I am so bad at gift wrapping. I think I inherited that from my dad, who is not above using cardboard tubes, newspapers and duct tape to get the job done. I failed this evening at wrapping a perfectly rectangular package and had to throw the paper out and start over.
I’m doing a better job with the album mastering…. except, it turns out, I’ve been doing it wrong.
MusicTech magazine’s current issue has a feature about mastering. I read it, and most of the advice is on the order of “use this $4000 worth software and these $3000 monitors” and uh, no thanks. But I did learn that editing the beginning and end of a track is “topping and tailing”, and that electronic music technology magazines in 2018 are pretty much overpriced garbage.
I got more specific, up-to-date advice from the first website that popped up on a Google search. It turns out that in general, you should meter in LUFS (“Loudness Units relative to Full Scale”) for loudness and dBTP (decibels True Peak) for peaks. Nobody thinks you should compress heavily to make your music as loud as possible, because many streaming services normalize everything to the same volume level anyway. And while I was being relatively gentle with my own work compared to the previous album, I was still going beyond recommended levels.
I’d been ignoring metering plugins because there’s nothing more boring than that, and I assumed dbFS peak and RMS as shown in Sound Forge were good enough anyway. But the free version of Youlean Loudness Meter shows the relevant info and how I’m breaking the rules. (-23 LUFS is a European broadcast standard; -14 seems to be a common goal for streaming audio but the important thing there is more “don’t over-compress”). And -1 dBTP is a recommended peak maximum so that MP3 converters don’t accidentally cause clipping.
Of course it would have been smart to do this research before “nearly finishing” all 11 songs. In a lot of cases I think I can just turn it down and be fine, but I’ll double-check I didn’t compress too much.
Sound Forge Pro 10 has been crashing on a semi-regular basis, and it’s a few years old now. I’m happy to see that it’s not abandonware and there is a new version — though Sony (having bought it from Sonic Foundry) sold it to Magix. Unfortunately, the demo crashes immediately on startup. I can use it okay after that as long as I never close the bug reporting window, but it doesn’t say a lot about the potential stability, so I’m not sure I want to pay for an upgrade. Maybe I will look for another tool in the future, though I do like Sound Forge’s dynamics tool and the ease of crossfading every edit.
A fancy hearse pulling off the road into a self-storage facility. Please tell me that doesn’t mean what I think it means?
A church with the name “Anchor of Hope.” Because when I think of a nice uplifting symbol of hope, I think of chaining myself to a heavy piece of steel half-buried in the bottom of the sea.
An oversized pickup truck with not just a TR*MP bumper sticker, but a McCain/Palin bumper sticker with the McCain half removed. This, mi amigos, is how you locate the most petulant member of the Orange Face Cult for miles around.
I’ve mentioned I’m in the process of mastering my fifth album of the year. But what is that, really? Or what is it to me?
What it used to mean was the preparation of a “master” copy of the final mix, to be duplicated — almost like a mold for casting. For CDs and DVDs, there’s a digital file of course — but for large-scale duplication, a physical glass master is prepared in a cleanroom with a laser burner and a nickel deposition process, and then a “mother” is created as a sort of negative of that, to stamp pits into the actual CDs.
Mastering requires making some adjustments to suit the limitations of the medium. For instance, if the difference in bass content between the left and right channels on a stereo LP is too great, it will throw the needle right out of the groove. Digital media have their own limitations, and some master for specific sound systems in clubs. “Mastering for MP3” or “for iTunes” might be a little snake-oily, but certainly earbuds or headphones are a different sort of target than a big speaker system. (Generally, I use headphones throughout the whole process, including as my mastering target.)
Historically, recording engineers found this was the best time to make adjustments to the final mix as a whole, so it sounds as consistent and appealing as possible. That generally means having a nice balance in different frequency bands, but mostly it means means loud.
Quiet recordings are more susceptible to noise, from random particles and errors in the medium to cosmic rays and other interference getting amplified along with the music. Also, louder music generally sounds “better” than quiet from a psychoacoustic standpoint. Some stereos have a “loudness” button which fakes a louder sound by changing the curve. But too much loudness causes distortion.
Certain kind of distortion sound great. The sound of the electric guitar is dependent on it. Different kinds of distortion are involved in synthesis. Saturation involves nice smooth curvy distortion that sounds “full” and “warm” if it’s kept subtle enough; you can get that by recording to tape a little bit louder than it was designed for.
But distortion can definitely be undesirable, too. There’s a reason why chords on electric guitars tend to be very simple, such as the open fifth “power chord.” Distortion creates more harmonics in the signal, and if the harmonic relationships are already complex going in, what comes out will be mushy and gross (technical term). And a too-loud digital recording is subject to “clipping”, where the peaks of waves are sheared off in a flat, sudden way that is very inharmonic and does not sound natural or organic at all.
Dynamics are important — the balance and change of quiet and loud over time. Dynamics in playing style creates drama, and is an important element in groove. Many instruments, such as drums, are highly dynamic in themselves. But excessive dynamics in a recording can be annoying (when you constantly have to adjust the volume to hear clearly) and cause technical challenges (too quiet overall, subtle details are easily missed, or the recording gets too loud at times). Often to make a recording louder and more balanced overall, the engineer has to reduce the dynamics through compression and/or limiting — usually in a way that doesn’t noticeably sound like the dynamics have been changed or anything has been lost — as well as “riding the gain” more gradually.
The actual dynamics in a file can include all kinds of weirdness we don’t perceive — lots of little spikes of volume that our ears and brains just smooth right over. That’s why these tricks can work. Both compression and limiting basically just turn down the volume as the signal gets louder, and back up as it calms down — but the devil is in the details. At what level this attenuation takes place, how smoothly or suddenly it applies on a volume scale, how quickly it applies on a time scale, and so on. It’s part science and part art.
(Don’t confuse dynamic compression with the kind of compression that makes an MP3, WMA or OGG file smaller than a WAV file. Lossless audio compression uses algorithms to represent the same data in less space, and is guaranteed to sound exactly the same as no compression. Lossy compression removes data that contributes little or nothing to what we can actually perceive, and is generally a compromise between size and perfection. Blind tests on thousands of listeners have shown that on average there’s a barely discernible difference between a 192kbps VBR MP3 and a CD it was ripped from, and nobody can distinguish 320kbps from the real thing.)
If you lower the relative volume of the spiky bits, you have more room to turn it up overall. There was something of an arms race or “Loudness War” which reached its peak (so to speak) in the mid 2000s, with Metallica’s Death Magnetic frequently cited as one of the most egregious examples. Things have calmed a bit since then.
There’s also equalization (EQ) — this is the raising or (more usually) lowering the volume of particular frequency ranges to get a nice, balanced, full sound. This can be combined with dynamics processing in tools such as dynamic equalizers and multiband compressors.
