There might be a way to add new waveforms to the Zadar, but the amount of "primitives" you have to work with (260) plus all of the complex transformation capabilities over those means that you probably won't ever have to do that. As for the ES-8 conflict issue, it shouldn't be a problem given that most DAWs these days allow you to specify which channels are being fed by/feeding which signals. You can probably set up a template that gives the Fireface and ES-8 clear directions as to who does what, and then use that as your main work template preference. I'd suggest going to Expert Sleepers' website (https://www.expert-sleepers.co.uk/) and looking over the Silent Way documentation as it relates to the ES-8 and using that with another USB audio interface. The other convenient thing, though, is that if you want to play live with only the modular, all you'd need is the DAW with Silent Way to handle your basic CV/gate functions with no need to pull the ADAT Lightpipes. And also, notice that the ES-8 has Lightpipe connections, so if there was a conflict, you could still fall back on those. But even moreso, the ES-8 has four inputs which the ES-3 doesn't have, allowing your modular to send control signals (such as clocking, triggers, etc) back into the DAW along with the capability of recording the modular's audio direct-to-digital.

No, the Erica cab doesn't have a fan. But what it does have is that amazing 140mm case depth plus excellent power specs, and the ability (if eventually needed) to add Erica's tube heater P/S addon for later tube modules, if you find some of those interesting. The Erica case is also vented, which helps to keep heat buildup issues (such as tuning drift) to a minimum.

Yep, modular is expensive. Nothing much you can do about that if certain functionalities that require specific modules are required. But while the build I put up looks complicated, ultimately, it's not. Everything's very straightforward as far as panel markings and controls, plus you have the ability to automate (via the ES-8) many of the Zadar functions, Akemie's algorithm selection, your mix levels, etc with the DAW/Silent Way able to track this via musical cues from the modular itself...yet there's still lots to tweak manually as well.

If the idea of cramming functionality into a given space is key here, I'd suggest looking into some manufacturers who specialize in the 3-5 hp range. Ronin makes a very good point that if you go too small (ie: basing everything on 2hp's modules) you'll wind up with a patch panel that's a total nightmare to navigate. Companies such as Ladik and Bastl, however, also do small well...but there's more "real estate" to work with with those, and you'll also wind up with a few more functions than the basic 2hp ones provide.

There is, however, another less-obvious issue here. Let's do some math...

Ladik does a number of different ADSR envelope generators, all with additional functions that allow them to specialize in different functions that go a bit beyond normal ADSR EGs. Each of these are 4 hp (except their fully-CVable ADSR, which is 8 hp) and each costs around $75 (the 8 hp one is $80). 2hp, OTOH, has only one basic ADSR, which is 2 hp (of course) at $89. On the surface, it doesn't seem like there's much difference, money-wise. BUT...if you start adding up how much it costs each module to occupy its space in terms of $ per hp, it becomes obvious that the smaller modules are actually more costly in terms of space they occupy. If a Ladik C-212 (ADSR with extra inverse output) fits in 4 hp and costs $75, this comes to $18.75 per hp filled. But then, doing the same test with a 2hp ADSR, you get $44.50 per hp.

That cost per hp figure is actually a valid measure of how economical your build eventually comes out to be. Let's say you have a single row at 84 hp to fill. With a $/hp of $1575, a row of Ladik ADSRs (if you were to do such a thing) is ultimately much more cost-efficient than that same row in 2hp ADSRs ($3738). But then...let's say you did a whole row of Ladik's CVable ADSRs. At $80 per module, each hp then costs $10, giving you a cost per that theoretical row of $840.

And this issue is in addition to the point that your build above is 3 x 84 hp. So, going tiny might be great if you're trying to do a very portable modular...but if you're going to build something bigger, not only would you have a difficult patchpanel to use overall, the result would be far more costly and inefficient, cost-wise. Not to ding on 2hp, mind you, but while their stuff is awesome for the small portable crowd, in a (relatively) large build 2hp's modules are better suited to fill gaps with functionality instead of throwing that away by putting in a blank panel. So, basically, if you have the space, use the space.

With the DLD, just leave it as-is with 4ms's stock panel. It's not difficult to read, so I don't think the expense/hassle of repaneling it makes sense. It's also a pretty control-dense module, so doing the work to get all of the controls, jacks, lights, etc lined up and resecured just seems like a major hassle.

The uClouds is simply a tiny version of the same thing as the Supercell. The Supercell adds more function controls, but at the core both are still the same open-source firmware. Same thing goes for the other smaller MI clones.

As for the sequencer...the ER-102 Sequencer Controller is definitely a must with the ER-101, given that it allows for memory-based control over the ER-101. The 102 has the SD card slot for memories, controls for loop-point changes and grouping loops, opens up new modulation possibilities, and can also do CV recording of things such as joystick, knob, etc gestures. Just as a composing "scratch pad" for grabbing clips for later use, the ER-102 is worth adding; in performance settings, it's damn near indispensable. Of course, there are other complex sequencers out there, though...maybe have a look at Hermod's Squarp, which is sort of a version of their Pyramid sequencer that's been retooled for Eurorack purposes. Something like that would also open up more space, too.

Actually, this is what I came up with earlier today. Would've posted it sooner, but I've been dealing with equipment shipping nonsense:

First up, the layout here makes a lot more sense, locating oscillators in one place, modulation in another, modifiers in a third, etc.

I've added VCA mixing control over the Jupiter Spirits' VCO outs via a pair of Doepfer A-135-2s. The Maths was also replaced by Doepfer's version of the Serge/Ken Stone VCS module, which freed up 4 more hp while still providing much of the same functionality. There's also a very complex random/noise module down by the Expert Sleepers modules; random sources were missing previously, and to get certain organic "irregularities" in pads, a tiny bit of random drift in conjunction with the DPLR's delayed output will do wonders...in addition to all of the other things that Ultra-Random Analog can do. As for the Expert Sleepers interface, I changed that out to an ES-8 with an ES-5 CV expander...this now gives you USB in addition to ADAT Lightpipe, plus four return channels to your DAW for timing, etc signals so that the modular can control DAW functions when needed (in conjunction with Silent Way or Volta, of course). The Doepfer VCAs are mainly for audio level control, so a third VCA/mixer for linear response for CVs is in the lower row. Next to that, you'll notice a Mutable Stages, which can do multiple sorts of modulation duties, and a TriATT, giving you attenuation, CV offsets, and inversion, plus mixing over three inputs.

