1. Now, press ENTER, then INC twice, then ENTER again, and you will see:

VG1 Allo
Solo Uni

This mode will "play" all output channels allocated to the voice group in unison. So if the 650 were in the "4" voice group mode, you would see all four sets of CV/gate/velocity jacks will output the signals for that note, and you'll see this by way of all four of the red gate LEDs lighting up when you play the note and going out when you release it. As it is, if you are following along with this writeup and you have your 650 in the 2/2 voice group mode, the first two output channels will be active and the V1 and V2 LEDs will light when you press a note. If you have patched your modular as above, you will hear your two VCOs playing in unison (or whatever interval you have tuned them to).

What if you play more than one note in these mono modes? Remember that "Priority" parameter we looked at back in the global parameters? It determines what happens in this situation: it can be set to output either the highest note played, the lowest note played, or the last note played (that is, the note most recently pressed). If this note is subsequently released while other notes are still held, the 650 will again determine which note has priority and the pitch CV will jump to that note -- without the gate signal going off and back on. This is sometimes called "retriggering" and can be observed in many monophonic synths. If you have the gate signal patched conventionally into an envelope generator, the new note will play while the EG remains in its sustain phase, since the gate signal didn't cycle. One use of this is to simulate the "hammer" and "pulloff" techniques used by guitarists. The gate signal will not go low until all held notes are released. If you want to play with this, you can go back to the Global Options and change the Priority setting, but make sure you change it back to "Last Note" before proceeding.

Press ENTER, DEC, and ENTER again, and you will see the next mode:

VG1 Allo
Solo Rot

In the "Solo Rot" (rotation) mode, the voice group will not activate all of its output channels at once. Rather, the first note played will be output by the first available output channel. When another note is played, that note will be output by the next output channel, and so on. This is a sort-of-monophonic mode in that only one gate in the voice group will be active at a time. But. if you have set up the demonstration patch described above, turn the Release settings on both of your EGs to long values. Now play some notes. Can you hear two notes sounding at once in places? Here's what's happening: basically you have patched your modular as a simple two-voice polyphonic synth. Now, when you play a note, its pitch CV and gate will be output by output channel 1 on the 650, and you will hear the note played by VCO #1, EG #1, and VCA #1. Now you play a second note. This note will be output by the pitch CV and gate jacks of output channel 2 on the 650. The gate for output channel #1 will drop, since the 650 is still in a mono mode. However, with a long enough release time on EG #1, you will continue to hear the first note as it releases while the second note sounds. Note that the pitch of the first note remains the same -- the first output channel continues to hold the pitch CV of the previously played note, even though it has dropped its gate. It has to do so in order that the pitch of the note can still be heard during the release phase. The 650 doesn't know how the EG is set up (or anything else about how the modular is patched up), so it most hold the pitch CV until it needs that output channel for a new note. This is actually a fundamental principle of control of any analog synthesizer: the control interface, whatever type of mechanism it may be, must hold the pitch indication of the last played note even after the performer releases the note, since an arbitrary amount of time may elapse between note release and the note fading to inaudibility. Going back to our setup, when you play a third note, the rotation will return to the 650's first output channel. If the previous note is still sounding owing to a very long release time, the pitch will jump as the output channel outputs the pitch CV and gate for the new note.

And that leads us into the the polyphonic modes. Go press ENTER, INC, INC, ENTER, and you will see:

VG1 Allo
Poly1

Now play first an E and then an A. Since the E is the first note played, it will be output on output channel 1. The A will then be output by output channel 2. You will hear both notes being sounded. Now let up first the A and then the E, and then play an F. Which channel outputs the F? In poly 1 mode, it's the second channel. Why? Because in this mode, the 650 tries to allow for the longest release time possible for each released note before it re-uses the output channel. Since the A was released first, and it was being output by the second channel, the 650 chooses the second channel to play the F. If we let up the F and then play F#, the F# will be output by the first channel, since that channel hasn't been used since the E was released, and that was longer ago then when the F was released. In computer science terms, this is called least recently used. The advantage of least recently used mode is that it is least likely to "cut off" a note that hasn't yet faded into inaudibility. Of course, with only two channels allocated to the voice group, it isn't going to make much difference if you are playing fast. But if you had a four-note poly setup, and were in the mode where voice group 1 uses all four channels, it might make a difference. More to the point, you can control the output channel usage by how you release notes, which opens up possibilities like patching each output to patches of unlike timbre, and then using it to create patterns of notes that vary in both pitch and timbre.

NOTE: none of the poly modes will work right unless the Priority parameter in the Global Options is set to LastNote! The Low Note and High Note priority settings interact with the poly modes in an unfortunate way: if you are holding a key on the keyboard, the voice group will ignore any lower-value or higher-value notes that are played, depending on the Priority setting. If you have your 650 in a Poly mode and it seems to be "missing" notes, check the Priority setting.

Note that this poly 1 mode will not "steal" a note; that is, if you are already holding two notes and you press a third, the third note won't be heard because there is no output channel available for it. However, if you press ENTER, INC, and ENTER, you put the voice group's VC1 alloc into "Poly1 St" mode. Now the third note will steal an output channel from a note that is still being held.

Press ENTER, and then keep pressing INC until you see:

VG1 Allo
Poly2

and then press ENTER to lock it in. Poly 2 mode allocates output channels to notes according to which note was played first. As in our example above, if you hold E-A with the E being struck first, the E will be output on the first channel and the A on the second channel. If you then let up these two notes and play an F, the F will be output on the first channel. Letting up the F and playing F# will output the F# on the first channel, and so on. As long as you play only note at a time, that one note will be output on the first channel. Only when you play two notes will the second note you played be output on the second channel.

What good is this? Well, it's the bees' knees for polyphonic portamento. You can enable portamento (glide) mode for the voice group (go back to the discussion above about the voice group parameters) and portamento will be applied to each output channel as it transitions from one note to the next. In the Poly 2 mode, you have control over which output channel plays which note by slightly arpeggiating the notes you play; the first one you strike will be played by the first channel, and so on. This way, you can control the glides between individual notes in a chord and prevent them from "crossing" each other. (Or make them all cross if that's what you want to do.) As in the case of Poly 1 mode, Poly 2 is non-stealing, but there is a "Poly 2 St" mode which will steal.

Finally, we have the polyphonic unison modes. Press ENTER and then INC as many times as needed until the display reads:

VG1 Allo
Unison

This is a rather strange but interesting hybrid mode. Play one note, and both output channels allocated to the voice group will output the note in unison. However, if while holding that note, you play a second note, it takes away the second output channel, and then it behaves as if it were in Poly 2 mode. This continues until all notes are released, and then the next note will be played by both channels in unison. When the VGrpType is in "4" mode, it gets more complicated:

Play and hold a note, It gets all four output channels, in unison.
While holding that note, play and hold a second note. It will be output on output channels 3 and 4, while the first note retains output channels 1 and 2.
While holding those notes, play a third note. It will take output channel 2, and output channel 4 will mute (the gate will drop).
While holding those notes, play a fourth note. It will take output channel 4.

If that isn't all complicated enough, there is also a UnisonSt mode that will steal.

The Aux Jacks and MIDI Controller Data

Before we go, let's talk about the functions of the AUX output jacks some more. The aux jack can be made to respond to MIDI control signals coming from the MIDI channel that is assigned to the voice group. This operates independently from the note tracking and voice allocation; unless the group is set to output velocity on the aux jacks, all of the aux jacks in a voice group will output the same signal. This can be set to track pitch bend, velocity, aftertouch, or any MIDI continuous controller number in the range 0-31. (Mod wheel, a commonly used choice, is MIDI CC 1.) Another voice group parameter allows you to set the scaling of the aux jack.

Something worth mentioning is that in the "2/2" or "1/1/1/1" voice group modes, it is not a requirement that each voice group be set to a separate MIDI channel. If two or more groups are set to the same channel, they will all respond to that channel. One use of this is to be able to output multiple performance control parameters. For example, in "1/1/1/1" mode, you could set voice groups 1 and 2 to the same MIDI channel, but set the aux jack for voice group 1 to track aftertouch, and set the aux jack for voice group 2 to track CC 1 (mod wheel).