Of course both EQ and dynamics can be used for “creative” effects as well; it’s common to compress drums more than is strictly natural-sounding, or to “squash” a singer’s voice into a narrow, telephone-like or old-timey-radio range, or to really bring out the breathiness in a voice or squeaks on a guitar fingerboard, and so on. Usually that’s done as part of the mix rather than mastering, though.
There are a lot of tools out there to help with mastering. Some plugins or services promise to do it all automatically with a single button or knob, and usually that’s better than nothing. I have a whole process and a set of tools I use.
I try to get levels reasonably okay in the original recordings, with the compressor/limiter ToneBoosters Barricade. I don’t push it very hard at this point because I won’t be able to undo it. The idea here is mostly to keep any unexpected spikes from clipping, and having a good monitoring tool to make sure I’m not recording too quietly with my headphones turned way up or vice versa.
My first pass at editing in Sound Forge Pro does only a little dynamics work to get levels generally okay — it’s mostly about overall sound, good first and last notes, and so on. I save the more strenuous mastering work for a separate step.
Sound Forge has a few built-in dynamics tools. There’s “normalize” which can raise everything to within a certain threshold, either by peaks (safest) or RMS (useful for general “perceived” loudness but risks pushing the peaks too far) and is good at reporting maximum peak and average RMS levels to compare the different songs on an album. There’s a fantastic graphic dynamics tool that lets you draw the response on a graph, and you can compare to levels shown in a recording. There’s a “clip detection and repair” tool that’s a kind of gentle compressor that lowers peaks to safer levels. And sometimes I highlight a section and crossfade into and out of a general “volume” tool to raise or lower the volume in a specific area.
I use other plugins with Sound Forge as well. u-he Presswerk is a full-featured compressor that goes a bit beyond my pay grade, but I have some standard favorites among its presets. I’ll almost always try “A Touch of Glue” and/or “AF Master Transparent” to see if either of them brings out subtle details and reigns in peaks a bit, but sometimes neither of them really helps. Undo is just a click away. The aforementioned Barricade is also good to try for a big boost; it can produce what looks like clipped-off peaks but in practice are carefully set to sound clean while maximizing overall volume.
I don’t do a whole lot of fiddling with EQ in mastering. Sometimes I’ll decide that if I cut out some sub-bass I’ll have more room for everything else, or that a particular note or frequency band is a little too intense. Sound Forge has a good graphic EQ (for more general changes) as well as a parametric EQ (for surgical edits to specific bands). Sometimes I want to reduce the strongest frequencies a little bit all across the file, whatever they may be, to enhance the timbre and make it “howl” a bit less — for this I use Melda MSpectralDelay‘s level transformation tool, being careful to disable the delay, spectral panning, and frequency shifting first.
EQ changes the dynamics, and often it’s best to cycle between different tools, make small and gradual changes, and keep getting feedback from one’s ears and the various measuring tools in the software.
Write drunk; edit sober.
— not Hemmingway, who wrote in the mornings, avoided alcohol until the afternoon, and was to avoid hangovers. It was Peter de Vries, and was not meant literally but to encourage both “spontaneity and restraint, emotion and discipline.”
Between the ultra-close attention this process demands, and the changes to dynamics bringing out more detail, it can expose flaws that were previously unnoticed. I suspect that sometimes the Firewire connection between my audio interface and computer gets a little overwhelmed, and there are any number of other things that can find their way into a recording. Usually it’s just subtle quirks of the modules and effects I’m using, or sometimes I pushed something a little hard for effect and got more than I bargained for. I accept a certain amount of this as a part of the process and the charm of working this way, and I’m sure Tony Rolando would agree. Sometimes I even bring these “flaws” out intentionally, such as enhancing background noise through manipulating dynamics and EQ — or creating the noises intentionally via modular or plugins.
But other times I want to repair things. Smoothing them out is rarely as easy as using Sound Forge’s “Clicks and Crackles” automatic tool, which has a penchant for making things worse. Sometimes I just need to zoom way in and literally draw a smooth curve over where there was a sudden jump, an edit affecting the tiniest fraction of a second. Or it might require some careful copying and pasting from another part of the file, being especially careful to keep the transition smooth, or just cutting out a tiny bit and stitching the edges together. Reverb can smooth things over so long as it doesn’t cause a sudden shift in timbre, or it’s done in an intentional-sounding way and fits in with the busy things that are already present. Sometimes mixing in something else will help mask it. There really are no hard-and-fast rules, and this bit can be time-consuming, but persistence usually pays off.
One of the goals of mastering is consistency in volume levels across an album, and generally in line with other music of a relatively similar nature. My goal is to get them where Sound Forge’s Normalize tool reads about -0.3dB peak and -10.5dB RMS. I wouldn’t read too much into those specific numbers though, because other tools are likely to report differently. (0db is the maximum possible level in a digital recording; “bigger” negative numbers are quieter.) That allows a little bit of room for the playback device to hopefully not clip, and seems to match the volume levels of other modern albums. I’m not too worried about a little deviation here, as long as it doesn’t sound so different from one song to the next that you want to reach for a volume control too frequently.
This sort of work can be tiring to the ears and mind, so I break it up a bit, get the headphones off and reset myself. Mastering Materials has seemed particularly grueling so far, but of course I hope the result is worthwhile.
I don’t know how many people are reading this now and how many might start following it later, but I find writing for other people helps me get my own thoughts in order. So, thanks for being my rubber duck.
I started writing out a much longer and more boring post but then killed it off as things congealed. In a condensed form, here are my hopes for “version 1.5” of my setup in the next few months:
A Sound Of Thunder goes. The space is reserved against other uses.
Field Kit FX and its reverb tanks go — replaced functionally by Chase Bliss Dark World (shipping at the new year) and u‑he Twangström (currently in beta and f**king brilliant).
The fate of EarthQuaker Afterneath depends on how I feel with Dark World on the board and if I think something else would be more useful/fun.
Reclaimed pedalboard space goes partly to a 16n Fader Bank, which is an open-source controller that works with MIDI, CV and i2c (used by Teletype). This should be amazing for sequencing in Teletype and controlling everything else. There’s no current ETA on a production run for it.
The rest of the space could go to a 4ms Pod or lunchbox case, giving me 32-60 more HP for Eurorack modules. This is contigent on other module vacancies/replacements — if I don’t do it I’m likely approximately cost-neutral with all these changes. Strong contenders for space include the successor to Mutable Instruments Clouds or a Qu-Bit Nebulae V2.
Of course another pedal or two could make it to the pedalboard, depending on space. But there are no really strong candidates at the moment given what Dark World is likely to do for me.
Mimetic Digitalis and/or Maze get replaced. (Maze is about to receive an update adding a feature I asked for which may be a game changer.) But I’m expecting great things out of u-he CVilization, which is the size of MD and is a 4×4 matrix mixer with several sequencer-friendly features, plus three other modes that may make it indispensible. It’s possible I’ll replace my A-138m with one also.