Rings got shrunk down via a Codex Modulex uRinks...same module, better price and footprint. Optomix remains for LPG and audio summing use, allowing you to mix down to one audio channel with LPGs or to create a stereo result with the same. Dual Zadar + Nins cover much of the Akemie's control, plus a Batumi + Poti are there for CVable LFOs. And while this seems short on conventional envelope generators, the fact is that you can use Silent Way/Volta via the ES-8/5 to send conventional envelopes from your DAW, allowing for more exotic modulation sources here in the cab. As for the little black 1 hp sliver on the left side, that's a Konstant Labs PWR Checker...allows you to see your power situation at a glance, which can be rather useful for a number of diagnostic things.

So...does this version make more sense?

Always start on MG with a case that seems too large. You can always pare things back down on here before committing money to hardware.

VCAs look boring, yes. But they're essential. They're what you MUST USE to create changes in level for both audio and CV/modulation, they can be used as ring modulators (of a sort), they can allow automation of panning, mixing, etc. Lots of people try and get away with one or two...and regret it later on.

That's an expensive case you're looking at, btw; if I configure it with power at 126 hp x 2, I get a cost of UKP 680 with VAT. Then there's this: https://www.ericasynths.lv/shop/enclosures/studio/2x126hp-monster-case-140mm-deep/ which has a much beefier power supply (2.5 A on each 12V rail), better depth (140 mm max!), real busboards, etc. OK, so it's not curvy...but remember: what you can do sonically is the important factor here. Think practicality.

As for the design...hm...let me think about that. The biggest problem is basing this around the Akemie's, because it's such a space hog, but there'll be other ways of making this work...

Lots of problems here...first up, don't consider putting the M32 in the Intellijel cab. The space in that is expensive, and the Moog already has a case. Save your Eurorack hp for things that 100% require it. Next, the tile row; that USB tile is "standard" and Intellijel uses a different spacing for their tiles, so that won't fit in there. Also, the other Line Out tile isn't necessary, since the stereo I/O tile works with the 7U case's 1/4" connections. See here: https://intellijel.com/shop/cases/104hp-84hp-7u-performance-case/ for reference. Also, with the stereo line inputs, the Ears is redundant...you could just make do with an envelope follower such as Plankton's ENVF to extract envelope CVs. That also saves 2 hp. Plus, check your module depths... Intellijel lists a maximum (over the power supply) of 53mm, but the real fact is that you'll need space (even with the Intellijel supply having loads of headers everywhere) for your power connections as well as airflow. Leave space in the case for these.

That ER-101 won't be any fun at all without the companion module, the ER-102 Sequencer Controller. The Plaits and Rings both have third-party builds that take up less space. In a small build like this, it's important to reduce size as best as possible. Also, leaving the M32 where it belongs will help with this. And don't go with replacement panels at the start of all of this; leave the Dual Looping Delay as-is to help avoid potential hassles that might arise from a panel swap.

Speaking of saving space...is the Supercell necessary? It's a very large module at 34 hp, and if you want the granularization of the Clouds, you should also look at a reduced-size version of that. For that, see Tall Dog's uClouds; they also do a uRings, but Codex Modulex does that and the Plaits in 8 hp each for less money.

Now, as for what's not there that should be...VCAs, LFOs, mixers are all pretty much absent here, and are all pretty necessary. Yet another reason why that M32 belongs where it should be...you will need that 60 hp for things that require it. This build also seems to have a case of the "Big Sexy Module" problem; if you build this as pictured, you're going to be fairly disappointed due to the fact that Big Sexy Modules require quite a few boring, utilitarian modules in order to get them to really shine at what they do. I also see very little in the way of envelope generators, save for the Stages, and EGs are important to make some of these (such as the Morgasmatron) do what they're capable of.

Don't look at a bunch of YT videos as your prime info source, btw...while some are good, a lot of them are rubbish as well, and there's no real rating system to cull out the ones where users don't know what they're doing. Simply because someone can post a video doesn't mean they're an expert (although I can think of a few examples where the posters might think they are). Instead of studying these, hit places online that allow you to study classic synths with classic architecture, synths that (in some cases) have been around for nearly 50 years that are still coveted devices make for much better templates than some random musings on YouTube. Instruments like the ARP 2600 (one of the best teaching synths, btw), EML 101 and 200, Buchla 100 series, the Minimoog, etc provide much better models to extrapolate on. Also, unless you have some galloping terminal illness, it's not a good idea to rush the process of learning what to do and then doing something immediately. TAKE TIME to examine options, study, create multiple iterations of builds, etc...otherwise, you'll wind up with something expensive and unsatisfactory. No one gets their first build on MG right. Not even people who've been doing this for decades.

Ultimately, my suggestion would be to delete this and then start again. But before doing so, take some time to look at what you're doing musically and then start building based on the strong points you see in your current gear as well as those you encounter while browsing through MG. You'll find that what you end up with after proceeding carefully will be quite different from what you have here, but at the same time it'll be several thousand dollars well-spent on an instrument that gives you many years of discovery and use.

Chowning FM really needs envelopes galore to be used properly. A pair of Xaoc Zadars with their Nin expanders would team up very well with that and not occupy loads of space. That would give you eight envelopes with full CV and memory storage in 26 hp. However, having a Maths for more modulation possibilities (such as with the SMR) is a very good idea. As for the SMR, yep, it's a very good stereo filter for pads, sort of like an amped-up version of the Korg PS-series triple resonator with loads more control.

As for granular modules, Tall Dog does an 8 hp version of the Mutable Clouds. Also, they do an 8 hp version of the Rings module as well, which would give you back 6 hp to use for something else.

Veils is a decent VCA mixer, but you save a bit by getting Intellijel's Quad VCA...essentially the same thing, $10 less. However, this would not make a decent output module, as you need to step your levels down to line level and, optimally, provide some isolation at the output to reduce noise and crud. In fact, given that both the Clouds clone and the SMR (and the Akemie's, for that matter) are full stereo, what you actually need is a stereo mixer...something with CV over levels, FX sends, and panning. Toppobrillo's Stereomix seems like a good choice, having all of those functions plus a few useful additions such as mutes. Now, with that, you would want to use the SMR then the Clouds clone after the Stereomix so that you can tamper with the sound of the summed pad. Plus, consider adding something for the Stereomix's AUX send/return that can add some stereo as well. One module that comes to mind here is SSF's DPLR, a mono-in stereo-out delay/chorus module, which you can use to fatten up pad sounds selectively and under CV control via the Stereomix's AUX level CVs. As for an output, have a look at Happy Nerding's Isolator, which gives you a ganged stereo level control, headphone amp, 1/4" TRS outs, and transformer isolation (which, I note, you can overdrive a bit to get some nice saturation) in only 4 hp. One argument for having a Quad VCA, though, would be that you don't have any other VCAs in this and those might come in easy for controlling both audio and CV levels as needed. Since you can break out VCAs from that as needed, that would take care of all of your individual VCA needs in this small a build.