In a later post (hopefully within the next week), I'll cover the 650's arpeggiator functions.

New additions to the Discombobulator

2010-03-17T00:01:00.000-05:00

Here are two new additions to the Discombobulator:

The above is the Synth Tech MOTM-650 four-channel MIDI interface. I haven't had a lot of chances to play with it yet, but it's pretty slick. Each channel converts a MIDI Note On/Off pair to a control voltage representing pitch, an envelope gate, a control voltage representing velocity, and a fourth output which can be associated with a MIDI control number. The pitch CV reacts and responds to MIDI pitch wheel messages also.

It basically has three operating modes. It can be configured as a monophonic interface, responding to MIDI Note On/Off messages and placing the same output signals on all four channels. It can be configured to be four-voice polyphonic on a single MIDI channel -- in this mode, each output channel pitch, gate, and velocity for up to four active notes on a selected MIDI channel. It can also have the output channels divided between four MIDI channels. I believe there are some combinations of the above, but I haven't had a chance to play with that yet. There are also significant MIDI clock conversion modes and a built-in arpeggiator function, none of which I have explored yet.

The arrival of this module means that I can now dedicate my JKJ CV-5 to controlling my EML 101. EML gear uses a different scaling standard for pitch control voltages, 1.2V/octave as opposed to the now-industry-standard 1V/octave. The CV-5 is one of the few MIDI interfaces that can be scaled to this standard. Now I don't have to mess with its calibration to move it back and forth between the 101 and the modular.

I'll have more info up on the MOTM-650 after I have had some time with this module.

The second new gem, pictured above, is the Encore Electronics UEG-01 Universal Event Generator. It's been out of production for several years, as Encore has switched its focus in module design to Frac format modules, but recently Encore solicited on Muff's to see if there was any new interest in the UEG. There was, so they are now doing a reissue, and this is a new unit from the current run.

So what is a universal event generator? It's an expansion and generalization of the concept of an envelope generator. We tend to think of a conventional ADSR envelope generator as having four phases, but it's really only three, and we only have full control over one of the three:

For the attack phase, we only have control over the time that this phase takes. We do not have control over what level it ends at; it always ends at the maximum level that the enveloper generator is capable of generating.
We have full control over only the decay phase. We have control over its time (the decay time) and over what level it ends at (the sustain level).
The sustain really isn't a phase of ADSR envelope generating, when you think about it. That's because as far as the EG is concerned, there's nothing happening; it's simply waiting for the key gate to go low.
We have control over the time of the release phase. We don't have control over what level it ends at; it always ends at the zero level.

A universal even generator gives you control over the time and ending level of every phase. (In theory; it's usually the case even with UEGs that the last phase always ends at the zero level. That's true of the Encore UEG too. The only envelope generator I've come across in a hardware synth that allows the last phase to end at a non-zero level is the pitch envelope on the Roland JD-800.) The Encore UEG has eight phases, each of which has a time control sweepable from a few milliseconds to about eight seconds. The first phase is actually somewhat faster when the time control is set to full CCW, so that it can produce a nice percussive attack spike. Each phase except the last also has a knob specifying what level that phase will end at.

So what can you do with these eight phases? This is an area where the Encore UEG really stands out. There are basically three operating modes that you can select with the pair of 3-position mode switches. Putting the top switch in the GATED position and the second switch in the RELEASE position selects a "conventional" envelope generation -- but with much more flexibility. In this mode, when the gate goes high, the UEG will begin executing with phase 1, going through the times and levels as set for each phase It will continue doing this until it reaches the phase that is selected by the LOOP END switch. After that phase ends, if the gate is still high, the UEG will jump back up to the phase selected by the LOOP START switch. The steps between the LOOP START and LOOP END selections therefore constitute a sustain loop, allowing periodic variation in level during the sustain phase, and therefore a more interesting sound if you're using the UEG's output to control volume or filter setting of a patch. When the gate finally goes low, the UEG will jump to the phase following the LOOP END selected phase, and from there it will execute the remaining phases until phase 8 completes.

You can see how this can produce much more powerful and interesting envelopes. Depending on the LOOP START and LOOP END selections, you can have up to 4 attack/decay phases preceding the sustain loop, and up to 3 release phases. Or you could have a sustain loop consisting of as many as 6 phases. A further variation can be introduced by placing the second-from-the-top mode switch in the FINISH LOOP position. In this position, when the gate drops, the UEG will complete the current iteration of the sustain loop before preceding to the release. (I noticed a peculiarity in this; if the gate drops when the sustain loop is in the phase selected as LOOP END, the loop will usually run one more time before the UEG proceeds past the loop.)

The UEG interpolates between the level settings of each phase to produce slopes in the output. How it does this can be selected by the SLOPE switch. There are three settings, two of which might be conventionally used in envelope generation. In the top setting, rise and fall is exponential -- it starts rapidly and slows down as the level setting is approached. This emulates the way that most analog envelope generators work. In the middle setting, rise and fall is linear, which may be more desirable when using the UEG's output as a pitch or filter cutoff modulation envelope.

In the bottom setting of the slope switch, the UEG jumps immediately to the level setting at the start of each phase, and holds that level for the time of that phase. That can make for some interesting envelope generation, but it's intended primarily for use with the UEG's sequencer modes. Yes, the UEG can be used as a (pseudo-) analog sequencer! There are two ways of going about this. With the top mode switch still in GATED, if the second switch is set to STEP, then the UEG will advance one phase every time the gate goes from low to high. In this mode, you can input a clock signal (say from an LFO) into the GATE jack, and the UEG will act like a conventional 8-step sequencer. You set the output values with the LEVEL controls for phases 1-7; in order that step 8 doesn't have to end with zero level, in this mode only, its TIME knob becomes a level control instead. The loop switches and the phase 1-7 time knobs are ignored in this mode.

Setting the upper mode switch to LOOP ONLY turns the UEG into a self-clocking sequencer consisting of 2-6 steps, as selected by the loop start and end switches. In this mode, the selected loop runs continuously (the GATE input is ignored). The time controls are active, so that each step can occupy a different amount of time. There is no way to sync the UEG to an external clock in this mode, so instead there is a provision for the UEG to be the sync source: at the start of each loop, it will output a short trigger pulse from the TRIG OUT jack, which you can slave other modules and functions to. The rate will obviously depend on the sum of the time settings within the loop.

Finally, the UEG can be used to generate staircases or other complex waveforms by placing the upper mode switch in the ONE SHOT position. In this mode, each time the gate goes high, the UEG proceeds through all eight steps one time (the loop switches are ignored). All time and level controls are active. A trick you can use here is to trigger it with an LFO so that it generates a repeating complex waveform. You can effect the shape of the waveform with the SLOPE switch.

The TCV jack accepts a control voltage which scales up all of the time settings at the input voltage goes up. The UEG manual notes that the absolute maximum time for any stage is 8.3 seconds, so as the TCV voltage goes up, longer steps will be limited to this max time while shorter ones continue to get longer. That's a limitation, but it could also be useful. A handy manual gate button completes the panel. I've noted that the manual gate button double-hits occasionally; the manual does note that it is not debounced and is not meant as a performance control.

Both the MOTM-650 and the UEG-01 are valuable additions, although obviously much different in purpose. However, I'll note one thing they have in common: they are both microprocessor-controlled modules. There are people who will object to this, but I think that MIDI control is the gateway to (among other things) getting away from the "keyboard paradigm" of modular synth control. And the complexity of the UEG would be near-impossible in analog circuitry. It's far past time that more digital functions started showing up in modular synthesis, and the Euro format manufacturers such as Harvestman and MakeNoise have gotten ahead of the 5U world in this area. So it's good to see some large format designers approaching (or in Encore's case, re-approaching) the 5U world with a fresh eye for digital methods. Incidentally, Encore is now soliciting interest at Muff's for a revival of the 5U version of its highly regarded digitally-controlled frequency shifter. If you at all interested, chime in on that.