If CVilization or the 16n don’t come to fruition or I find them lacking, there are several other options. For sequencing: Befaco Muxlicer or Pittsburgh Micro Sequence. For control, Noise Engineering Lapsus Os or Michigan Synth Works Plancks II. For matrix mixing, the 4ms VCA Matrix or Rebel Tech Mix 04.
If Cvil and 16n and “Clouds 2” are all utterly brilliant, I see myself with about 19HP of free space with no claim on it, without adding a small case. That’s my hope.
Since I don’t love the 2hp Trim for pedal conversion, it’d be nice to put in something better suited; it wouldn’t take much more space anyhow.
I have no plans to add more VCOs even if space becomes available. If I want to change anything there, something’s got to go. Double Helix or Hertz Donut perhaps, but I like both of those so it’s not highly likely.
I’ve just submitted two songs to the Ambient Online “Fire” themed compilation, which should be released late this month or in early 2019.
I have one called “Electrostatic Dust Fountain” in the previous compilation, The Moon.
Cover art for Materials is done. I need to jump into mastering it, but with my process that doesn’t take too long. (If I sold enough to justify it, I would definitely look into professional mastering, probably checking with Obsidian Sound first. But I think I do pretty okay at it.)
I expect the next project will be an unthemed album, and there is no ETA at this time.
Having written up my modular system as it currently stands, I’m thinking about shaking things up a bit for a version 1.2, or 1.5 perhaps. More on that in another post, probably, once I’ve worked out some plan versions.
I’m not that much into TV, but sometimes get caught up in whatever my spouse is watching. The latest thing is The Great British Baking Show. All too often it makes us want desserts, and I’ve had a couple of dreams about it.
Bee and Puppycat is a mostly excellent, weird, cute, occasionally creepy show. The whole first season (if it can be called that) is a little over an hour. Marina Sirtis and Ellen McLain have minor roles in it. Here’s the pilot and here’s the rest. There will be more in 2019.
Steven Universe, my favorite show, is finally going to end its 4-month hiatus on December 17. It’s a Christmas miracle! It feels like it could be the end of the whole show, except that we’ve been told it’s not.
I was kind of into The Expanse and need to remind myself to look for season 3.
All but one in this final row are by Mutable Instruments, and all but three are 2018 releases. Three songs on the upcoming album were done on this row alone.
Starting at left, Mutable Instruments Marbles is a humdinger. The left half generates three trigger streams, from a steady clock to a jittery one to drum patterns and random variations; it can use its own timing or follow an incoming clock or learn rhythmic patterns. The right half generates random CVs synchronized to the left half or an external clock; it uses a clever quantizer that filters notes by probability, which you can train by playing pitch CVs into it. It can also sample CVs from an input and distribute them according to its clock timing and shift register logic. For each side, you can engage “Deja Vu” which plays the rhythms and/or CVs in a loop, with an adjustable probability of altering that loop each time it plays through. There’s also another random CV generator just for kicks. It may sound like a lot of complex stuff, but it’s mostly easy to use and the results are fantastic — it made me change my mind about modules that generate random signals. To me it’s a great partner for Teletype. 5/5.
A shift register passes a value along a chain each time it is triggered, in “Row, Row, Row, Your Boat” fashion. Some variations on shift registers feed back into themselves in order to generate repeating or chaotic patterns; linear-feedback shift registers (LFSR) were used in the 80s to generate noise for arcade games.
Sporting six sliders, Stages is a very versatile modulation source. Each segment can be set to ramp, hold, or step behavior, and segments can be grouped together by patching in gate signals. In an intuitive way you can patch simple or complex envelopes, LFOs, sequencers, sequential switches, delays, sample-and-holds, slews, manual sliders, and combinations of those. You can run audio through it to make it a little grungy and/or filter it, or generate chiptune-like audio. You can chain multiple Stages together, or to itself for the “Ouroboros” easter egg mode, a harmonic oscillator (I haven’t tried that yet). Extremely cool and very well designed, especially good for smaller systems but useful everywhere. 5/5.
Plaits is the successor to Braids, the company’s first big hit in the Eurorack world. A digital oscillator with 16 different synthesis models, each with three parameters, and many having cool variants on the secondary output. It also has a built-in decay envelope and LPG, so it’s possible to use it with a minimum of other help. As I’ve said previously, it’s a very good partner for Rings. A very solid module. 4.5/5.
Fourth and fifth from left I have a pair of Rings, the module that got me into all this. I’ve mentioned it before. I would not have it as the only sound source in my system (that would be Plaits for a tiny setup, or E370 for less small) but I do like it a lot. At least for Materials, it was kind of awesome having two — time will tell if I keep them both. 5/5.
With one red knob installed, Shades is a three-channel atteunverter and mixer. Any channel without an input patched generates an offset, and without an output patched, mixes with the next channel. It’s nicely controllable. 4.5/5.
The penultimate module is Tides, 2018 revision. It’s a function generator, but one where you can set the cycle time and shape separately. This is perfect for VCO and LFO use, and unusual but functional for envelopes. Tides also has its own filter/wavefolder combination, and a PLL mode that follows an incoming clock or VCO. The 2018 version, aside from being more accurate and having attenuverters for every input, has four modes that can shift the level, phase or frequency (with harmonic relationships) across its four outputs, which extends it capabilities nicely. 5/5.
On the corner is Livestock Electronics Maze. It’s sort of a combination matrix mixer (like the A-138m) and sequential switch — there are 16 “pages” of matrix settings which can be selected by buttons, stepped through with a trigger or selected by CV, and it can jump or fade smoothly between them. It’s a thrilling concept, but in practice I don’t find myself using it as much as I thought. I often consider whether I should replace it, but then I come up with scenarios where it was exactly what I needed, and nothing else its size would do. 4/5.
This is the current state of my pedalboard, or really, a shelf hanging over the Mantis at a 45 degree angle. Guitar FX pedals offer some neat alternatives to software-based effects or Eurorack effects, but they run at lower voltage levels generally, thus the need for attenuation before and boosting after to work with them.
In the upper left are a pair of spring reverb tanks. I use those with the Koma Elektronik Field Kit FX, in the lower right. The FKFX has a Eurorack panel option, but it’s pretty wide. Aside from the spring reverb driver, it has a PT2399-based digital delay (cheap and gets weird and crunchy at long delay times, which can be great), a frequency shifter, a 4-input VCA mixer with tone control on 3 channel and overdrive on the fourth, a little modulation source that can be a 4-step sequencer or an ADSR envelope, and four assignable CVs. I’ve used it particularly with the Dynamo for setting up feedback loops with the reverb and frequency shifter. Spring reverb is kind of fun to mess with, but also very touchy to work with. 3.5/5.