As for other additions, I'd suggest some LFOs...Xaoc's Batumi is a pretty useful and space-saving quad LFO; adding its Poti expander would add a few more functions that you'd find handy. You can also swap the LFOs from this with the envelopes from the Zadar for a different control method over the Akemie's, although the Zadar can loop its envelopes. Adding a second poly-VCO might be useful as well...something analog such as Recovery's Jupiter Spirits paired with a couple of simple 4-1 mixers (such as 2hp's Mix) so that you can create a stereo mix out of the triangle and saw waveforms the poly VCO outputs. Perhaps pairing this with a MakeNoise Optomix would be a good idea, as that would allow you to have lowpass gates over this stereo voice for a different note attack sound against the Akemie's, or against the Jupiter Spirits if that makes sense for the patch you're working on. Note also that the Optomix will also need some very basic envelopes; the Doepfer A-140-2 gives you a pair of ADSRs in only 8 hp with CV over overall time.

As for one thing to avoid...try and make do without multiples here, as this build is going to be pretty tight. Instead, use stackcables or inline mults. Passive is fine; since the oscillators are all being fed direct from the ES-3, buffered mults aren't necessary.

Lastly, is this going into a powered case or do you need to add on a power inlet?

How I multitrack is pretty much how I've always done it, even back in the 2" tape days. I'll have separate instruments on each track that comprise the basic parts of a piece, recorded in one pass via an Orion32 (if external to the DAW) or direct in Live (if I use internal sources), or both as needs be. Once all of that's down, then I'll start doing overdubs while at the same time starting to work out what processing to layer onto the initial tracks. At this point is where we diverge from tape technique, though. In beginning the mixdown process, I try and combine related tracks as submixed stereo stems...say, all percussives on one group, bass and pads on another, "ear candy" bits on a third, and so on. By submixing and then subprocessing these stems in this way, you actually have quite a bit of control over the main mix with a minimum of faders in play and a minimum of CPU load because, once the stems are tracked, you can turn off all of the processors you used on the individual channels in the stem, plus your submix is now under the control of a single stereo fader pair and, when needed, you only need to add processing across the two tracks of the stem.

Stems are sort of a given these days with a lot of producers and engineers, but it wasn't that long ago that they were a rare thing, doable only when you have the massive budget needed for extra multitrack machines, tape, etc. With all of that tech out of the way, though, you can generate stems whenever you like and however complex you can deal with inside your DAW of choice. Then, once your stems and solo tracks are ready to go, the mix gets easier...you're not juggling a couple dozen faders all at once. Also, DO automate things such as levels, etc within your stems so that that 2-channel result is exactly the way you need it.

Mind you, this tends to take buttloads of practice to get used to envisioning how your mix should work prior to even mixing it. Ability comes with time and diligence.

Now, as for normalization...that process raises your overall track levels relative to the highest level. So, if it takes +8.5 dB of change so that the loudest peak comes up to your normalization threshold (and never, EVER normalize to 0 dB...always leave "excursion room" of 0.5 to 1 dB below 0 dB in case something gets raised by dithering, codec artifacting, etc), everything in the track gets raised by that amount. This doesn't equal apparent loudness, though. It just means that your peak level is where it goes and everything else went with it.

To increase apparent loudness, you need to use some form of compression. So, let's say you're cutting a track and your peak levels are hitting -1 dB, but your overall level outside of those peaks is about -12 dB. That's a pretty wide dynamic swing between peak and average, so what you'll want to do is to compress that 11 dB swing down to, say, 4.5 dB. Once properly compressed, your peaks should be back at that -1 dB level, but your average will now be raised by 6.5 dB, ergo the apparent loudness of the track is higher. Ultimately, you could even "brickwall limit" a mix so that everything is squashed into 1-2 dB of swing (or less, if you're some kind of sadist), but when you lose your dynamic swing, the track will just sound like a loud band of sound with no variance.

Mind you, all of this is for nothing if you don't have adequate monitoring while tracking and mixing. Especially the latter. Trying to get a good result with a pair of computer crackerboxes is akin to trying to read an important document without the aid of reading glasses if you're blind as a bat at close distances. You literally will not have any idea of what you're doing outside of certain inferences about what the end-result will be on everyone else's listening platforms.

And one other point along these lines: once you have your rough mix set up, you'll want to put your last set of processors on the DAW's mixbus, with the program compression last. That way, any changes to the signal levels caused by equalization, enhancers, stereo imagers, etc will still get dealt with properly just prior to being tracked or rendered.

Optimally, I prefer to break out of digital for initial mixing of stems and then for controlling stem levels; I simply like having the faders in hand for tiny adjustments. And doing this sort of mixing in analog on a quiet system with 24-bit audio (even at slower sample rates) still puts any noise and garbage signals way down in the mix where they'll disappear into the Least Significant Bit ranges when the track is reencoded at 16 bits for CD and other distribution methods. That is, if I want that; sometimes certain noises and noise amounts can actually add a bit of a presence.

Make sure to leave 3hp for the Batumi's expander, Poti. This adds some useful waveform selection and sync functions. As for slewing, you'll have part of the Maths to use for that, when necessary, but a useful addition for that might be Ladik's C-012, which is a dual slew lim with switchable up/down/both CV detection.

Some mixers are AC coupled. Some aren't. The best way to make sure you're not passing DC to your amp, though, is to make sure of this in the source itself. In this case, you're looking for output modules that have transformer isolation or offer balanced outputs (which generally implies transformer isolation). Happy Nerding offers two that are worth consideration: the Isolator is a basic, two-channel transformer-isolated output level attenuator, and their OUT not only offers this, but adds a headphone amp plus a second parallel stereo input which can be used as an effects return, plus you get stereo LED level metering. Also, Bastl's Ciao! does this w/o metering, but has input clip indicators for both stereo in pairs and a potential second stereo (3.5mm TRS) line out and a few headphone monitoring tricks that can allow you to have one stereo pair going to the outputs while the other can be monitored on headphones, allowing you to work on an entire second patch at the same time that the other is playing. Any of these would be very useful.

Not a bad set of choices for an expander cab. One thing I'd suggest for your open space would be another complex function/envelope generator such as the Intellijel Quadra + Quadra Expander mkii, Sputnik Quad Function and Trigger Generator, or the Doepfer A-143-1 (if you can fit a 50mm depth in your case). Since the Shared System will have a Maths, having a different strategy to create complex envelope/modulation curves might be an asset. Also, if we're talking about the original Shared System, you might consider updating to the Rene mkii, as it works together with the Tempi module to unlock a bunch of cross-communicated features. Keep the original, though, as having extra sequencers should never be a problem.