Painting

2010-03-11T22:49:00.000-06:00

Screw heads, that is. I want the screw heads to match the finish of the modules in the Discombobulator. It's hard to find small wood screws that are painted. So I roll my own. The paint is gloss black modeling enamel. Unfortunately, it's quite humid here so they aren't going to be dry until tomorrow.

Meanwhile, it occurs to me that it's been quite a while since I update the module lineup of the Discombobulator. I've decided to name the four function blocks after moons of Saturn. I was going to use Jupiter moons at first, but Jovian names have already been used considerably in the synth world, including the very successful Jupiter and Juno synths from Roland. But as far as I know, Saturn names have barely been touched except for one unremarkable string synth from Roland in the early '80s (the SA-09 Saturn). So the four existing blocks are now named Titan, Dione, Rhea, and Iapetus. A fifth block, currently in the parts-ordering stage, will be named Tethys, which will cover the moons discovered by Huygens and Cassini (the astronomer, not the satellite). So here's the lineup of the three blocks in the room with me:

Titan (so named because it was the first moon of Saturn discovered, and the first function block I built:

Synthesizers.com Q141 oscillator aid
Synthesizers.com Q106 VCO
1U open
Synthesis Technology MOTM-190 VCA
Synthesizers.com Q109 envelope generator
Synthesizers.com Q123 standards
1/2U open

Dione:

Synthesizers.com Q130 clipper/rectifier
Synthesis Technology MOTM-410 triple resonant filter
Synthesis Technology MOTM-510 WaveWarper
Cynthia Synthacon VCF
Synthesis Technology MOTM-440 OTA VCF
Synthesis Technology MOTM-890 micro mixer

Rhea:

Synthesizers.com Q106 VCO
Synthesizers.com Q161 oscillator mixer
Synthesis Technology MOTM-320 VC LFO
Synthesis Technology MOTM-101 sample and hold
1/2U open
Synthesis Technology MOTM-890 micro mixer

Iapetus is out on the workbench getting some new modules installed. Here's a pic of one of them:

Oh darn, the flash washed out the image. Guess you'll just have to wait until this weekend to find out what it is!

Alternative Keyboards

2010-03-08T22:05:00.001-06:00

We all know that it was a technological accident that the piano-style musical keyboard became associated with the synthesizer. In the early 1960s when Moog and Buchla were devising the first practical analog synths, the CPU power needed to process signals from an instrument such as a guitar or a flute or a marimba, and transform them into the control voltage and gate signals necessary to control a synth, didn't exist. The keyboard presented an easy-to-implement method: conceptually, just mount pushbutton switches under the keys. And in comparison to the capacitive plate sensors and things that Buchla was experimenting with, the piano-style keyboard had the advantage of already being familiar to millions of musicians, thus lessening somewhat the rather steep learning curve for this new type of instrument. These early keyboards were neither velocity nor aftertouch sensitive, but then again, neither are most pipe organ keyboards.

This association of the keyboard with the synthesizer eased its entry into the world of music, but it also placed limitations on how the instrument is played that its designers didn't intend. The limitations of the piano keyboard have been recognized since long before the synthesizer existed. The biggest problem that the keyboard has always had is that, due to the two-row layout with all of the naturals on the bottom row and all of the accidentals on the top row, the performer must usually change fingering in order to transpose a chord from one key to another. This frustrates what should be a simple operation; the guitar player playing a barred chord can transpose it simply by moving up and down the neck, but the keyboard player must keep shifting fingers around to insure that each finger hits on the correct row. The additional manual dexterity and muscle memory requirement makes learning the different keys on the piano a slow and frustrating process. From my own experience, it also introduces the temptation to use teaching shortcuts that cause the student problems later on: a common technique is to start the beginning student out learning the C-major scale, which is played all on the white keys. This introduces a sort of fear or puzzlement at the black keys -- what are the for? When does one use them? And then when the teacher starts introducing other scales, the use of the black keys seems arbitrary and unsystematic, and the student gets a bit freaked out. By contrast, guitar pedagogy treats the accidentals as simply other notes in the chromatic scale, which they are, and the guitar student has relatively little trouble understanding how to play different scales and keys.

A number of inventors have tried to tackle this problem with "uniform" keyboards, which work by mixing the white and black keys across octaves. Paul von Jankó patented a uniform keyboard now known as the von Jankó keyboard in the USA in 1892. His keyboard uses six alternating rows of two patterns of mixed naturals: one row contains the sequence A B C# D# F G A, and the next row contains A# C D E F# G# A#. von Jankó's keyboard has six rows total (I'm not sure why; it appears that four would have done). If you go look at the diagram at the link above, you can see that, for example, if you play an E minor chord (E-G-B), you play with the index finger (or thumb) on E on one row, and the other fingers play the G and the B on either the row above or the row below, whichever is more convenient. That is admittedly harder than on the conventional piano keyboard, in which E minor is all white keys.

However: Now let's play an F# minor. This is F#-A-C#, and on the conventional keyboard it's very awkward because it's a white key in between two black keys. Unless you have good dexterous fingers, you wind up either trying to slip a finger in between G# and A# to play the neck of the A key, or you resort to bad technique and use your thumb to play the A. However, on the von Jankó keyboard, it's very easy: you take the E minor formation and just move over one column to the right. The pattern of the fingers doesn't change. Well, what about B minor? There isn't any B on the row you're on. However, if you just go up or down a row, keeping your fingers positioned as they are, and then put your index finger on B, you're playing it. And without repositioning fingers. A keyboard that has this property is called "isomorphic" because a given chord has the same fingering shape regardless of what key it is played in.

A piano fitted with a von Jankó keyboard. From an excellent page on uniform keyboards from squeezehead.com.

The von Jankó keyboard is neat because it's so easy to transpose things, and also because its span is a bit more compact (it's a whole step between every two keys on the same row). But, well, it's rather expensive and awkward to implement. At least on a conventional piano. But maybe reduce the number of rows a bit, and take the concept from the piano to a MIDI controller, and now you've got something that might be practical. (The controller outputs conventional MIDI note messages, and the synth that receives them doesn't know they came from an alternate-layout keyboard.) The Chromatone (most of the site is in Japanese; the link given should put you on the English home page) is a five-row uniform keyboard, of which at least a small production run has been made. Unfortunately, it's a bit tough to tell if it's still available, and the company who makes it has gone through several name changes. One problem I see is that there is absolutely no identifying info on the keys -- they're all white -- although that could be fixed.

The Chromatone Von Janko-style keyboard

The Bilinear Chromatic Keyboard is a bit more practical, but so far it's only been prototyped in a two-row version. With only two rows, you have to learn two finger patterns (an "up" pattern and a "down" pattern, according to which row the root note is on) to play a given chord in any possible key. This one also hasn't made it to production, and it appears that the Web site hasn't been updated in the past year.

H-Pi Instruments started with the concept of retaining the same fingering for a chord in all keys, but then went in a different direction with it to create the Tonal Plexus keyboard. This keyboard solves the fingering problem in a simpler manner than the von Janko keyboard -- it put all twelve tones of the equal tempered scale in a row. To ease fingering so that all of the fingers don't have to be in one line (which is difficult on account of the fingers not all being the same length), the notes are laid out in a staggered row, and the accidentals are "split" into two halves, a half-row above and a half-row below.

The Tonal Plexus TPX-2s. This is the smallest of four versions available.

Here's where it starts to get more tricky. As you know if you've studied non-equal-tempered tunings, the equally tempered scale reduces the number of possible tones in the scale by making adjacent ones "enharmonic" -- they are tuned the same. For example, in the equal tempered scale, F# and Gb are the same note. It also essentially eliminates the use of double sharps and double flats, since these are just one whole tone up or down, e.g., A-double-sharp is simply B. However, because of the mathematics of non-equal-tempered tunings that rely on "perfect" intervals, these assumptions are no longer true; F# and Gb are not the same note, and A## is not the same as B.

