Hello! We've been working hard on our drum machine which includes some of the analogues sections of the Portal Kick/Drum euro module.  About 2 months ago we provided a discussion about how a standalone drum box made lots of sense for how we like to work and some unsurety about whether we would release a euro module containing the 'membrane' subset of this machine. The good news is that we have decided to release the eurorack drum module!  Currently it is 20HP in size and has gotten many overhauls in the VCO, transient section, gain-staging, and EQs.  For those that are size-concious, rest assured that this module's increase in HP is absolutely necessary and is as compact as we feel it should be (there are 15 switches and 18 pots (5 of which are dual-concentrics to maintain playability while saving significant space).  We are glad we delayed the release of the euro version as we know the changes we have made have far exceeded even the original prototypes which we already felt were quite nice.  I think we're about 3 years in continuous development of this circuit and the time spent has been worth it as it has led to more great things which will be coming down the pipeline! One of the reasons in our decision to release the euro version was due to the fact that a full-fledged drum machine in our style may not fit certain musicians' workflows - esp. those that are heavy into the modular for sound design and even performance.  The other major deciding factor was that the euro module and the drum machine would not have much overlap in what they could do and the kinds of sounds that will come out of each. We are finalising the panel layout in the next few days and finishing up all of the ranging and 'external modulation play' (which is also required for the drum machine - which will indeed also be patchable!) before beginning final layout and the pre-production prototypes.  We'll be doing pre-orders once we have manufacturing set up and the prototypes are fully vetted. Happy New Year!RE

Gruetzi!  Hello!  We have a major update for y'all - this time for our Portal Drum/Drum/Kick (name still TBD).  This is a pretty long article and goes from lightweight stuff to some pretty darn fundamental philosophies for us and how that has shaped where we are now.  But let's start with the lightweight stuff: We took an early prototype to SuperBooth 19 that many folks got a chance to try/see.  (but dont' get too attached to that guy :D) This prototype has gone through *massive* changes.  I've said before I wasn't 100% happy with the prototype.  The base kick sound was surely there.  But something lingered that didn't sit right with me.  I couldn't quite pin-point it but I know that my gut is usually right when it comes to these sorts of things.  The good news is now my feelings are good about this thing but I figured I'd give y'all a little background on how it is coming, what changes to expect (!), and some of the reasoning.   Firstly, this module started off for us just to make LF type drum sounds based around at least 1 membrane - mostly kicks.  At the point of the SB prototype, it made really great kicks - esp. those in what we call the 'pillow kick' sound.  These are kicks that mix really well due to the special envelopes we developed for it and have, for us, a very nice artistic quality to them.  They were definitely 'new' sounding to us and we liked that as we were pretty bored with the typical fare and didn't want to do some variation of these.  So this module was good.  Initially, my lack of feeling 100% with it was caused by my thought that it needed to go to different places under modulation.  If we were to make a kick module that made great, static kicks (which is arguable how many kicks do reside in a mix - static, unfortunately (?)), then why would this make a good module unless the modulation was on-point?  Modular is mostly about modulation - using the jacks.  Through this exploration we found that the circuit was far more flexible without losing any of the original kicks it could develop.  Mostly, this was just a matter of tuning it higher or lower.  As not to avoid losing sounds in poorly ranged controls, we felt it a perfect reason to simply add a coarse fine tune.  Keeping the FINE to be the same range as the original module. OK!  Next.  The pillow kick is awesome.  But what if we could take our envelopes but add more power and air movement ala the classic 909?  We do not mean the 909 sound - which this wont really do as it has different FM and AMP envelope shapes as well as a different VCO, filters, transient generation, etc. - but what we mean is can we get our sound to serve a similar functional role on a dance floor?  We explored this quite a bit and we found this space.  We put a switch on there for choosing pillow or more punchy/powerful types of kicks. We then went through and worked more on the various circuit elements: the VCO, click/tick generation, portal (wave folding), EGs, VCAs, gain staging, and the EQ-ability of the drum.  Every section received refinements.  From a 1V/oct tracking HPF and VCO, reformulation of the click-tick topologies (we found some counter-acting controls that hid click/tick too much for our liking), the EGs got more tuning, and the EQ got a massive overhaul.  And so the drum was quite nice!  