Basically, the Pittsburgh Structure line was designed (and for a time, built) by Monorocket, who had a reputation for making extremely solid cabs with very ample power supplies. The Structure 420 shows this, both in its built quality and the fact that you get an overload and short-protected 6000 mA on each 12V bus, and 5000 mA on the +5. Compare this with the v.3 power supply on the A-100 p9, which is 2000 mA on the +12, 1200 on the -12, and 4000 mA on the +5V rail.

I think you should seriously consider Pittsburgh's Structure 344, though. 5000 mA on each 12V rail + 3000 mA on the +5, same protections + extra RFI filtering, and you get a really well thought out 1U utility row with some very sensible utilities provided. More expensive, yes, but the extras + the form factor (more like a Doepfer p9 in size) make a good bit of difference.

Yup...in fact, I have used (and plan to after the studio upgrading here is finished later this summer) a pair of tube bandpass filters in the analog side of my mixchain which I set up to roll off everything from 10 Hz down and from about 15 kHz up. The filtering eliminates potential low-end losses due to detrimental subsonics and/or DC, as well as countering the 'brittle' sonic issues that result from excessive high-frequency aliasing, particularly in the octave immediately prior to the Nyquist frequency. Plus, this puts 11 active 12AX7A stages per channel into the mixchain to add some extra euphonic even harmonic emphasis, thereby warming things up a tad and deepening the presence of the higher-mids and highs.

Back when the first synths were concocted by Bob Moog and Don Buchla, you didn't encounter a lot of DC-coupled amps. Those came along a little later, as solid state electronics for power levels typically used in amplification became more common. As a result...plus as a result of the idea that you should be able to tap a CV at any point in a patch where it might be present, including the output...the idea that the output stage of a synthesizer should be DC-coupled was rather typical, and even persists to this day with some module designers. And that's yet another reason I strongly suggest to users that they make use of a proper output module, because these tend to be (but not always!) AC-coupled only. When you have a transformer or other isolation device before that final output point, the potential for DC-caused failures drops to zero...and you also get a bit of iron in the signal path that might sound good if pushed into a bit of saturation, plus that also helps with noise and ground-loop issues. Note that this isnt the result you get by simply using typical attenuators to step the levels down for output...that, in fact, is guaranteed to pass any DC that is present on the input side of the attenuator, just scaled down but still quite capable of causing harm.

EDITED May 23rd 2019, 04:06

Also, you may not want "hi-fi". For example, if you had an original ARP 2600 displaying all of its sonic capabilities, you'd wind up with damaged speakers and possibly the amp as well...because the ARP 2600 had DC coupled VCAs going right to the output. And passing DC to your speakers WILL wreck them...but you'll be getting everything the 2600 has to offer sonically!

For that matter, once you've dived headlong into electronic music, you'll wind up wondering what "fidelity" is anyway. A lot of inventive work in the various aspects of the field has come out of mistakes, errors, and general screwing around...and NOT trying to achieve some pristine-fidelity result from the instruments and/or processing. The only place you should be concerned with "hi-fi" is when dealing with your DAW's A-D and D-A conversion so that whatever results you got (be that "hi-fi", "lo-fi", "no-fi", or just plain screwed up) are being recorded and reproduced properly. Beyond that, "fidelity" means zilch in a form of music where there's not exactly anything that you're trying to be faithful in reproducing, and in many cases a result that was a pristine "fidelity" result would be utterly useless.

I'm in agreement with Ronin here...this build is all but useless. There's so much stuff missing that would be essential to a modular's operation that I can't even tell exactly where this build is going.

You need to back waaaaay up and study how synthesis works...not just modular, but in general. This build has audio sources...and then it's missing everything forward from that that should be in the audio signal path until you get to the Rainmaker and Audio I/O. The sole modulation source is the Black VC EG...but without filters, VCAs, and so on, that's pretty much useless unless you want to modulate the oscillators to make funny noises. In short, it'll be a rather expensive and unsatisfying not-really-an-instrument.

If you insist on going with modular, you need to understand how it works before doing the musical equivalent of tossing $3k into your fireplace. My suggestion would be to stop messing with MG for a while, and instead get a copy of VCV Rack (https://vcvrack.com/). It's free, it functions more or less identically to a Eurorack system, and it has a very extensive module set. Learn what does what, how, and why...and also why the UNsexy modules in a build are sometimes more important (when taken as a whole) than those really superduper ones with the blinkenlights and twistenknobs. And if you insist on spending money on a physical device, start with a patchable synth that has most of the building blocks you need built in so that you have a proper device to learn on.

This is a problem that actually crops up on here a lot, btw. Right now, you have a lot of people running around going bonkers over modular synths, thinking they're the new essential (sort of like the hysteria over the Roland TB-303 in the mid-90s)...but the fact is that unless you have some real sonic ideas and goals in mind that you know require something other than a bespoke instrument, and unless you also know the tech that makes those ideas and goals possible by having learned them either via a good text on the subject, various software tools like VCV Rack, or hardware that's already taken care of the module selection process, the end result will usually be a lot of money spent on an unworkable system.

A power supply fuse? I'd get inside that cab if I were you and see if there's not something shorted or misplugged on your power bus or module ribbons. That's more likely, given that you got the synth turned on and used it before the fuse popped.

Right...ratcheting is a rapid repetition of a single stage's voltage settings, generally at some fraction of the main clock pulse. Although, I will point out that while that Koma Komplex is a rather spendy device, it has a lot of possible control routings that can give you this result along with many others...particularly if you have a Boolean logic module (or two) in your build. Also, modules such as comparators can be used to output controlling gates in conjunction with using a sequencer row for timing CVs. Given enough logic, clock counters, dividers and multipliers, extra gate/trig sequencers and comparators, you can make even a fairly simple sequencer turn backflips.

No, that's fine. While you should always have extra current capacity in a power supply (mainly due to inrush current values on power-up), your total module draws are pretty much within the "safe" figures to avoid an overload.

The thing you're looking for here isn't a sequencer...it's a clock multiplier, and the function you're talking about is "ratcheting". Have a look at Doepfer's A-160-5. That module can output multiple clock pulses based on an incoming clock, and the amount of repetitions can be voltage-controlled. Now, as to how to get it to work...that's going to vary from sequencer to sequencer that you use it with. By and large, though, you're looking at clock modulation and logic functions when you're talking about something of this sort. In some cases, you might be able to trigger the ratcheting from a stage pulse. Others might require something a bit more elaborate with some logic gating. It all more or less depends on which sequencer seems like a good fit for you...and once you sort that out, then the next step will be figuring out how clocking should work for you and all of the related fun with that.