The Tonal Plexus keyboard accommodates just and other non-equal-tempered tunings in two ways. First, each note on the keyboard is not actually a single key. It's a column consisting of either three or four groups of related-but-not-identical notes. They are arranged so that moving diagonally up and right from a natural note gives the sharp on the next column to the right, then the double-sharp on the column to the right of that, and even the triple-sharp to the right of that. Similarly, moving to the left and down yields the flat, double-flat, and triple-flat. Using a tuning table that does not tune sharps and flats enharmonically, this in itself goes a long way towards being able to play a number of non-equal-tempered tunings. However, there is one more feature: each "note" actually consists of one main key that plays the note per the tuning table, plus two keys above and below that allow the note to be played just slightly sharp or flat (about 6 cents per each key up or down, using the default tuning table). Since, for example, just intonation is specific to the base key of the intonation, and notes have tunings that change depending on the base key, the micro-sharps and flats would allow just intonation scales to be played without the keyboard having to be tuned a priori for a specific base key. A pretty clever system, although trying to sight-read the possible fingerings makes my head hurt, to be honest.

The Tonal Plexus actually works by transmitting each MIDI note on a separate channel, and preceding it with a pitch bend message that sets up the pitch bend to accomplish the micro-tuning of the note. This depends on having a multimbral synth; all channels in use have to be set to the proper patch, and the pitch bend range has to be settable to a small enough interval so that the pitch bend can be used to tune the individual notes. So the pitch bend is not available for actual bending. It's not clear from the documentation how the keyboard actually divides up the notes; it seems to suggest that the keyboard is divided into zones, which would seem to put restrictions on the chords that can be played and the total polyphony. Maybe I'm not understanding that part right. The Tonal Plexus is available in 2-, 4-, 6-, and (incredibly) 8-octave versions, and H-Pi claims that all are shipping now. The Cortex Designs Terpstra implements a similar idea, but it doesn't appear to have made it to production.

C-Thru Music took yet another tack towards the generalized keyboard. Discarding both the duplicate key rows of the von Janko keyboard and the microtonal playing capability of the Tonal Plexus, they went with a very unusual-looking layout using hexagonal-shaped keys in their Axis-64 controller. (A smaller and less expensive version, the Axis-49, is also available.) Starting from the bottom of each column, the keys play a Pythagorean circle of fifths as you go up the column. The next column to the right starts such that, for any given key, the key that is up and to the right of it is a major third above. So, playing two keys that are in a lower-left-to-upper-right diagonal plays a major third interval. If the key that is directly above the lower key is then added, the result is a major chord. Three adjacent keys, very easy to play.

C-Thru Axis-64. It isn't clear to me whether the note legends on some of the keys are a standard feature or not; I've seen photos without them. Photo from Steelberry Clones.

As the mathematics of it works out, for a given key, the key that is up and to the left of it plays a minor third above, so then adding the fifth plays a minor chord. Adding sevenths is another adjacent key; the C-Thru Web site gives the shapes for many common chords, and as it happens, in many cases the major and minor chords are mirror images of each other about the vertical axis, which makes them easy to remember.

The tradeoff is that it makes some other common chords a bit weird:

But it does neatly accommodate the tritone!

All of these devices offer alternates for those who either aren't facile or feel constrained by the normal chromatic piano keyboard. There is a lot to be said for having isomorphism of chords, as any guitar player can tell you. However, there are also tradeoffs, not the least of which is that all of these devices are considerably more expensive than a standard semi-weighted keyboard. The C-Thru Axis hexboards look like the most accessible (and most affordable), and should open up new possibilities by making certain things such as large-span chords a lot easier, plus the fact that the chord shapes can help with learning music theory. If you have a desire to play in non-equal-tempered intonations, and a lot of patience, you might want to check out one of the Tonal Plexus keyboards.

Review: Korg Nano controllers

2010-03-07T01:40:00.002-06:00

I recently picked up all three of the Korg Nano line of controllers -- the Nanokey mini keyboard, the Nanopad drum pads, and the Nankontrol control surface. All three controllers are USB devices that draw their power from the USB bus (there is no provision for self-powering). They are packaged in usefully small, slim, and reasonably light-but-not-too-light cases. They all have rubber feet and will stay put when placed on a smooth hard surface. I purchased all three with white cases; the Nanokontrol and the Nanopad are also available in black. All three devices are about 13" (32 cm) long and 3" (8 cm) from front to back. The Nanokey and Nanopad are about 1/2" (1 cm) tall; the Nanokontrol is somewhat taller.

All three devices came with USB cables. The devices are supposed to be class compliant, but Korg supplies specific drivers for Windows XP/Vista/7 and Mac OSX, and they strongly recommend using their drivers with XP and Vista. Korg also supplies Windows and OSX versions of their configuration editor, a single application that edits all three devices. The editing application is required to set up and configure the devices; it can't be done using the controls on the devices themselves.

Top to bottom: the Nanokontrol, the Nanokey, and the Nanopad.

The Nanokontrol is a general-purpose MIDI control surface, that could be used as, for example, a synth patch editor or as a mixing board for DAW software. It boasts a complement of nine "channels" and six tape transport function buttons. Each channel consists of a fader with a throw of about 40mm, an encoding knob, and two function buttons. This physical organization does not mean that the controls in a channel are restricted to related functions. Each control can be individually programmed to send any desired combination of MIDI Controller or NRPN messages, within any given range of values.

I was pleasantly surprised by how smoothly the faders operated. At this price point, you aren't going to get Penny & Giles quality, but the faders are as good as those on most semi-pro mixers and better than some. The physical feel is good, and the MIDI output is nicely linear without any dithering or jitter. The knobs feel a bit stiff by comparison, but I suspect they will get better with use. They too produced smoothly linear output. The buttons can be programmed to be either momentary (sends an "on" value when pressed and an "off" value when released), or toggle (sends an "on" value on one press, and an "off" value on the next press). They have a reasonably good tactile feel and I didn't notice any misses or double hits.

The left end of the Nanokontrol, showing the transport buttons, the first two channels, and the scene select button and indicators.

The transport controls are legended as being rewind, play, fast forward, return/cue, stop, and record. They can, however, be programmed to send any MIDI Controller messages you want, so they don't have to be used for the indicated purpose. They can also be set to send MIDI Machine Control (MMC) messages; however, in this mode, you get no choice about which specific messages the buttons send. That's fine for most stand-alone recorders and sequencers; most of them use the MMC messages as defined. Many DAW packages do not recognize MMC messages; you have to set up Controller messages to do the functions and then program the buttons to those messages. So the ability to program the buttons on the Nanokontrol to send Controller messages gives it an advantage over, for instance, most Akai MPC/MPD devices, whose transport controls can only send MMC. Each button has an internal LED that lights the button when pressed. It would have been nice if Korg had provided some means for turning these LEDs on and off by sending them MIDI messages, so that the controlled device could use them to indicate the current mode.

The Nanokontrol's internal memory holds four "scenes", or sets of configurations. Each scene can contain a completely different configuration. So, for example, you could set Scene 1 to send MIDI messages to control the mixer in your DAW; Scenes 2 and 3 to edit particular synths, and Scene 4 to be performance controls for a soft synth. The editor application can load and save scene sets from/to a disk file on the computer.

Nanokontrol verdict: An incredibly handy and useful multi-purpose MIDI controller. Great for almost anything you'd need a control surface for.

The Nanokey is a two-octave, velocity sensitive mini keyboard. To be honest, I wasn't expecting much out of the Nanokey based on previous experience with mini keyboards, but the Nanokey is surprisingly playable. The feel is not great but decent -- the keys do actually have enough travel that it doesn't feel like you are playing a hard touch surface. And they keep up with fast playing without missed notes. There are certain playing techniques, such as blues notes, that are difficult because of the layout of the keys -- the white keys do not extend between the black keys, and the black keys are not positioned noticably higher than the white keys. Then again, if the white keys had been done in the conventional manner, the black keys would be ridiculously thin. As it is, the keys are a good width for a player with big fingers (like me) to play without constantly bumping adjacent notes. That's no mean feat on a device this small.