It was substantially better than the prototype by quite a large margin.  At this point, we'd been working on a way to macroise controls and this is our DynaControl system.  DynaControl is a way to control many parameters with a smaller amount of parameters.  A dimensionality reduction.  In the case of the kick, it could be just 1x dimension... 'how hard you press the kick pedal or hit the drum with the stick.'  Drums are easy in this way (insert drummer joke here :D).  DynaControl was added.  OK!  Let's dig on in now.  I promised this would go a bit deeper! Fun Hardware and its Relationship to Creation Human Empathetic Hardware At this time, we were already at work on our sequencer and also this colour workstation unit.  For both of these, we knew we didn't want to only put them in the modular format.  Why?  For the sequencer, we were working on a highly playable almost instrument-like way to 'play programmings' so we could make hardware (at least) exist on the fun side (we formalise what we mean by this in a bit).  And so this (currently) is looking like it will be a lap instrument like a flat-neck guitar (I miss my Dobro :( ).  Great ergonomics and a box you can just plug into some other system - be it a modular, another voice or set of voices, a MIDI synth, or even just a computer running various plugins.  That became important to us.  To build something like this in hardware means you are really interfacing to a human.  And hardware instruments must be human-empathetic instruments for us. A modular is very restrictive in the ergonomics department.  3U Eurorack is particularly difficult to design for and it usually means making many ergonomic concessions.  For one, we have little control over the shape or how or where it is mounted.  This immediately unravels any work we put into careful UX design.  We are thinking of a human here!  Patching Flexibility and Default Curation For the colour workstation, we wanted a 'make stuff sound better' playground but also to make sure the default 'patching' of the device is well curated.  We also realised this box, too, should be able to self-modulate and be reconfigurable, incrementally, as the user wishes.  We like this semi-modular or 'selective' modular approach.  You start with something curated, then beyond this, you can do whatever you like but it's different than pure modular.  Pure modular puts the burden on the user before any sound at all comes out.  That burden isn't really needed, and there are ways to ensure that the flexibility is at least as good (or in our case, as we mention below, better flexibility).  It started reminding me of systems like the MS-20.  Immediately, out of the box, it's a well curated set of sub-modules.  And then you have a solid starting point and an instrument with a clear identity.  Even the cheap plastic and feel of the machine becomes a part of this.  It's an object with identity.  If each piece of the MS-20 were separate modules, you realise a few things: 1) there is no default curation between the modules, 2) we have minimal control over the UX, and 3) there is no strong identity.  Now!  That is partially what modular is about.  It's about doing whatever you want.  It's very unconstrained.  That is great for certain things, but in our experience, it can be a very laborious and cumbersome process - even just to start having decent sounds come out.  We are modular users - but it routinely takes us at least about an hour every time we go to the machine to work on something new.   And an hour is being generous.  Most times, a solid patch we end up liking and using in music takes several hours to several days.  Some patches took over a week of tweaking.  And you will NOT get it back.  This became a problem at some point in the musical workflow for us, overall.  We aren't saying modulars should have preset-systems - to the contrary - a modular is the ultimate volatile machine and that's where it works best (trust us, we've done the whole presettable-modular path already).  It set us onto a path trying to figure out 'where' these machines exist in the musical workflow.  But first, we spent a great deal of time working on a model of the creation workflow, itself.  We think the following is a very solid understanding of this (it's been active without major tweaks for over a year now and seems to be holding up!): A Creation Workflow Now, folks should realise that we are constantly in motion.  Most of this motion is at a very high-level and has to do with how these machines are used to create music.  Sounds simple enough, but it wasn't entirely clear until we started formalising a pretty stable understanding of how musicians create music with instruments.  How they transcribe ideas, compose/arrange, perform, and curate.  These are all very human things and far, far away from the reaches of low-level circuit design and whatnot.  Farthest to the left is where you begin a journey of creative output.  And farthest right, you are finished the output.  Now think about various creative outputs you've not finished and also those you have completed.  Think about the process you took.  Think about your placement from Genesis to being Finished.  What tools did you use?  What mind-sets did each point require?  And also think about the iterative element - how many times did you go backwards?  