Depth in search results

If you sort by depth the actual module depth is displayed in the module boxes beside the HP info.
Modules without an assigned depth will be excluded from the result.
-- modulargrid

Woo-hoo!!! A very welcome change, especially with all of the 40mm-and-less-type cabs hitting the market these days.

OK, let's answer some of this, sort of out of order...

Not only will a cheap mixer cause noise, there are definite sonic differences between something cheap and something that costs more. And the difference there comes from component quality. Cheap stuff (like Ammoon, Alto, Harbinger, et al) cuts corners on components, with the result being looser tolerances, which sort of cascades as your signal path goes through the board. One sub-par component is bad enough...now consider what a couple dozen of them in an audio chain will cumulatively do. Plus, certain mixers have a very specific sound quality to them, most notably the English-designed/made ones. This is what makes a pre-Behringer Midas desk so desirable...but not so much a post-Behringer one, as these don't have the same rounded "English tone" anymore.

Computer line-ins aren't the right thing to use, nope. The culprit here is noise at your A-D conversion stage. This is due to the A-D on a typical sound card (or sound card on the motherboard, depending) being typically unshielded from in-case electronic noise, plus the fact that that connection is going to be a consumer-level (-10 dB instead of +4) line-in and it's also unbalanced, which tends to allow more electronic crud into your signal chain. The line-out isn't as problematic, but to get really good results on recording, you need an outboard interface that's +4 dB, takes either XLR or 1/4" TRS balanced lines, and has a proper ground. And one other point: everything in a recording setup should be star-grounded. By this, I mean that everything you use needs to have a ground that is the same as all other devices, usually done by grounding everything to a single ground point (hence the name). By doing this, you can lower noise and help avoid ground loop issues.

QuickTime is not only the wrong tool, it's also VERY out of date. Use a proper DAW. You already have Audacity, so try recording in that instead. I actually multitrack in Ableton 10.0.6...but I chop loops and clips and also do my final editing and normalizing in Audacity. It works better for that, while Ableton works beautifully on multitracking, track comping, and so on. Ableton is also not the only choice; you might look at Bitwig, which is similar but has some of its aspects more streamlined than Ableton Live.

Now for the last pile of questions...first, EQ. Technically, there's three types: parametric, graphic, and program. Parametric is the type where you can specify the frequency per band, the level at that frequency, etc; you often see these on mixing desks in some form or another. Graphic EQs are the ones with fixed frequency bands with level controls, and tend to see more use in live applications for room correction, but can also be useful for similar purposes in the studio. And program EQs are things such as Pultecs, where you have specific boost/cut stages with their own tailored frequencies, often also working on the overtone spectrum of the selected frequency. This last bit is very typical of the Pultec EQP-1A's low end cut/boost control, where the 'boost' also works on the overtones of the selected frequency, but the 'cut' acts like a normal shelf, with the -3 dB point at the frequency.

For the most part, a program EQ is the only EQ you should boost levels on. All other equalizers should be used to subtract from what's present in the raw signal unless you're using the EQ as an effect in some way. The reason for this is that it's easier to compensate for lower levels of something in a mix than it is to correct levels of some type that're too hot. For example, let's say that one track has a band in the lower-mids that's sticking out, a sonic 'lump' as it were. It would be easier to isolate the 'lump's' frequency and reduce that on that one track than to bring everything else up in various levels and bands to even out the 'lump'. But with a program EQ, what's being done is more akin to "sculpting your mix's tone color"; accordingly, most of the time you'll see program EQs on the final mixbus to do those timbral adjustments.

However, tinkering with EQ without a good monitoring chain...flat, unforgiving response from as low as is feasible in the bass all the way up to the ultrasonic...is basically pointless. It's like trying to read a map, but you've forgotten to put on the reading glasses you need...ergo, you're probably going to get lost. Never skimp on monitors...unless, of course, you're trying to check your mix on a more "real-world" equivalent, in which case you need to incorporate those "everyday" monitors alongside the other, more precise ones. And this, btw, is how you check your mix; if you need to know how something sounds on, say, a typical set of computer speakers, by all means use some of those after you've done your mix on the mains. But if something needs fixing as a result, do that work back on the mains again. Motown studios always had a pair of 6"x9" car speakers in some cobbled-together wooden boxes in their studios specifically because Berry Gordy wanted to know how their stuff sounded in your typical car...and of course, Motown stuff sounds great in the car because of this "check". Headphones, however, are not something you mix in unless you're specifically mixing for headphones.

As for compression, there are again several types. Limiters basically "smash" everything above their threshold level and hold the dynamic limit right there. More typical compressors have various (and often adjustable) settings for how aggressively the compression happens as the desired level is approached, plus what sort of degree of compression (ie: ratio) is needed. And program compressors, like their EQ counterparts, are more for riding gain and "gluing together" a mix while used on the mixbus. As for the right way to use these, first keep in mind that anything over 4:1 ratio winds up behaving and sounding like limiting, especially with a hard "knee" (that "aggression" setting) at the threshold level. To get a compressor to behave transparently, use lower compression ratios and softer "knee" settings, which will then allow the compressor to compress over-level signals enough to fix level problems but not to make the track in question sound like its being "mashed". Unless, of course, that's what you want, since compressors are also useful for adding distortion and overload character to sounds that can use a beef-up.

Program compressors, though...those are a bit different. In their case, you use compression to get the overall stereo level on the mixbus to "float" around the desired track's loudness without exceeding 0 dB. So the meters on a program compressor might be floating at around -3 to -5 dB, and you'll use the makeup gain to bring that result's level up to where you need it to be post-compression. These are a good bit trickier to use well; like anything else in music worth doing, they require practice.

As for what to use...that's up to you, and what sort of sound you're going for. A good place to start, though, would be KVR Audio (https://www.kvraudio.com/), which has a trove of free plugins. You should, over time, be able to find the ones that work for your music and workflow...but again, this takes time, because in this process you're actually tailoring your DAW to be your bespoke recording "instrument".

Hopefully some of that is of use...

Simple: you have output 4 and output other 4.

That's an ongoing problem...when you're dealing with little boutique manufacturers, you're not dealing with a company that can farm out its PCB fab and stuffing to someplace in Shenzhen. Sure, big firms can do that easily enough, but when the "manufacturer" consists of one or two people, supply and demand kick in with a vengeance.

As for stats, well, Sweetwater has a whole website and print catalog section dedicated to Eurorack. If there wasn't a sizable market for that, they wouldn't have bothered.