In the upper right is a Rochambeau Musical Apparatus Monobius, custom 6-knob variant. This is a combination ring modulator and fuzz with bandpass filter. It’s very noisy and odd, and was one of those trades I did on a whim rather than a plan, but it adds some neat flavor. 3.5/5.
Left of the FKFX (because on pedals, the signal flow usually goes right-to-left) is WMD Geiger Counter, in a rare distressed black colorway (they are normally screaming yellow). It’s an 8-bit waveshaper, distortion and sample/bit reduction device. With precise enough control over input levels, it is a nice alternative to more traditional wavefolders. 3.5/5.
Next is Red Panda Tensor, a sort of quasi-tape looper, pitch shifter, time shifter, reverser, randomizer thing. It is cleverly set up; it listens even when “off” so you can instantly get a reversed repeat of what you were playing; it can judge when to (smoothly) reset its buffer to prevent overflows when you’re playing back repeats more slowly than they came in, and so on. Sometimes it feels like a human playing counterpoint to me; other times it just makes a neat background wash of stuff, or a sweet chorused sound. Its stomp switches are the non-clicky type (and can be switched between momentary and latching), which I prefer since I don’t use it with my feet. 5/5.
Then there’s EarthQuaker Devices Afterneath. It lives somewhere between delay and reverb. with a little chain of echoes that can be diffused. It’s very easy to get infinite feedback going with it and keep it under control. The minimum pre-delay time is longer than I would prefer, which limits the flexibility a little. A nice non-clicky stomp switch on it. 4/5.
And finally there’s the Zoom MultiStomp MS-70CDR. It’s a multi-effect with several classic and modern versions of chorus, delay, reverb, and other modulation and a few utilities like EQ, noise gate and compressor. With three push encoders and an LCD screen it’s a little friendlier for my purposes than some multi-effect pedals, but it’s still occasionally just a little bit tedious to set up. The processing is pretty great, the sound quality can be a bit noisy at times (as guitar pedals sometimes are) but not terribly so, and the price was fantastic for something this flexible. 4/5.
I have a Chase Bliss Dark World on preorder, which should ship around the start of the year. It provides three reverb algorithms on the “world” side, and three effects on the “dark” side to add gloom or shimmer or the infinite void. You can run the two sides in either order or in parallel and there’s a master tone control to darken it more if need be. The demos have been impressive, and I may see this replacing the FKFX (thanks to a spring reverb model) and maybe the Afterneath too. Hopefully a 5/5…?
Since writing that last post, I had a bee in my bonnet about replacing the Tyme Sefari + A Sound Of Thunder, and spent the weekend researching the alternatives.
Long story short: I have decided to keep the Tyme Sefari. Sure it’s lo-fi, but I had occasionally thought about getting a Doepfer BBD (analog bucket-brigade delay) precisely because it is lo-fi, in kind of a similar-ish way. Durrr… anyway, running Tyme Sefari’s main output through a filter and turning it down a bit dark actually sounds really lovely.
And I like the logic of how Tyme Sefari works, which is unique. Only one other module I’ve found can do everything it can — SDS Reflex Liveloop — but that one works through a bunch of different modes instead of a unified, fluid design.
I’ll probably ditch the A Sound Of Thunder expander though. The only feature on it I really like is the slightly awkward extra channel for stereo, and I can let go of that. Most of my hardware stuff runs in mono channels anyway, or else mid/side encoding.
Mono (monoaural) vs. stereo is probably familiar — either one audio channel, or two related channels. Stereo typically uses “LR” encoding — left and right, predictably enough. “MS” or mid/side stereo, instead treats one channel as the middle of the sound field, and the other as the differences that happen on the edges.
Mid/side is not very intuitive to think about, but it’s pretty simple to convert from LR. Just add the two channels together to get the mid signal, and subtract them from each other to get the side signal. Some attenuation might be needed to avoid distortion if the inputs are already loud.
With mid/side encoding, you can’t make distinct left/right movements — But it’s very handy for getting a balanced yet wide and dramatic sound by processing the mid and side differently or even using different sources, and I like working with it especially in modular.
There are a couple of mid/side encoder modules available, but I usually do the conversion with a plugin.
With that wrapped up, it’s time to look at the top row of the Mantis!
At left is the second half of the Doepfer A-180-9 Multicore mentioned previously. Plugged into the top are the stereo outputs from one of my FX pedals, sent onward to my audio interface without really being processed by any of my modular gear.
The next module, with the amber-on-black screen, is Monome Teletype. This module lets you write short snippets of code with a computer keyboard (wireless in my case) which run whenever triggers are received or its internal metronome ticks. It has a single CV input, and can output gates/triggers and control voltages — so you can use it to manipulate gates, play stored sequences or generate them algorithmically, record CVs as a new sequence, generate envelopes or LFOs, and so on. It’s extremely flexible, it just requires thinking a bit like a programmer — which I am. Mine’s mounted upside-down with a firmware-flipped display, to keep the jacks out of the way of the screen. 5/5.
Algorithmic sequencing or algorithmic composition is the practice of using some relatively simple math and logic to determine rhythms and/or pitches, rather than explicitly composing them. Algorithms may include some stored patterns, but these are the basis of further logical or mathematical operations.
Generative patches are those where the interaction of electronic circuits drives the music. The Krell patch is an iconic example.
Sometimes the lines between the two are blurred or erased, with digital algorithms feeding analog processors and vice versa.
Circuit Abbey G8 is on its right. It’s a clock divider or clock distributor, which is something Teletype can do very easily. But unlike the Teletype, it runs fast enough to send audio through it and get back squarewave audio in lower octaves. Sometimes I keep the Teletype busy with other things anyway. It’s useful enough to justify keeping it around. 4/5.
And then there’s Noise Engineering Mimetic Digitalis. It’s a 4-channel sequencer laid out on a 4×4 grid, which can be navigated via triggers or CV in two dimensions or linearly or randomly, all at the same time. I really like this one in theory; the problem is I don’t often find myself using it that much in practice. I can imitate most of its behavior in Teletype and in another module later in this row. I still feel like I should give it more of a chance, though. 3/5.
Next, with some cables plugged in and sent overhead, is ALM Busy Circuits S.B.G It attenuates an audio signal from Eurorack levels down to the levels expected by guitar FX pedals, and raises a pedal-level signal back to Eurorack levels. It offers a dry/wet knob, and a further converter from Eurorack CV to 3V or 5V “expression pedal” inputs for FX pedals. It does the job, though I feel like the knob response/ranges are a little odd and the layout could have been friendlier (I should at least turn it upside-down). 4/5.
Dry/wet or simply “mix” is a common control on FX. The “dry” signal is the input of any FX unit or chain, while its output is “wet.” Blending them is a nice way to avoid overwhelming your audio with too much of a good thing, e.g. reverb.