The key velocity sensors are a bit sensitive; a soft touch is required to produce velocity values much below the maximum. Three velocity response curves are available, but I didn't notice much difference between them. A fixed velocity can be programmed. The key feel is a bit "clicky", somewhat like an old IBM typewriter keyboard; it's not unpleasant, but it may be disconcerting to players accustomed to weighted or heavily damped keys.

A small contol panel at the left end contains octave shift up/down buttons. The keyboard can be shifted up to five octaves in either direction, which means it can cover the entire MIDI note range. Two LEDs change color to indicate which octave the keyboard is on; it's a very effective system once you get used to it. Like many small keyboards, the Nanokey has buttons that simulate the action of pitch and modulation wheels. A pair of pitch up/down buttons can be set up so that when the button is pressed, the pitch bends up or down a configurable amount at a configurable rate; it goes back to zero when the button is released. The modulation button behaves similarly.

The left end of the Nanokey. The three LEDs above the buttons indicate octave shift down, octave shift up, and keyboard CC mode engaged. The "Korg" logo lights up when power is applied.

The final button on the control panel is the "CC Mode" button. When this is engaged, the keyboard sends Controller messages instead of note messages. One use for this is to "play" the value of a parameter, such as a filter cutoff frequency. You could set up the Nanokey above a conventional keyboard, and play a melody on it while using the Nanokey to "play", say, the cutoff frequency of a filter.

Nanokey verdict: Don't rely on this as your main keyboard. However, it's great for playing notes and chords to try out patches while you edit a synth, or step-entering melodies into a sequencer. It's fine for adding simpler backing parts to a song. It's easy to move around the studio. And as noted, you can use the CC Mode to "play" synth parameters while you play a melody on another keyboard.

The Nanopad is a drum pad controller with 12 finger-playable pads and an X-Y control surface a la the Electribe series. The pads are velocity sensitive and can be set up to play notes, generate Controller values, or sending program change messages. The X-Y surface can generate a Controller value for each axis, and can also generate Controller messages on touch and release, with a programmable "envelope generator" to contour the response.

There are two function buttons which work with the X-Y pad to produce effects typical of how drummers play. Pressing either the "Flam" (double stroke) or the "Roll" button engages that mode, but the effect will actually occur only when the X-Y pad is being touched while a drum pad is played. In the "Flam" mode, the vertical position on the finger on the pad determines the velocity of the second stroke. In the "Roll" mode, the vertical position on the finger on the pad determines the rate of the roll, and the horizontal position sends pitch wheel messages. When either Flam or Roll is engaged, the normal configuration of the X-Y pad is overriden. A third button, "Hold", has nothing to do with drum effects; it just holds the last position on the X-Y pad when you take your finger off the pad.

Like the Nanocontrol, the Nanopad holds four scenes, which must be configured using the editor software. A button on the panel selects a scene, and LEDs indicate which scene is active.

The Nanopad's control area and X-Y pad.

Unfortunately, this is as far as I can go with the Nanopad review, because the unit I received was defective: most of the pads are very erratic, and two of them don't work at all. Hopefully this is just an anomaly, and when I receive a replacement, I'll fill in some playing impressions.

Verdict: Review incomplete. Seems like a nice device, with very flexible configuration options.

Statescape Nevada

2010-01-01T21:58:00.002-06:00

A new Statescape, Nevada, is here. There's some info on the Web page; more here later.

The Bi-N-Tic is on board, more or less

2009-11-25T11:03:00.001-06:00

It's up and running, after some debugging. I expected some chaos, but not quite as much as I got...

When I first powered up the Bi-N-Tic, I got nothing out of it. Quick investigation showed that the VCO was not working. From reading a note on the Bridechamber page for this module, it turns out that there is a routing error on the board. Pin 1, the -15V supply pin, has no trace routed to it. Putting in a jumper to tie it to pin 4, which is also connected to -15V, solved that problem.

But when I fired it up and ran a signal through, it sounded terrible. There was major non-linearity and intermodulation happening inside the filter, with all kind of amplitude-modulation artifacts and distortion happening. Sending impulses through it revealed that there was a strong standing resonance at about 1200 Hz, which was impervious to any combination of control settings. It was this standing wave that appeared to be the source of the intermodulation. Plus, the frequency response of the filter section kept drifting around, and it didn't seem to have anything to do with the VCO.

If you look at the schematic of this filter, the audio section basically consists of four stages. Each stage consists of an opamp with some passive components on its inputs and in its feedback loop. The first stage buffers and combines the input with negative feedback from the second and third stages, which are the two actual filtering stages with the switching capacitors. The fourth stage buffers the output and filters out clock noise from the cap switching. To try to figure out what was going on with the filter, I tried eliminating all of the feedback paths. Turning the resonance path all the way down eliminates feedback from the second stage; unsoldering a 100K resistor eliminated feedback from the third stage.

In this configuration, I noted two things: the third stage output floated to a very high DC level, and it produced weird buzzing and clicking noises. And the output level varied as I touched things. By putting a 100K resistor between the output and the inverting input of the IC, I stabilized the stage and got rid of the DC offset. This led me to discover two bad solder joints. One was on one of the switching caps in the third stage, which accounted for the buzzing and clicking. The other wasn't in the module itself -- it was at a power supply return connection.

Now that I've fixed these things and re-soldered the resistor I removed, it's apparently behaving. I say "apparently" because I'm not really certain how it's intended to behave. I'm not sure what I expected when I ordered this module, but whatever it was, this wasn't quite it. The module's behavior is best described, I think, as a bandpass filter that has a comb-shaped response within the passband. It's highly resonant and close to self-oscillation most of the time. It's also possible to make the bandwidth very narrow -- so narrow that it can pick out individual partials from a square wave!

The final step was to make some panel graphics mods to correspond to the changes I made to the module (intentionally and inadvertently). I've done this before; I type/draw bits of graphics in OmniGraffle, print them on an inkjet printer, cut them with scissors to the needed shape, and then tape them onto the panel with transparent tape. Only this time, the "transparent" tape wasn't so much...

I'm not sure what happened with that. I used this same tape before on an MOTM panel, and it worked fine. Something is subtly different about the finish on this Bridechamber panel such that the tape doesn't make full contact with the surface. It sticks, but it's not all the way down, so to speak. Maybe at some point I'll try it again with different tape, but right now I don't feel like fooling it.

Here's another view, with different lighting:

And a close-up of the modified graphics:

Finally for some audio samples. First, some basic filtering of a square wave:

Bi-n-Tic Demo 1

Second, using the self-excite, with various selections of the excite waveform rotary switch, and the cap bank split/combine switch, in addition to playing with the resonance and bandwidth:

Bi-n-Tic Demo 2

Finally, a freak-out using excite and external sync from an LFO, in addition to square and pulse wave input. Note a few spots near the end where the filter is pushed up into the supersonic and produces some really strange artifacts:

Bi-n-Tic Demo 3

There's still some work to be done; the calibration of the FM inputs is way off (about one semitone per volt on the V/octave input). But I'm moving on to other projects right now.

The Hammond Novachord: Too Far Ahead of its Time

2009-10-24T23:28:00.007-05:00

There is a great thread taking place on VSE now concerning the meticulous restoration of a Hammond Novachord. What's a Novachord? Produced by Hammond in the 1939-1941 period, the Novachord is generally considered the first practical, mass-produced synthesizer.

Novachord undergoing restoration. All photos in this post are courtesy of D. A. Wilson.

A synthesizer in 1939? Yes, although herioc measures were required, and the instrument's reputation suffered for it. Surprisingly, the 144 vacuum tubes required for the note generation circuitry were generally not the problem. The Hammond engineers realized that the instrument would experience a tube failure every few hours if they designed the tube circuits per normal design practices of the day. They severely derated the tubes in order to increase the mean-time-between-failure from hundreds of hours to thousands (a technique that would appear again in the ground-breaking ENIAC computer a few years later), cutting the heater voltage from the nominal 6.3V to 5V, and limiting plate currents to a fraction of a milliamp. Here's what 144 tubes in one place looks like:

We'll get back to the reliability issue in a bit. First, let's go over the instrument a bit. Is it really a synth? Yes indeed, and a fully polyphonic one to boot; you can play all 72 notes simultaneously. Each voice has one oscillator that produces a sort-of saw wave, an ASR envelope generator, and what amounts to a VCA. (It isn't a VCA as we know it, in terms of actually feeding it a control voltage; the VCA and the envelope generator are both part of the same circuit. Nonetheless, it is an amplifier with time-variant gain, and it responds to the envelope generation and does what a VCA conventionally does in a modern synth.) Attack is controlled by panel controls, while release is controlled by pedals.