Ah...now, if the idea of that uFold is to treat the two VCOs, you might be better off trying to jam a Tiptop Fold Processor in there instead. That module gives you dual inputs, and you also get a square-wave frequency divider for Roland monosynth-ish suboctaves. And even though it's 12 hp, it comes in at a lower price than the uFold. The circuit is a tad different, but the results from the Tiptop folder should be more or less the same...provided you only use one input, that is; the ability to unity-mix and waveshape, though, that's a definite plus and puts Tiptop's module into a different level of functionality.

Actually, having a pair of the same or similar VCOs makes perfect sense. If you're using the Dixie II+ as a 'primary' VCO, then the II is an excellent choice as a doubling VCO, since the waveform purity and control behavior should be extremely similar. The Rene choice is also just fine, given that you've paired it with the Tempi; the two of those together are a very potent combo due to some shared functions between the pair. About the only change I'd consider here would be to remove the LxD and the Intellijel Unity Mixer (the TriATT next to it is sufficient), and swap in a pair of Erica Pico LPGs which then give you internal decay envelopes on 'ping' plus some resonance adjustments for better LPG control.

OK, I admit it...the amount of super-useful modules that dropped in the past month was so unexpectedly HUGE that I'm sort of stuck. I got half of the ones for the month looked over...and still made three-plus pages out of that alone.

It's really only a problem for me, though. The deluge of Eurorack has seemingly reached the point where the run-up to Superbooth this year was a flood of amazing ideas interspersed within really good basic modules. So while it became almost impossible to keep pace (and work on my own music, studio upgrading, etc), it does mean that what we're seeing now might be an outbreak of some of the best new modules in quite some time. Brilliant ideas are afoot...the use of embedded processors, the hybridization of analog + digital, modules that literally change the whole game up...these came out in a torrent starting last month. This month, and while Superbooth is going on as I type this, has been just plain jaw-dropping.

So while it's a hellish time to review Eurorack offerings, I think it's safe to say that this is the most amazing time to be a synthesist since Bob and Don cobbled together their first systems. There is such a confluence now of the old, new, strange, and relatively normal that, while it's become nearly impossible for me to keep up with developments for the KICK ASS!!! columns, we have a wealth of new devices out there that're worthy of serious attention. For example, did I ever think someone would kick out a clone of the Korg KMS30 MIDI-to/from-DINsync box? No. But did Pharmasonic do exactly that? Yes. Or something such as 4ms's six-voice Spherical Wavetable Navigator, virtually a synth in of itself? Who could've seen THAT coming? And whole new lines in which everything was a "nailed it!" module, like Starling? What are the odds, really?

So, bask in the glow of many, many shiny new toys, MG denizens. Eurorack has come a long, LONG way since the days when it was just Dieter and a handful of others working with a weird adaptation from test and industrial process equipment to concoct the format. The gamut of manufacturers has exploded, ModularGrid has certainly played a part in ramping up the viability of the format, and the now-huge user base's demands for complexity, quality, and new ideas is being loudly and clearly heard. And all of that together is a very, very good thing indeed!

EDITED May 10th 2019, 23:19

First up, your build's got some problems. That A-135-1 is outside the span of the rails. Plus, if this is a 3U Rackbrute cab, you need to put the Arturia power supply in there because you'll lose 5 hp from that necessary module.

Once those are corrected, my first suggestion would be to move to a 6U Rackbrute, at least while you're sorting out things on MG. Not only is it a good idea to "build big" then pare things down, generative music tends to require quite a bit more than you have here. For example, there's NO modulation sources here (Disting notwithstanding)...and modulation sources are essential to creating the internal variation that generative composition requires to sound effective. I also don't see anything that would deal with variations in timing, such as trigger/gate delays, logic, randomization. There's no random signal source (also pretty necessary in generative music).

My suggestion, ultimately, is that you should probably try and explore generative patch architecture before you jump into the cash outlay on a physical modular. If you haven't already, download a copy of VCV Rack (at https://vcvrack.com/) and load it up with all the free module packs. Then start in there...NOT with a physical device, at least, not until you've got a clearer picture of what a self-regulating generative patch requires in terms of hardware to make it work. Trying to create something of this sort without some adequate research will probably only result in an expensive device that generates more frustration than music. Then bring what you've learned back over to MG, whip up a 6U Rackbrute, and start.

EDITED May 3rd 2019, 02:49

Well, if the objective is to keep this very West Coast, I'd yank the DPO altogether and put in a Maths instead. Since Maths is a complex-wired version of what's basically a Serge Dual Universal Slope Gen, it fits. You can use the two slope gens as modulation, or you can cycle them at audio rates for extra VCOs. Lose the buffered mults; you don't have enough tuning-critical signals in here to justify their expense, and the rig itself is small enough that inline mults or stackcables make a lot more sense here. With those removed, you then might want to put in a 2hp delay to go along with the 2hp reverb (since reverb was key to the Buchla sound, and delay [via the Wilson Analog Delay] was a significant part of the original Serge sound). And since these simpler modules are "appropriate", then you can remove the DSP as being more or less redundant.

Another redundancy might be found with the A-149-1. The Wogglebug is actually an augmented version of the Buchla Source of Uncertainty, so having two of these is a bit much. The A-149-1 is also WAAAY too deep for a Rackbrute cab; you want to keep your module depths around 40mm or less so that you can have space for your ribbons and airflow inside the cab. With that gone, you might then want to add a stereo filter to tone-shape your mixer output; while VCFs that are typical to subtractive synthesis aren't 100% "West Coast", adding something that can act as an overall timbral shaper would be right. Have a look at WMD's Overseer, for example. It's not a "normal" VCF, but if you used it post-mix as a timbral control (either manually or modulated) that would be more in line with what the Buchla 291 is for, especially since you can CV the response behavior which makes it act more like the 291e's "morphing" capability.

Last, something additive as a VCO. The Verbos Harmonic is nice. It's also HUGE and as such, isn't really appropriate for a small build. The Mannequins Mangrove, however, is only 10 hp and while you can't directly address individual harmonics, you still have quite a bit of leeway in manipulating harmonic content with it.

This should be a lot more spot-on with the West Coast model. I am wondering about that MScale, though...if this is in an Arturia cab, wouldn't it be linked with a Minibrute 2 or 2s and not a Moog? Or does the MScale deal with a Moog that's somewhere else in your rig?

Totally different...the A-110-6 is an analog VCO/LFO with the appropriate circuitry for TZFM and quadrature outputs, while the Plaits is a microprocessor-based digital oscillator with several very useful synthesis methods. While it might seem that that might not be a "proper match", the fact is that the A-110-6 is one of the analog VCOs that has the timbral complexity potential to hold its own against the Plaits when in use as a VCO, and has ample quadrature-phased waveform outputs for LFO use that can make the Plaits go morphing-bonkers when using the A-110-6 in that mode.