So slim you might miss it, 2hp Trim is next. It’s a dual passive attenuator. Along with the S.B.G I’m using it to lower Eurorack signals to guitar FX levels, since I’ve got a few FX I want to use separately. It’s actually not calibrated for this purpose, with the right range being somewhere in the lowest 1/10 or so of the knob. 2/5.
To its right is Circuit Abbey Gozinta. It’s an amplifier to give a clean boost to, you guessed it, signals coming in from FX pedals. Or any other source where the voltage might be lower than you want, or where you want to overdrive a signal to distort it. It does its job admirably. 4.5/5.
The wide module after that is Pittsburgh Lifeforms Double Helix. It’s got a West Coast vibe overall, with a pair of oscillators, an LFO, a wavefolder, an LPG, and a dual modulation bus that makes it easy to assign things to modulate other things. It’s also haunted by crosstalk and weird interactions within the module, which can annoy or please in equal measure. The oscillators are kind of “chewy” and well suited to the character of the folder and LPG (which is a simplified Dynamic Impulse Filter). It sounds great, and whenever I consider dropping it, its unique character grabs me and changes my mind. 4/5.
Next, with the big knob backlit in blue-green, is a DIY build (by someone else) of Mutable Instruments Warps. The general idea behind this one is it combines and mangles two audio signals in various ways — though it also has its own VCO which does rather nice phase modulation. I sold my first one, and months later, traded for this because I missed it. I don’t actually use it a lot, but it’s occasionally welcome. 3.5/5.
Mutable Instruments is one of the popular Eurorack module brands. Its founder started with open-source, DIY desktop synths and chose to continue using that model upon entering the Eurorack market. This has its pros and cons; parts of Mutable Instruments code live on in many other open-source projects and people have made alternate firmware to add new functions to the modules. People have also independently done DIY builds of the modules, or smaller redesigned versions, for themselves or others.
The black module with the blue display is uO_C, or micro Ornament & Crime. This is part of an open-source hardware and software project, which is itself partly based on Mutable Instruments code. I have Hemisphere Suite installed on mine, which is a set of dozens of utility apps, ranging from envelope generators to Euclidean generators to quantizers to quixotic sequencers. There’s almost always something useful for it to do. 4.5/5.
The narrower black module to its right is Erica Synths Pico A Logic. Given two inputs, it returns the sum, difference, maximum and minimum voltages. Hemisphere Suite can do that too, but this is better suited to audio signals since it’s not limited by rate. I don’t use it a lot, but it fills an awkwardly small space for which I don’t have a better use at the moment. 3.5/5.
The neighbor with the red buttons is Ladik P-075 Dual Switch. It’s a simple, passive module where you can connect or disconnect any signal via a manual toggle switch and button combination. The button inverts the state of the switch, which is clever. It’s handy to run constant voltages through just to use as a gate source, or to mute parts of a patch. 4/5.
The wood panel on the right side end is a Bastl Dynamo. It’s a busy module, where the top part combines an envelope follower, comparator and some inverters and rectifiers to create a control source for a VCA for compression — lowering the level when it gets too high. Of course it can be patched in other ways as well. The bottom section is an inverter and a very fast A/B switch, and I don’t fully understand the intention of including it rather than a VCA, but it can do some cool stuff once in a while. Again, I don’t like the wood panel and was supposed to have received an aluminum one. 3.5/5.
One more row to cover, plus some FX and mentions of favorite software and maybe a couple other things!
While writing this post, I started patching up a generative piece using two LFOs ANDed and XORed together, inverted, fed to the G8… all those gates clocked Mimetic Digitalis, some envelope generation, delay syncing and so on. By the time I finished it was quite a busy patch, the kind that uses up most of my cables. If I still think well enough of it tomorrow, I’ll be submitting it to the Ambient Online Fire compilation.
Onward to the second and third rows, where we’ll actually get to stuff that directly makes noises! And has orange knobs!
But first: on the left we see a Pittsburgh Modular Lifeforms Dynamic Impulse Filter. This is a modern non-vactrol LPG with a classic design, with a mode switch that lets you select whether it’s going to act as a VCA, an LPG, or a lowpass filter. Its response to triggers is not to my liking, but it works well controlled manually or via CV. It has a uniquely woody tone, and controls that make it handy in any mode. 4/5.
Filters, particularly VCFs (voltage-controlled filters) are important building blocks in synthesizers. (Subtractive synthesis, the most common technique, is performed by pairing a harmonically rich simple oscillator with a filter.)
Filters reduce frequencies in part of the spectrum, while optionally emphasizing it in others (called resonance).
A lowpass filter allows lower frequencies to pass through while quieting higher ones; a highpass filter does the opposite. A bandpass filter allows only a narrow band of frequencies to pass through freely, while a notch filter reduces only a narrow band of frequencies. A multimode filter typically will either switch between these or provide all of them simultaneously.
Different circuit designs have widely different characteristics in terms of slope, resonance, stability, phase characteristics and so on. Many designs will self-oscillate when the resonance is high enough, producing very pure tones even without input.
Equalizers (EQ) and tone controls are an especially gentle type of gentle filter.
Next is the Doepfer A-196 PLL (phase-locked loop). This is a strange beast! Given incoming signal (audio rate or slower), it attempts to match the frequency and create its own set of pulses. It does this by constantly measuring whether it’s ahead or behind, and feeding back a signal to itself to make adjustments… but depending on settings, the material it’s tracking, and how much you intentionally confuse it, it can be joyfully terrible at its job. The open nature of this module lets you do all kinds of bizarre tricks with it. One of my favorites is patching in a different oscillator instead of its own — perhaps one that’s trying to modulate itself and therefore can’t properly track pitch without extra help. 4/5 (it’s pretty situational, and there’s sometimes a razor-thin margin of error between good tracking and garbage).
The leftmost orange-knobbed module is the Hertz Donut mk2 by The Harvestman (now called Industrial Music Electronics). It’s a proudly digital take on the Buchla 259-style complex oscillator. It’s not a very clean sound to start with, and includes phase feedback as an option on its modulation bus, a harsh “discontinuity” control rather than a typical waveshaper, as well as an XOR output which is a rough sort of ring modulation. But FM with it is super-easy and the thing really can be beautiful, especially through an LPG. 4.9/5! (If it remembered its state when powered down that’d be perfect.)
An oscillator generates a signal that repeatedly fluctuates between positive and negative voltages. This is the most important building block in any synthesizer. (Cue people arguing that the filter is more important, but I say, not without something to filter!)
A VCO is a voltage-controlled oscillator, which is the kind most relevant to us in modular synths. VCOs use the 1V/octave standard for frequency control and typically work at audio rates (20 to 20,000 cycles per second, or Hertz (Hz)) just like sound waves.