The filter section is paraphonic. There are three resonant bandpass filters, and several other filters for general tone control, that all of the voices feed into. The filters can't respond to the keyboard; they aren't voltage-controlled (that concept would have to wait another three decades for Moog and Buchla), so cutoff is controlled manually. The tradeoff is that they are true LC resonant filter circuits, and have a very definite effect on the sound. The overall effect is somewhat like a fixed filter bank on a modular synth.

A vibrato circuit serves more or less the same purpose as chorus circuits on modern synths. The vibrato is configured so that different notes are varied at different rates, which produces more of a chorusing effect when dense chords are played. The Novachord has a built-in amp and speakers (which unfortunately are underneath the case and aimed at the floor), and connections for external Hammond tone cabinets. (Which means that presumably a Leslie could be connected, although I haven't heard of anyone doing that.)

Here's a shot of the panel, with its stylish Bakelite pointer knobs. You can see from this panel that Hammond anticipated the ways in which synth control panels would be laid out decades later. In general, the controls are grouped with the filter controls on the left, and the envelope and vibrato controls on the right. In the photo below, two of the resonant-filter controls can be seen at the left edge, and the attack time knob is just to the right of the big downward-pointing knob:

Hammond made approximately 1000 of these beasts, which was actually a considerable number for the day, especially since production only ran for about three years. However, only a handful are known to exist today, and fewer still are in operable condition. Considering that in 1939 a Novachord cost considerably more than the average automobile, one might think that the people who bought them would have taken more care with them. One thing that might contribute to the few number remaining today is that the things are massive and heavy; no doubt some were abandoned (say, when the owner of a house containing one died) because they were so difficult to move. Another factor is that production was halted rather abruptly in 1942 after the U.S. entered World War II and some of the parts became restricted to military uses only; likely some Novachords were stripped for parts during and after the war.

However, another clue might be in this photo:

The resistors connected to that elegant key mechanism are carrying a whopping 270 volts. That was the plate voltage for the voice generating tubes. Which brings us to the actual waveform generation method used for the voices. The Novachord uses a top-octave division setup, but unlike top-octave architectures today, the octave division is not done using counter circuit -- that concept was unknown at the time. Instead, what it uses is a sort of hybrid between a top-octace divider and a VCO with a sync input. As mentioned previously, each voice has an oscillator which generates a decreasing (downward-sloping) sawtooth wave. The oscillators for the top octave are free running; their frequency is set by a resistor and a capacitor having very precise values.

Oscillators for the octaves below the top octave have a mechanism that will accept a hard sync input, but only after the oscillator has passed the approximate halfway point in its cycle. The signal from the next octave above is wired to the sync input. When an oscillator begins its cycle, it jumps to its maximum voltage and begins decreasing. When it reaches about halfway, the sync input (the waveform from the next higher octave) pulses, but the oscillator doesn't accept it yet. Its output voltage keeps decreasing until the sync input pulses again, at which point the oscillator will accept a sync input. At that point, it syncs itself to the octave above. Each oscillator syncs to the oscillator an octave above it.

It should be apparent that there are a lot of timing dependencies here, not only in determining the frequencies for the top octave oscillators, but in computing when each oscillator in the lower octaves has passed its halfway point and will accept a sync input. Unfortunately, the combination of the very high voltages and the paper capacitors that Hammond used (the only good alternative in those days was mica, which was prohibitively expensive) made the instrument's oscillator circuits very sensitive to temperature and humidity changes. Obviously, drift in a top octave oscillator would cause that note to go out of tune in every octave. But what was more distressing was "octave jumping", in which an oscillator would mis-sync, and not only jump that note to a different octave, but also the same note in all lower octaves, due to the way the synchronization was chained. Some sources also suggest that the fast-cycling relays in the vibrato circuits were prone to sticking and throwing the whole instrument out of tune.

It's actually rather interesting that the Novachord was created in the first place. Laurens Hammond was a music lover, but he was also a businessman, and he had originally gone into the organ business to create an additional market for the synchronous motors he had invented. But the Novachord contains no motors; in fact, other than the keys and the vibrato relays, it contains no moving parts at all. The circuitry is completely different from the Hammond organs, and there appears to be no parts commonality. Did Hammond really dream up a full-blown concept for a synthesizer in the 1930s, decades before almost anyone else? Maybe. It's a myth that Hammond was tone deaf; in fact, he had a keen ear for music and timbre, even though he himself did not play. He had learned quite a bit about tone and harmonics from the experiments leading up to the Model A, and perhaps the Novachord was what he saw as the next step down that path. Also, possibly, the Novachord was intended to be a technological pathfinder for future fully-electronic organs.

In any event, although apparently the company discussed an improved version of the Novachord to go into production after the end of the war, that never happened. And it would be quite some time before Hammond did anything in the direction of synths again. So Laurens Hammond left the electronic music world a rather strange legacy, an intriguing but peculiar footnote in synthesizer history. One can't help but wonder how that history might have been different had Hammond pressed the Novachord idea just a bit further.

Finishing the Bi-n-Tic, and A Couple of Oopsies

2009-10-21T23:23:00.005-05:00

I finished up the panel wiring last night. Here's the results:

Before I did the last of it, I reviewed the existing work and I found three mistakes. The first two were caused by my misreading of the CGS Web page on how to wire up the "excite" input, which takes divided-down waveforms from the counter and allows them to be injected back into the filter input. For some reason, I looked it and saw a four-position rotary switch, selecting one of the Q4/Q5/Q6/Q7 counter outputs, going to the switch contact on the external input jack. But in fact, it's a five-position rotary switch, and the jack is wired to the fifth position. So that will teach me to look more closely. I had to re-do the wiring of the jack, and of the first position on the rotary switch (fortunately, I had not soldered the rest of them yet). I was worried about the rotary switch since it's a plastic bodied type, but it appears to have survived. Here's how that came out:

Note all of the unused contacts. This is actually a 12-position switch, but I've got the stop set at position 5. To the right of the switch is the pot that controls the exite input level.

The other big booboo I made was that I put the banwidth pot in the panel where the resonance pot is supposed to be, and vice versa. In the picture below, the big double-gang pot (the bandwidth pot) is supposed to be at the far right:

That wouldn't be hard to fix except for one thing: remember me talking about those locator pins on the pots, a couple of episides back? Well, the big pot has its locator pin in a different place than the small Alpha pot. So in order to put these two controls where they are supposed to be, I'd have to drill two additional holes in the panel for their locator pins. At this point, I really do not want to be drilling on the panel and getting metal shavings all over the board. I can still detach the board, but the wiring makes it difficult to get it more than a few inches away. So instead, I'm going to "fix" it by putting new legending on the panel. If reality doesn't conform to one's pre-conceived notions, then change reality!

I waited until the end of the build to insert the ICs into the board. I wound up wishing I had done that before I did the panel wiring; some of the socket locations were difficult to get to, and I bent a few pins. But eventually I got them all in. I'm glad I took a photo of the bare board before I started assembly, because with the sockets in, it was impossible to see the silkscreening that shows which IC goes where. But with the photo, I was able to figure it out. The kit substituted TL071 opamps for the LF356s indicated on the silkscreen; apparently the LF356 is no longer available in a through-hole version.