Which brings up another point: when looking at modules, always also look at how certain modules can synergize with each other. In this case, you could opt for two different sets of possible uses, both with some major sonic plusses to them...but if there's two that are obvious enough on the surface, there's apt to be MUCH more lurking there, waiting to be discovered once they're in a cab.

Another thing to keep in mind is that the Tiptop Fold has a suboctave divider; you can get up to four octaves down from your initial pitch with that. That would kind of disqualify the DixieII+, since you'd be duplicating a function with it that's better-implemented in the Fold. Instead, try and aim for adding more modulation possibilities with the VCO choices while maintaining cost-effectiveness. That Doepfer A-110-6, for example, is $21 more than the DixieII+...but it gives you quadrature outputs both as a VCO and LFO, has thru-zero FM capability, plus both linear and exponential FM. This makes it useful as a modulation source in LFO mode, and as an audio VCO with some very strange FM capabilities. Sounds like something worth $21 to me!

The Tiptop Fold is definitely something to consider, but I'd be more inclined to use it on the modules since the MB2 already has its own waveshaping via the "Brute Factor" control. One thing you might seriously think about, though, would be something more potent as a second VCO to tandem with the STO. Not only would this give you the ability to create nice, thick, detuned sounds between the two VCOs, you can use the dual inputs of the Fold as a de facto mixer, bringing them directly into the waveshaper. My suggestion would be to look at the Doepfer A-110-6, as this would also add thru-zero FM possibilities and the modulated timbral capabilities that brings. Or for around the same price, you could even go with the VOID Gravitational Waves, which would put something more West Coast-ish in with a complex VCO to team with the STO.

That's sort of unfortunate, really...even though having balanced power can be a bit of a wiring hassle, balanced power = CLEAN power. I switched from having that same (more typical in N. America) single hot-leg setup to proper balanced 120 V some years ago, and the results were amazing. You could definitely tell that the noise floor in the studio had dropped, and because of the phasing done by the balancing toroids, various clicks and pops and other intrusive powerline garbage were totally eradicated. And all of my gear worked just fine on that balanced arrangement, as well. True, it's definitely a hassle to wire, because mistakes in wiring that sort of thing (floating grounds, especially!) can lead to a lot of problems that you normally don't see in unbalanced power circuits...but, again, the results were worth the trouble.

FYI, if you're seeing 110 volts across these devices, then one of your voltage legs has a wiring fault. Power systems in Europe tend to use balanced power, which puts half of the total line voltage on each voltage leg. This sounds very much like one of the legs is going directly to ground, which is a serious -- and potentially dangerous -- problem. Again, I cannot more highly stress the need for ground fault testers to be part of any electronic musician's "must-haves", both for studio use and to check live venue power situations.

Complex VCOs are just as capable of doing ambient music as they are for creating gnarly, raucous racket. The deciding factor is simply how you use it. The nice thing about them, though, is that you can easily whip up complex spectra within one module...which was something of a key thing for the Buchla, which is where the complex oscillator idea comes from. In Buchla's synths, the main working method was to build up complex sounds, then run the results through a low-pass gate, basically a tandemmed LPF and VCO under the same envelope's control.

Anyway, the upshot is that you should be able to use a complex VCO however you see fit. The Buchlas were just as capable at creating delicate, atmospheric sounds as anything else, but the architecture allowed you a little "more" in terms of working with the sounds in real time.

Heh...actually, my idea with all of those VCOs is to have them handy as either an audio or modulation source. All of them have that convenient VCO/LFO switch, which I wish we'd see more of as it's super-handy to be able to flip that and alter the oscillator function on the fly. Plus, given that the Gravitational Waves's oscillators can flip functions like that as well, you have the on the fly ability to completely alter the audio-range oscillator in one of those pairs by radically changing the FM rate. I like that; that functionality was one of the more convenient things about the ARP 2600's VCOs.

There's a new complex VCO on the market, just hit in the last month: VOID Modular's Gravitational Waves. It has everything you'd expect out of the DPO, etc, plus an onboard ring mod. The two big differences here are the size and price: 18 hp, $250. You could fit one of these into a DPO's space and only need to clear 8 more hp to put in a second...and two together cost less than a single DPO. Sounds like a win to me!

And yes, you do need three VCOs for maximal possibilities for sound design. Instead of the Rubicon, though...my take would be two of the above, plus a Doepfer A-110-6. That way, you also get TZFM capability along with a pair of West Coast-ish VCOs with a minimal footprint and minimal $$ outlay.

I'll second the #3 above...but only inasmuch as the MU section still needs something to accomodate half-height modules. And yes, even though COTK has a different idea of what MU half-height is, I would figure a "generic" half-height row measurement would work in the same way the 1U tile rows work in Eurorack for both normal and Intellijel format. But at this point where Moon and COTK are both heavily into this size format and there's others inching into it as well, it would seem like something that needs accomodating. A similar situation exists in the Buchla universe with the H series modules plus the 1U "ModuleModules" that Eardrill's got, but that seems more difficult to fix given that each module slot can have either two H-series or four ModuleModules, and this isn't a per-row thing.

That's certainly what I'd do. Fact is, if something has its own case, leave it there...given the cost of a Eurorack case and how much each hp costs, it's best to leave things that're cased in their cases, and use the higher-cost Eurorack cab spaces for things that require them.

In fact, let's look at this for a bit. Assuming both are Intellijel cabs, and not taking the 1U tiles into account, there's 376 hp of 3U space between these two cases. Then the street cost of those cases together is $1248...so a little simple math shows that each hp in those cabs has a pricetag of $3.32. That's not an insignificant number. So when you take a module that comes in a case (which, since the case is OEM, we'll put that at $0 per hp) like the DFAM, and drop it into a Eurorack case...well, with the DFAM, you're using $199.20 worth of Eurorack case, meaning you actually lose that much money by putting the DFAM in there. Not good!

Certainly...that's what the Expansion board was intended for. However, keep in mind that the Werkstatt's pitch CV is set to something very abnormal -- but it can be rescaled for standard 1V/8va. Go here: https://api.moogmusic.com/sites/default/files/2018-04/Werkstatt_01_Manual.pdf and you'll find the recal directions plus a few other things you'll need to know.

This is a rather smallish build, for starters. In addition to the above advice, I would add that you should try for maximum functional density. If there's a 20 hp module whose functions could be found in 16 hp, then go for the smaller size. If you find a VCO module that has two VCOs in the same panel space that currently houses one, then go for the two. As noted, Clouds, Braids, and Z-DSP are all either discontinued or superceded, so you can start by yanking those out if you don't have them already. But here's an example of what I'm talking about...