If it’s slower than that, it’s an LFO — low frequency oscillator — and rather than making sound, it will control or modulate something else. Some LFOs can synchronize to a clock signal to stay on rhythm; many VCOs also can synchronize, which can have interesting effects on the sound. (It’s rare for VCOs to be much faster than audio rate, but there are specialized uses for it.)
The shape (or waveform) produced by the oscillator determines the sound (or timbre, which annoyingly is pronounced “tam-ber”). There are a few common ones in synthesis because they’re easy for analog circuits or very simple code to produce — triangle, sawtooth, pulse/square, and sine — but some oscillators can create extremely complex shapes.
Complex oscillator is a term for a specific configuration as used in the Buchla 259 and many imitators. It consists of a pair of oscillators that can track together and modulate each other through FM and in other ways, frequently using a “mod bus” to assign and control the amounts. The “primary” oscillator also includes a waveshaper or wavefolder which further increases the harmonic complexity. This is the quintessential West Coast design, rarely seen outside of modular synthesizers. (Some people use “complex oscillator” in a more generic adjectival sense, but this is discouraged.)
Generally, analog oscillators are prized for a sort of mythic “warmth” and other subtle qualities, while digital oscillators are much more diverse in terms of shapes they can generate as well as potentially more precise (or imprecise in specific ways). A common type of digital oscillator is the wavetable, which keeps shapes in memory and can smoothly change between them. Other digital oscillators are based on mathematical models and algorithms computed in real time.
FM, frequency modulation, is a synthesis method where the output of one oscillator modifies the frequency of another oscillator. Slow FM is just vibrato, but at audio rates the result is instead a change in timbre, which can be brassy or bell-like or incredibly noisy depending on the frequency ratio between the two oscillators. Many non-modular synths like the famous Yamaha DX7 actually use phase modulation (PM) which is similar but simpler in some ways. In the modular world, one deals with exponential vs. linear FM, thru-zero and other complexities having mostly to do with keeping things in tune while being able to shift the timbre at will.
There’s also AM or amplitude modulation, which is simply controlling a VCA with a second oscillator harmonically related to the first but has somewhat similar timbral effects; RM or ring modulation, and filter FM which modulates the cutoff frequency of a filter at audio rates, producing a combination of amplitude and phase modulation.
(An aside about my aside: some people love to “debate” analog vs. digital audio. This is less common among modular synth types, probably because we know it’s like arguing whether a wrench is better than a screwdriver. If you use the most practical and appropriate technology for the function at hand, you’re probably going to have both.)
The next orange knobbo is The Harvestman Kermit, after Kermit Washington, the One Punch Man of basketball. It was designed as a dual LFO based on a small wavetable, but it will also run at VCO rates, where it’s very noisy and fizzy and unstable in a beautifully disastrous way. Something about it makes it especially well suited to amplitude modulation. It would benefit from fine-tuning knobs, normal-sized knobs for the modulation inputs, and remembering button states on shutdown. The clock/pitch following mode is a sloppy mess. But the sound is unbeatable, and it works very nicely for LFO purposes. 4/5.
Then we have The Harvestman Tyme Sefari, named for a time travel company in a Ray Bradbury story. It’s a lo-fi digital audio buffer that can capture and play back audio — a very flexible sampler or a delay/echo depending on how you configure it. Like Chronoblob, it has an open feedback loop that adds flexibility. But unlike Chronoblob, the sound just can’t avoid being grungy and that doesn’t always suit. It’s also pretty large, especially with its expander module. There’s a firmware update supposed to be coming soon which should make it less glitchy, and I’m pretty much waiting to see what I think of that before letting it go for something else. 3.5/5.
To its right is its expansion module, A Sound Of Thunder (which is the title of that Bradbury tale). This adds a second channel to the TS which can be used for stereo or other nefarious purposes. There’s no mix control though. It also adds some marginally useful options to make the sound even worse or to shift the pitch. It’s overly large for what it does. 2.5/5.
Getting back to white knobs again, we have the Antimatter Audio Crossfold. This is a nice wavefolder with two inputs, which lets you crossfade between the two, control the folding amount two different ways, and fade between the original (or mixed) and folded signals. This combination of features gives it some great sonic versatility. 5/5.
Here comes Bastl Cinnamon, a multimode filter. It’s pretty straightforward and sounds good, with some switches to make it rougher. I won it in a charity auction and it was supposed to have been the aluminum panel version, not wood — which is weirdly thick and doesn’t fit well, and makes the text labels hard to read. Still 4/5.
Last are a couple of passive infrastucture bits. The Doepfer A-180-9 Multicore lets me make 14 connections between this rack and the case on the other side of my desk through a pair of Cat-6 network cables and very compact panels. (I am in the habit of keeping two lines to my audio interface, which on the other side are stereo outputs of one of my FX pedals.) Unfortunately I still had to run a separate ground wire between the cases to eliminate hum. If the two network jacks were both at either the top or bottom — and better yet, if they had a rear option — it’d be a little less unwieldy. 3.5/5.
The last tiny module with a row of nothing but jacks is a Bastl Multiple. It passively connects all 9 jacks together, and you can configure it by snipping wires on the back. I divided this one into (2, 2, 2, 3) just to have a handy way to patch into 3 inputs and an output on the back of my audio interface. Cheap and useful, 5/5.
The bottom row of the rack has just 5 modules, but the first one’s a doozy. It’s a beta test version of the Synthesis Technology E370 Quad Morphing VCO. It is the absolute monarch of wavetables, with four independent or coordinated VCOs. Borderline overkill.
My main gripe about it is that dynamic linear FM is quite tricky to set up, compared to the ease of the Hertz Donut. I believe it could be solved with a software tweak (a menu option for a DC blocking filter on the FM input while in linear modes) but can’t convince the DSP guy to try it, so it is what it is. However, all the other audio-rate modulation you can do to the thing is fantastic — the internal 2-Op FM model, exponential FM, PM, AM, wavetable morphing, wavefolding — and you can create FM wavetables in the first place. And the cloud mode is stupefyingly good. So it’s a 5/5 for sure.
The next module with a screen is Dave Jones O’Tool+. It’s a dual-trace oscilloscope, voltmeter, frequency meter, tuner, spectrum analyzer, and metronome. It’s invaluable for learning new modules, setting up tricky patches, solving problems and giving you answers that your ears don’t. 5/5.
A patch is a “program” of modules and connections. It can refer to everything you’ve got set up at any given time or a functional subset; it can be a general term (a bass patch, a Karplus-Strong patch, a feedback patch) or refer to exact settings. The term is sometimes used outside the modular realm it’s also used to refer to a synth program or a configuration of gear. Some people diagram them or photograph them; I sometimes take general notes.
A patch programmable module is one that takes on different behaviors depending on what’s patched into it — like the function generators I mentioned in the last post which can generate envelopes or LFOs, delay a gate signal, filter audio, smooth a stepped signal, and so on. Multifunction modules, on the other hand, require changing between different mode to determine the behavior. Of course, there are modules blur that line.