The photo below shows the pads for the panel wiring to the resonance and bandwidth pots. The silkscreen didn't indicate which pads were supposed to be connected to which terminals on the pots, so I had to follow some of the traces away from the pads to see where they went, and compare to the schematic. Here's the photo:

The three pads at the right (note that we're looking at the board upside down, relative to the silkscreen) go to the resonance pot. As near as I can figure, they go to the pot's terminals in the same order as they are on the board: left (blue) goes to the left terminal, center (yellow) goes to the wiper, and right (orange) goes to the right terminal. We'll see when I try it; it may be that I've wired it so that the pot works "backwards". If so, I'll switch the left and right at the pot. To the left of these pads, there are two pairs of pads which go to the two gangs of the bandwidth pot. In each pair, the one on the right goes to the wiper terminal; the other goes to the left terminal. (I've connected the wiper and the right terminal together, as shown on the schematic, but I don't really know that that does anything.) Note the two pads to the left of the bandwidth pads, labeled "in". These are filter inputs that don't go through the input level pot. (The one that does go through the input level pot is elsewhere.) I've used one to connect the exite signal from the excite level pot, and the other is unused.

Here are a few more detail shots of the assembly. I wound up running four grounds. The filter signal input got its own ground. The other inputs share a ground, and the outputs share a ground. The jack wiring:

The grounds connected at the pads that are intended for a Eurorack 16-pin power connector. Since I'm building this for a 5U configuration, I'm using the MOTM-style power connector above and to the right, and the six ground pads on the Eurorack area made a handy place for bringing grounds to. Note there are four here; the three I mentioned above, and a fourth that goes to the excite input level pot.

The switch that I installed for the capacitor switching mux mod:

Finally, I put the knobs on (not indexed yet, just sitting there) to show what it's going to look like when it's finished:

Bi-N-Tic Filter: Panel Assembly and Hackery

2009-10-14T22:32:00.010-05:00

Didn't turn on the soldering iron tonight. This was the night for mechanical work on the panel. The first thing I had to do was decide what to do about the locator pins on the pots: cut them off, or use them? Using them meant drilling additional holes in the panel. What I decided to do was cut off the pins on the pots that are soldered to the board (they aren't going to rotate anyway), but use them and drill the additional holes for the pots that are panel mounted. Here's how it came out:

Another angle:

This doesn't show from the front; the knob body covers the hole:

So I went ahead and drilled the rest of them. Holding it up to the light:

Note the butchery on the second from the left. This pot is a double pot; two on one shaft, and I neglected to note that it has a larger body and so the locator pin is further away from the center. Fortunately, the knob will still cover the mess up. The hole in the very center is the one I added for the switch that will select which outputs from the counter will be routed to the capacitor-switching muxes.

I had been a bit up in the air about what knob I was going to use on the rotary switch, so I test-fitted the switch and knobs to the panel. Here are the candidates. First we had the knob that came with the kit; it is of course popularly known as a "chicken head" knob. It's a style from the 1950s. I like retro-modern as much as the next guy, but to be honest, I just don't know if I care for the chicken head knob on this panel.

Second up, we have a knob from my knob collection. I don't know where this came from, but it is closer to the style of the other knobs. However, it's rather large, and it covers up some of the panel legending above the jacks below the switch:

Here's an interesting one. It is one from a Radio Shack two-pack that I bought in 1980 for a guitar repair job. As it turned out, I only used one, and this one has been in the original baggie since then. Here it is:

And, just for fun, the bag it came in:

Yes, it says $1.89, and that was for two. Ah, those were the days. Of course, in 1980, I was a starving college student, and 1.89 was a week's worth of Kraft Macaroni & Cheese. Anyway... what I'd really like is the style of knob that Synthesizers.com uses on its rotary switches. Here's an example:

So now it's time to assemble the pots:

A close-up the resonance and excite pots. Note that the shaft of the resonance pot is shorter:

Next step was to get the rotary switch installed. I discovered, somewhat to my surprise, that it also has a locator pin:

At first I couldn't figure out how to set the rotation stop limit -- it's a 12-position pot, and the circuit only uses 5 positions. Then I realized: the way to do it is to pry up the little washer that you see at the very base of the shaft housing. It has a little tab that sticks into a hole, and that is what sets the stop. You pry it up, rotate it to the proper position (they are marked on the body face), and then push it back down.

Having done that, I decided what the heck, and I went ahead and drilled a hole in the panel instead of cutting the pin off. I was concerned about whether or not the knob body would cover the hole, due to the radius of the switch body, but I went for it anyway. Here's what it looked like as I was trying to install the switch -- I had to drill the hole out a couple of times with progressively larger bits in order to get it to fit:

Sure enough, the body of the chicken head knob didn't cover the hole:

But that was OK since I didn't plan to use it anyway. The body of the aluminum knob covered it just fine:

Next step was to fix the paint abrasion that I caused when I drilled out the hole for the new switch. This is easily done with a bit of black modeling paint on the tip of a paper towel:

I didn't put all the knobs on yet, but I did one to see what it will look like. From the front:

And from the side:

The bottom flange of the knob is up off the paned about 3/16", which seems a bit much. But I checked with some other modules I have, and they are all in that same range. I really don't want to have to get out the hacksaw and cut 1/8" off of the shafts, although I am concerned about the resonance pot with its short shaft. It'll look funny if the resonance knob is right down against the panel and the others are all standing off of it this much.

The last task for the night was to cut the locator pins off of the pots that are soldered onto the board. This turned out to be easier than I thought; applying sideways twist with a pair of needle-nosed pliers snapped them right off. The results:

And a close-up -- look at the right edge:

Next: install the jacks and start wiring up the panel.

Bi-N-Tic Filter: Panel Wiring

2009-10-13T22:44:00.003-05:00

I just finished wiring all of the panel leads to the board. Here's the results:

If you look at the right edge of each of the four pots, you can see that they have a locator pin. I'm still trying to decide if I want to use them, or cut them off. If I use them, I have to drill seven additional holes in the panel. Lot of trouble. On the other hand, if the locator pins are used, it pretty much assures that the pot bodies won't rotate behind the panel and mess up the knob indexing.

For my mod, which will provide for passing the Q4 instead of the Q3 bit from the counter to the most significant bit of the muxes, I had to solder a wire directly to that pin of the 4024 counter IC.:

It's a bit hard to tell from the photo, but that pin of the IC is bent outward and is not in contact with the socket. The yellow wire soldered to it will go to the panel switch for this mod. The output from the switch returns as shown in the next photo:

The blue wire brings the signal back to this otherwise-empty hole under the pin socket that I broke off.

One of the few things I don't like about this board is that the various points on the board, where the pads are for the wiring to/from the panel, don't have ground pads in the vicinity. This means, among other things, that you really can't use twisted-pair wiring. In fact, grounding points are kind of scarce. The only convenient place to bring out grounds is to use the set of pads that are intended to be used for a Eurorack-style power connector. It has six pads connected to ground. I've soldered in two wires, with the idea that one will be the ground for the input signal jack, and the other will be the ground for all of the control and output jacks. But I'll look at again when I put together the panel. Here's the ground wiring; note the MOTM-style power connector above:

Continuing on the Bi-N-Tic Filter

2009-10-07T22:20:00.003-05:00

All of the parts are stuffed, except for the socketed ICs that haven't been inserted yet. Here's how it looks:

That blob in the upper right corner is where the VCO's expo converter is; it consists of an integrated matched pair with a tempco in contact with the top of the package. Here's a close-up:

The white stuff is heat sink compound, which should improve the thermal contact between the IC and the tempco. The board silkscreen calls for an LM394 here, but the kit came with a SSM2210, which caused me a bit of confusion until I went back and looked at the Bridechamber writeup for this module -- it specifically mentions this substitution, as having been done for improved tracking.

This gap-toothed socket is where the 4024 counter will go:

That's pin 9 that is broken off, which is the counter's bit 3 (8's) output. I am going to add a switch that will allow that line to be switched to either bit 3 or 4 (16's), which will alter the scanning pattern of the capacitor banks and should produce interesting results, I think. I'll bend that pin of the IC out so it doesn't go into the socket, and tack a lead wire directly to it to route to the switch. The return from the switch will come back to the un-contacted pad, to be routed to the cap switching muxes. Here's the underside, showing the as-yet unsoldered pin 9 pad:

Next step is the panel wiring. I'll solder in MTA-100 headers for some of it, but there are some places where there won't be enough room for the connectors.