A Braids module occupies 16 hp of space. Mutable's upgraded version is the Plaits, which fits into 12 hp. But if you look at Codex Modulex's "shrunk" third-party versions of the Braids and Plaits (their uOsc-I and uOsc-II respectively), you'll notice that these are 8 hp, so this means you can effectively fit two of these (or one of each) into the space the Braids currently fits in and, in the process, double your oscillator compliment. Similarly, while Intellijel's uMIDI occupies 6 hp, it only gives you one channel of voice CV/gates. But if you went up 2 more hp, this allows you to use an Expert Sleepers FH-2, which gives you eight assignable CV/gate outputs and two inputs so that you could use clocking on the modular to control DAW tempi, allow a CV to change parameters in the DAW, and so on. Plus, you can expand it if needed, and it has a lot of functions the uMIDI doesn't but which you'll probably find useful.

A couple of other things to consider: first of all, if you're tempted to add mults, don't. Use inline mult devices or stackcables instead of losing functional HP to multiples. Second, if you can find a powered cab to use instead of one that requires the power supply to be placed on the patchpanel, go with that and free up four more hp. Lastly, consider what you're missing here, VCAs being the obvious one. Sure, they're not sexy...but they're essential as they allow level control for both audio AND modulation/CV. Perhaps look at a case with a 1U tile row? Intelljel has these, but they're formatted for their special tile format. Or you could simply try putting everything in a bigger 3U cab...many of us tend to recommend "going big" while doing initial builds on MG, then paring things down from that. Case in point (pun intended), have a look here: https://www.ericasynths.lv/shop/enclosures/studio/2x126hp-monster-case-140mm-deep/ Now, this is not only powered, it also has a depth that will allow pretty much anything to be mounted in it, and if/when you're ready to enlarge your system, you can add a second one of these with Erica's dual-case sidepanels and still have everything in one handy unit.

There's LOTS of options here; don't try and get everything right in your first build, because no one ever does. Use MG as the modular building sandbox that it is, and work out possibilities to the point where you're sure there's no more possibilities to work out...THEN spend the money, as you'll be spending it a lot more sensibly that way!

Sounds right...the thing that a lot of people starting in modular forget are those, and they're pretty essential. Using them on audio is the obvious thing, but the fact is that when you start using VCAs to control things such as modulation levels, or use them as modulation tools to impose a second modulation onto a first, that's when the voodoo that modular has starts to get really apparent. Plus, they have audio uses that aren't so obvious, such as crossmodulating one audio signal with another via AM using a VCA, resulting in some rich ring-modulation-type timbres. They just look boring, that's all.

Yep, VCAs are the missing things here. Consider adding some for audio and CV/modulation, given that you want that sense of gradual shift/flow with that sort of music. I'd suggest something that can break up the two different functions while also providing mixing, such as Happy Nerding's 3xVCAs. At 6 hp, they're small enough to fit in nicely, and they allow you to separate two mixed VCAs from a separate one in case you need to split up the VCAs' functions even further like that.

No list suggestions, but one technical one: try and keep things dealing with audio away from the power supply. That area is fine for modulation sources, CVs, etc, but if there's a bit of noise on your DC that creeps into the P/S, it can also sneak into the audio paths. The MScale, buffered mult, and MIDI interface are fine, but you might want a little more distance for the In/Out and the A-119, especially since both have preamps that can boost low-level audio. Otherwise, this is looking pretty damn good!

Yeah, if you're a routine user of Scala, you'll really like the FH-2. Plus, since it understands USB hubs, you can have both a controller and computer connected via that so that you can have your performance setup intact without having to yank everything to "blow" new Scala tables into the FH-2. And since you're going bigger, you might also look at Expert Sleepers' Disting...sort of a Swiss army knife of DSP-type functions hiding behind a 4 hp faceplate.

Actually, you managed to nail the architecture as best as you could in this tiny a space, which is pretty impressive. However, if you plan to use this with a Beatstep, you might also look at Expert Sleepers' FH-2. This will need a 4 hp expander for 5-pin MIDI use...but if you went with a Beatstep Pro, then you could directly link to the FH-2 via USB, freeing up 4 hp for other uses. The FH-2 uses standard Scala format scale tables for microtunings, and offers a bit more freedom in configuration as the eight outputs are user-configurable. Plus, since it has four inputs for CV or gate, you can get the modular to "talk back" to the BSP depending on how the FH-2's been configured.

A better idea, though, would not only be to use the FH-2 instead of the Yarns, but to move up a bit more in size. An Intellijel 4U x 104 hp cab would not only let you use more than double the present 3U space for standard modules, you'd also have the 1U Intellijel-format tile row in which some other basic functions can be placed, such as I/O, a basic mixer, noise + S&H, mults and so on. The form factor would work nicely with the BSP if you went that route, also. Plus, you can put a 1U Zeroscope in that row, which has a tuner that displays in Hz...which, if you're doing tuning-critical stuff, would be majorly useful!

EDITED Apr 7th 2019, 23:16

It did slow them way down, though. I saw this after they apparently gave up last night, and they'd been reduced from several posts per minute to about one per minute, and when I hit the site they'd seemingly given up about ten minutes previously. So, it might not be blocking them but it appears to be discouraging their efforts.

I'd yank the USB power ports, for starters. In a small rig like this, anything you can do outboard needs to be done outboard. This would give 15 hp, so...Xaoc Batumi + Poti = 13 hp, and a Circuit Abbey Twiggy dual ringmod/quadrant multiplier to use both for ring modulation or as a spare VCA or polarizer. That's where I'd go, fwiw...

No, cases aren't usually included with proper modulars. While the complement above isn't bad, what I'd suggest is to perhaps look at something along the lines of a patchable synth first to get your feet wet and get some understanding about how synthesis architecture works first. By doing that, you'd have a firmer grasp on what needs to be in a proper system and you'd have the system's fundamental building blocks in the form of the patchable, onto which you can expand as needs be. Something like a Moog Mother32/DFAM pair, Soundmachines Modulor114, Plankton Ants!, Kilpatrick Phenol, Pittsburgh Microvolt or Blackbox, or Arturia MiniBrute2 and 2s pair might make a lot more sense from a learning standpoint for right now than trying to build up a complete system.

It's getting there, yep...I still think the Maths would be preferable to just the single VC Slope of the Contour because of all of the internal routings you can do with it to reconfigure it in some very complex ways. Why not pull the Contour in favor of a Doepfer A-140-2 or an A-141-4 for some additional "proper" ADSR EGs? Also, you might consider a comparator or two, since adding those plus a Maths would give you a decent compliment of modulation sources on which you could use the comparator(s) to fire gates when the modulation curves pass given voltage points. Put this together with some logic to work with the Varigate, and you'll have lots of rhythmic mayhem possibilities.