The plain-looking module with the eight knobs is Epoch Modular’s Twinpeak filter. This has two inputs that can be mixed, and a knob to fade between a single lowpass filter mode and a dual bandpass. At the highest resonance settings it doesn’t continuously self-oscillate, but a trigger into the input will make it ring in a lovely percussive way for a second or two. I am not a big filter connoisseur since I lean more toward non-subtractive synthesis, but this one has proven itself to be pretty cool. 4/5 (a mix CV input would have been nice).
Frap Tools 321 is a three-channel offset and attenuverter, with a pair of mix outputs (one mixes everything, the other mixes only the channels where you haven’t patched the outputs). Very useful for getting signals to the appropriate levels. I’m not in love with the tiny black switches on the black panel with tinier labels, but it does the job. 3.5/5.
Finally there’s the S********k S***r P***r G***n. It’s a power supply module. Some time after I bought it, the owner of the company revealed himself as a misogynist alt-right toolbag and proud trust fund baby who, he says, doesn’t need any of us. So I covered his logo with sparkly star stickers and never did business with him again. The power supply works okay.
More happily, behind the panels I’ve got a Genus Modu Low Impedance Bus Board (LIBB) helping to distribute that power and keep it clean.
And that’s my rack! Part 3 will cover the top row of my TipTop Mantis case, which is full of fun things that need explanation. 😉
And here’s the bit where I get into what’s in my modular synth cases, and what I like and don’t about each bit. I’m going to take it slow, and try to keep things as interesting as I can for people who are new to synths and for more experienced types curious about what I’m using and what I think of it. As a result, this is going to take a few posts.
Let’s start with the rack on the left side of my desk. It’s a desktop 19″ rack stand, of the kind you could fill with servers and backup drives, or with mixers and effects. At the bottom I’ve left one space empty for cable clearance, and then a 1U power switch bank — handy for devices without their own switches, and also for rebooting the router and cable modem without crawling under the desk.
At the top, there’s this:
Left: A “monitor lizard” who lost his natural habitat back when I went from a CRT to LCD.
Middle: CV.OCD by Sixty Four Pixels — a very customizable MIDI converter with 4 CV and 12 gate outputs. With this I can sequence notes, other gates, and synchronize from the computer to modular gear. I have mine set up so MIDI channel 1 outputs pitch, velocity, gate and an extra trigger; channel 2 and 3 each have pitch and gate; channel 10 assigns five specific notes to gate outputs, and there are three clock sync signals at different divisions.
A quick bit about how Eurorack modular synths communicate: they don’t use MIDI, they use analog control voltages (CV). Each patch cable carries one signal. Signals have no intrinsic meaning and are left to modules to interpret, but there are some typical signal types.
Frequency (pitch) has a standard of one volt per octave (“V/OCT“).
Gates, which determine whether a note is held or some other temporary event is active, are either “low” (0V) or “high” (some arbitrary positive voltage).
Triggers are simply short gates and the important part is the rising edge where they jump quickly from low to high.
Clocks are triggers that repeat to define timing.
Audio is a voltage that fluctuates quickly enough to be audible if connected to a speaker.
Everything else is probably modulation of some sort, though a fixed steady voltage is often called a reference or offset.
These signals are mostly interchangeable, with varying standards and consequences — discovering these is part of the fun of modular.
Right: The thing with the eye is a Bleep Labs Thingamagoop 3000. It’s a weird sound generating toy with a “LEDacle” that can blink in different patterns and speeds, and a light-sensing eye that responds to it (or other changes in light and shadow). Good for drones and weird warbly noises, I don’t use it often but it’s fun to play with. It can generate and respond to CVs too, but it’s much easier to just use it standalone and only tap its audio output.
Bottom: Focusrite Saffire Pro 40. This is how I get audio in and out of my computer. 8 analog input channels, some output channels, plus digital I/O that I’m not using, with easily accessible level and monitoring controls on the front. Perfect for my needs, except it’s Firewire rather than USB and initial setup was kind of a fight.
My top row of modules starts with Manhattan Analog DTA on left. This one was a prototype unit I bought directly from the maker. It’s two single channel VCAs in series, plus an extra drive stage to add some saturation and weight. Not the controls I expected (which would have been a manual bias as well as CV level) but still useful. Shame it’s AC-coupled. I’d rate it a 3/5 with no rancor; it’s not exciting but it does its job pretty well.
A VCA, voltage controlled amplifier (or voltage controlled attenuator) controls the level of one signal using another signal — an automated volume control. Typically you use this to shape the dynamics of a note using an envelope (a signal that responds to a gate with a voltage that rises, sustains and falls according to timing you set). But there are dozens of other uses, and a cliche phrase “you can never have too many VCAs.”
An attenuator is just something that brings the voltage closer to zero, but this is vitally important in modular. An attenuverter can also flip it between positive and negative. An amplifier pushes non-zero voltages farther from zero. All of these are multipliers — attenuation is multiplying by a value between 0 and 1, attenuversion multiplies by a value between -1 and 1, and amplification multiplies by a value higher than 1.
This brings up unipolar vs bipolar signals — which is simply whether negative voltages are expected or ignored. Audio is bipolar, but the control input on a VCA is typically unipolar — send it a negative voltage and it remains fully silent.
But there arebipolar VCAs which invert the incoming voltage if the control input is negative. These may be called “four-quadrant multipliers” or “ring modulators” or “balanced modulators” because apparently one term for the same thing wasn’t enough.
AC-coupled means it responds fully only to audio signals; any signal that changes too slowly will fall to zero. There are some technical reasons why one might want this, but usually DC-coupled is preferable since you can manipulate non-audio signals with it.
The second module is Instruo tanh, which has three channels of smooth saturation/limiting of audio or control signals. This can give audio a bit more of a push or “warmth” depending on levels and usage, can be used to keep feedback somewhat under control, or can be used to give extra curvature and interest to control signals. I like running envelopes through it to give them a sort of “sticky” feel (words fail a little). 4/5; those mini knobs are tolerable but not lovable and to really help with feedback control it needs an attenuator afterwards, and I feel like two channels with attenuation would have been nicer than three without.
Next in the line is Rabid Elephant Natural Gate, the holy grail of LPGs. The sound is gorgeous, the panel finish and art are gorgeous, even the box it came in is gorgeous. It’s extremely controllable and playable, with easily tweaked manual “open” sliders as well as decay times and a 3-way response curve switch. It also has a tendency to get louder if you trigger it rapidly, which can intensify the drama. I had the chance to try it at KnobCon Six, and bought one within minutes of the release announcement. Since then the demand has skyrocketed and the price has increased significantly due to the need to switch to a different contract manufacturer. But it’s soooo good. 5/5!
An LPG or lowpass gate is related to VCAs, but as the control signal decreases, the higher frequencies (treble) go quieter more rapidly. This can give more natural dynamics and is favored in West Coast synthesis.