CGS Bi-N-Tic Filter

2009-09-30T21:13:00.003-05:00

Well, the Super Psycho LFO rebuild / hot rodding is on hold for a couple of weeks. The issue: Each of the six oscillators has a 1M pot in the circuit that controls the osc rate. After thinking of various ways of combining the pot in series or parallel with a Vactrol's light-dependent resistor, I've decided that I'm not happy with any of them. There's just no good way to do it such that there isn't some value of the pot setting that makes the Vactrol ineffective, and vice versa. The right way to do it is to take the pot out of the oscillator circuit and have just the Vactrol in the circuit; then make the pot be part of a voltage divider that will supply an offset voltage, which is added to the external control voltage via an opamp voltage-adder circuit, and the sum of the voltages drives the Vactrol's LED. Problem: Although I can make a 0-5V voltage divider using the existing 1M pots, it's a bit iffy as far as staying well clear of the opamp's specified offset and bias currents. I'd feel more comfortable using a 100K or 50K pot, so it can supply a bit more current. But I don't have any. I'm in on a group buy at Muff's place, but it's going to take a few weeks for that to come together. So until that happens, the Super Psycho is in a wait state again.

In the meantime I have something else to play with:

This is a kit for a Ken Stone / CGS 57 Bi-N-Tic switched-capacitor filter. The kit is from the wonderful folks at Bridechamber and includes all parts. A close-up of the circuit board:

And the panel:

The kit incorporates nearly all of the mods and hot rodding that Ken calls out as options in his notes on this module. (The control that Ken labels "DAMPER" in his notes is called "RESONANCE" on this panel.) Switched-capacitor filters are very unusual in the synth world. I'm not quite sure why that is; they have a reputation for being noisy, but I think that rep might be the result of some early designs of the switched-capacitor concept that used mechanical commutators and crummy caps. I'm curious to build this and see how it sounds. I am going to add one mod: a switch to take the most significant bit of the digital count that goes into the multiplexor which selects one of the eight caps, and move it from the 4-bit to the 8-bit of the counter. That will have the effect of dividing the eight caps into two banks; instead of scanning all eight caps in order, it will scan over a bank of four caps twice, then switch to the other bank. I'm not quite sure what this will actually do; I think it will give the filter two resonance peaks. Anyway, it will be interesting to find out.

The Wrong Vactrols

2009-09-20T10:58:00.007-05:00

UPDATE: I've posted a video demonstrating the Vactrol turn-off delay.

So I was all set to prototype the control voltage input circuits for the Super Psycho LFO rebuild. The basic idea is to feed the control voltage to the LED half of a Vactrol, and put the light-dependent resistor (LDR) half of the Vactrol in series with the rate pot of the specific oscillator (so there would be 6 Vactrols total). I have some VTL5C7's that I've had laying around from a group buy I participated in on Synth-DIY several years ago. I've never done anything with them.

So I went to the data sheet to find out how much current would be required to make the LDR vary between about 0 and 100K, which seems like a good operating range. (You don't want to get into the high resistances because the data sheet says the response is not specified above 1M or so.) And I saw these curves:

Note in particular the turn-off curve. If I want to bias it to operate between, say, 1K and 100K, then the turn-off time from full on is about half a second! Obviously, if you feed the input something like a 10 Hz square wave, the output is going to flatline. Never mind modulating at audio frequencies.

I know this is a fundamental characteristic of LDRs. However, Perkin-Elmer does make different types. On the same data sheet are the specs for the VTL5C6. Its turn-on and turn-off curves look like this:

Note the difference in the time scale. This one has a turn-off from 1K to 100K of 2 milliseconds. That's more like it! Trouble is, I don't have any of these. My usual go-to parts source, Mouser, doesn't carry Perkin-Elmer. Google showed me that Allied carries them, and I checked and they have hundreds in stock. So that's on order, and I expect them Wednesday or Thursday. Until then...

Rebuilding a Super Psycho, Part 2

2009-09-13T23:15:00.000-05:00

OK, I know it's been a while, but I ran into some employment difficulties. But that's all resolved now. Anyway, I've been working on the Super Psycho rebuild. The first step, as I documented in the previous post, has consisted of moving everything from the circuit board that was falling apart to a new board. Below is a photo of what I've gotten done:

A few things to note. I replaced all of the resistors along the top of the board with new ones from my stock, mainly because desoldering the old ones was going to be such a pain; that was an area of the old board where I'd had to add a lot of jumpers and kluges. Most of the capacitors were salvaged from the old board; I tested them all first. The two green 100 uF caps at the left are new from my stock because the old board was an earlier revision and it didn't have these. There's also a 10 uF cap above and below these; I replaced the one at the bottom because the original was borderline (measured 8.1 uF vs. a specified tolerance of 20%).

I've installed MTA-100 headers everywhere that panel wiring will attach, a la Dotcom. This should help considerably with the soldering mess that the old board had along the top edge, where 42 individual wires were soldered in. Look, it's much cleaner now:

(This photo was taken prior to washing, which is why it appears to have a lot of excess flux; it does. I used organic solder for this part.)

I wound up replacing the transistors and the IC sockets. Really, trying to salvage sockets just isn't worth the trouble; I destroyed one of them during the desoldering, and even though I got the other two out, they had distortions that made them very difficult to insert into the new board. Sockets don't cost much. I also decided to replace all of the transistors after two of them had the base leads break off while I was stuffing them. The replacements aren't the same type; the originals as specified by CGS were BC557, while the replacements are 2N4403. (Why those? Because that's what I had on hand.) All they do in this circuit is drive the LED indicators; almost any PNP type should work for that.

So the basic board is almost ready to go. There are four 470K resistors that go on the board just to the left of the 8-pin socket at bottom center in the first photo. The originals were butchered and I didn't try to salvage them. These resistors form a passive mixer for the four oscillators on the board that don't have the switchable waveform. I want to take those signals off the board at this point, so I need to get new resistors that I can stand up and have a long lead to solder a wire to. This is in pursuit of one of the two mods that I've decided to make: adding a second output bus with mix capability. Below is a block diagram of what I have in mind:

There are six of these, one for each individual oscillator. I'll pick off the four non-waveform-selectable ones from the board as stated above, and I'll intercept the other two at the waveform selection switch common terminal on the panel. The signal will come into this block at left. Each will have a pot to mix it to the B bus. There will also be a switch that allows the signal to be removed from the A bus (the on-board bus). To save some panel space, I decided to use pots with pull-out on-off switch capability to implement the A bus switching. I wasn't going to do that originally because of the cost, but I found some inexpensive ones from All Electronics. Only problem: The switch is SPST and the "on" state is when the knob is pulled out. I want the opposite sense, so to create it, I'm going to invert the switch signal and then use it to switch a bilateral switch that will feed the signal back to the A bus.

The other thing I'm doing, which I haven't drawn yet, is adding control voltage capability. I'll do this by putting a vactrol in series with the resistor that determines the frequency for each oscillator. There will be two CV inputs, an A and a B, and for each oscillator there will be a switch that switches it to the A input, the B input, or neither. I haven't decided yet if I'll make any attempt to compensate for the non-linear response of the vactrol.

I'll build all of the new circuitry on a piece of stripboard. I'll make an auxiliary panel to hold the bus mixing pots, the A/B CV input switches, and the extra jacks. I haven't figured out yet how I'm going to do this. One possibility is getting a blank MOTM-format panel with studs on the back for a board mounting bracket; that would allow me to mount the stripboard to the auxiliary panel without any screws coming through the front. Another possibility is using JB-Weld to glue standoffs to the rear of a panel and mount the stripboard that way, Dotcom-style. The third option is simply to mount the stripboard on the bottom inside surface of the case, behind the panels. However, that would make it harder to move the combined modules later.

I had to order some more 470K resistors from Mouser to replace the ones I messed up. I also had to order additional 100 nF (0.1 uF) polyester box caps; they are used for decoupling, and the rev B board uses more of them than the old board did. I've got these now and I'll be installing them next week.