VCAs respond immediately to changes in control voltage, but LPGs close (go quiet) more gradually than they open. A quick trigger to a VCA would let through a “beep”, but the same trigger to an LPG would give a natural, percussive “ping” or “strike.” This is such a common usage that some people forget you can use an envelope with an LPG just as with a VCA.
Classic LPGs use an electronic part called a vactrol, which were originally made using toxic materials. The EU has imposed strict limits on them in recent years. They also vary a lot in behavior and have to be hand-picked for desirable response. It’s possible to make vactrols using greener materials, but overall they seem to be falling out of favor, replaced by other circuits or digital emulation.
Soundmachines LS-1 Lightstrip: a handy touch controller that my spouse bought me last year. It will generate both a gate and a voltage according to where you tap or slide your finger, can hold the last touched level, and it can record and loop a few seconds of movements. Run it through a scale quantizer and you can use it like a micro keyboard! 4/5; sometimes the hold or record modes activate when you don’t expect, or don’t when you do expect.
A quantizer takes a signal and “rounds” it to match the nearest note value. Some round to the nearest 1/12 of a volt, while others let you choose between different scales or define your own. It’s like imposing frets onto a fretless instrument. This is great for random signals or imprecise controllers, and can be fun when sliding between notes.
Make Noise Contour is the aluminum one with the vertical row of four white-capped knobs. It’s a basic ARSR envelope generator similar to Moog designs, with adjustable response curve. The first envelope I tend to reach for when patching the left rack, it’s just so friendly and easy to dial in. 4.5/5.
Envelopes are often described by various types of stages: A for “attack” (onsite or rise), “D” for decay (fall), “S” for sustain (hold steady while the gate is high), “R” for release (a separate final decay after a sustain), and “H” for hold (hold steady for a fixed time).
The most common envelope types are:
D: immediate rapid onset followed by an immediate decay, like a percussion instrument or plucked string, or a vactrol strike. Gate time is irrelevant, this envelope treats its input as a trigger. If another trigger is received while still decaying, behavior can vary from module to module.
AD: rise to maximum, followed immediately by decay to silence. Again, gate time is irrelevant, and the behavior when receiving a second trigger can vary from module to module.
AR (or ASR): rise to maximum, sustain while the gate is high, and release to silence. If the gate is released while still in the attack stage, immediately release.
Many AR envelopes are implemented by slewing an input signal. That is, if the input voltage rises, the output level rises more slowly. If it falls, the output level falls more slowly. If the input happens to be a gate, the result in an AR envelope. If it’s stepped voltage levels, the result is smooth voltage levels. If it’s audio, the result is filtered audio (or silence if your attack and release are too slow). Modules that do this (and also AD, using a separate trigger input) are often referred to as function generators, and typically have EOR (end of rise) and EOC (end of cycle) trigger outputs in order to chain and synchronize other events.
ADSR: rise to maximum, decay to a specified sustain level, hold while the gate is high, and release to silence. With a zero sustain level it can imitate an AD envelope, and with a maximum sustain level it imitates an AR envelope.
ARSR or ADSD: a simplified form of ADSR where the decay and release rates are the same.
Multistage envelopes can have multiple stages of various types, as can vector envelopes which define level over time with a series of X and Y coordinates. Even rare are “spring” and “bouncing ball” envelopes based on physics models.
ADSR is the classic East Coast envelope. AD and AR envelopes, and function generators, are generally associated with “West Coast” synthesis. AD envelopes typically have a loop switch which allows them to rise and fall continuously, taking the place of the LFO (low-frequency oscillator) which is more typically East Coast.
All this East/West stuff is largely an arbitrary historical distinction between how Robert Moog and Don Buchla designed their synths. While there’s some truth that certain design philosophies work together well, the “coast thing” should not be taken too seriously!
To its right is Make Noise Function, a function generator that also has a “hold” function that freezes its current level. Much like Contour, the knob ranges and response just make it super easy and “musical” feeling. 4.5/5.
And then WMD/SSF Mini Slew, another function generator. This one has more CV inputs and a built-in VCA — it’s cleverly arranged as an attenuverter for one of the module’s outputs, with a bipolar CV in jack. You can run audio through that CV in jack, and the module’s envelope will control its volume. But the response of the controls is much more fiddly than Function, the EOR/EOC outputs are a bit flaky, and unless the trigger is above about 6.5 volts, it won’t retrigger during the falling stage — which can be advantageous but it’s an awkward default. And for some reason it requires twice as much current from the power supply as Function. 3.5/5.
Make Noise ModDemix is next. This is a pair of bipolar VCAs with kind of a nonlinear twist right around zero that gives it a little extra spice. There’s also some normalling which straddles the line between confusing and useful. I should use this more because I often forget it’s there — maybe if I stuck it over on the left side between DTA and Natural Gate it wouldn’t sneak away from me. 4/5
Normalled connections are default connections that are interrupted when you plug something into an input. For instance, on the DTA, if you don’t plug anything into a CV input, a constant positive voltage is normalled there instead so you can use the knob as a manual volume control. The same is true of the audio inputs on Natural Gate, so you can use it as a D envelope generator when it’s not processing audio.
The module with the cute pink blob is Alright Devices Chronoblob, a digital delay effect. I have a lot of delays in software, but it’s good to have at least one in the modular system as well. What makes this module special is (A) a sync input which lets you lock delays to the beat, (B) the choice of whether to change delay time with a smooth tape-like pitch shift or a clean crossfade that lets you bounce delays all over the place to create interesting patterns, and (C) send and return jacks for the feedback loop which let you affect the echoes or do some other clever hackery. 4.5/5 (a level input knob would have been swell and a clean feedback limiter would have been ultra).
Ending the first row is a Doepfer A-138m matrix mixer. This has four inputs on rows, four outputs on columns, 16 knobs to assign them, a switch to choose bipolar or unipolar control of each column, and the top row is normalled to a constant voltage. It can act as a manually controlled voltage source, independent attenuverters or mixers, a handy way to mix sources in stereo, control feedback, rearrange the signal flow of other modules patched into it, and even can be used for sequencing by feeding it combinations of gates. 4.5/5 (higher frequencies can leak through just a little even when the knobs are turned down fully).
A sequencer stores sets of voltages and/or gates as discrete steps, and can play them back synchronized to an internal or external clock. Sequencer designs vary widely to cover many different needs and styles of composition; some have multiple channels, are rhythm-oriented, can rearrange the step order, and so on. Since musical pitch is a primary use for sequencers, many have built-in quantizers.
So that’s what’s in my first row. Stayed tuned for the other four and the pedalboard & miscellaneous stuff… and then some words about the logic of module arrangement, why I chose one module over another, (unsponsored) recommendations, various sequencing methods I use, and a lot of other blathering. 🙂