You will find that things you have completed require a net forward movement (including any iterative loops).  Think, in particular about all of the projects you've started that have died.  Think about why that might be.  In the 100s of hours we've spent thinking about this and developing this workflow, we came up with some very powerful conclusions.  It now has control over our music and the way we make instruments for humans that want to make music. And so we began looking into where various instruments or tools fit into this paradigm.  They have a location positioned right to left in the timeline but also a width!  For example... look at a guitar instrument.  Where would you place it and what would its width be?  For me it can easily exist all the way to the left - the Genesis - as a pure beginning.  I would pick up a guitar and have no finished music or solid concept. Maybe I'm just noodling around when something I hear ignites a fire (curation).  Now I stay on that instrument and carry forward and refine the concept.  It might be adding a variation or interest or even working on different sections (you see, it does have a width!).  How wide is up to the needs of the tune but also where you find this 'tool' no longer can do what you need.  You are moving to the right now and for that, things get more specific.  There's less fun rigour now - and at some point it's like you are actually working.  For guitar based music, you actually just record the guitar onto some tape.  That's it.  You play it right, how you want, and it's done.  There's no intermediary steps.  You could even argue you do a bunch of basic mixing before it's even recorded onto tape.  You might use the neck pickup vs the bridge for a solo, which sits it differently in the mix.  Anyways!  You ask yourself how far right have you gone with that instrument?  And during this time, are you having a good ol human time?  For me, the answer is yes, I'm having a great time with this object and I've gone very far enough to the right such that the concept doesn't die.  This death is usually the result of not enough forward momentum and energy with a note that fun can revitalise energy.  You've spent too much energy getting to some point in the timeline for a particular intermediate output and the output is not 'good enough' for you to want to come up with more energy to carry it forward.  It's dead.  You are too tired before the thing is finished. Now.  Think about electronic instruments.  Think about your electronic instruments.  The first thing to notice is that many producers of electronic music don't necessarily need to play an instrument real-time.  In modular, you may even develop (program) a completely generative system.  Or you may use the benefits of a sequencer (which even folks who can play the shit out of a piano or other input device still use for its benefits)!  What we found, in general, was that most electronic instruments are not very wide, nor are they necessarily 'far left' devices that are fun to use.  At least not like ripping on a guitar or what not... And hardware electronic instruments always seem to have very narrow widths.  You run into all sorts of walls and so you can only go so far to the right - to the completion side of a song.  So!  We don't want to keep going on this workflow stuff as this article would be much longer than it already is!  The gist of it is that our goal for *hardware* instruments is to 1) make it fun (we are interacting with a human in a creative mind-space) and 2) make it go as far right as we can without sacrificing 1).  3) is being able to go backwards a bit in order to iteratively improve.  Simple!  One thing we notice is that not all hardware is inherently fun or exciting to use. I like using the idea of programming an entire musical passage just using encoders.  Is that fun?  Is it even faster than just using a mouse in a PRV?!  Probably not.  At least not for us.  We discovered this early on when we would spend 5x the time entering data with some encoders or 'unfun' hardware than just hopping on a laptop and scribbling in the notes into a PRV with a mouse.  Now, some people hate computers and have fully hardware setups.  But what we realise is that if you build a piece of HW that tries to rival the power of a DAW, you will greatly sacrifice it's left side creative/inspiring/fun performance.  And so you ask yourself "why?"  Why do I have this hardware touching my fingers?  Why have I sacrificed the point of human-touchable hardware in the first place by saying you want it to control so many different things - now you need hidden menus, shortcut keys, encoders, etc.  Feature by feature, you slowly reduce the machine to a box with little identity and that sits in an unusual space in the creation workflow.  Many of these hardware machines don't even start at the far left.  They start somewhere towards the middle.  So then what do you do on the far left?  Now some hardware is fun only.  It might be placed extreme to the left.  It is fun, exciting, and inspiring.  But what of the width?  Think about the most fun hardware you have.  Think about how far right you can actually go with it.  Have a fun HW sequencer?  Think about what you do to add a musical variation.  Or to refine it.  Or even just to bake it in so you can work on another song section or part.  I'm sure everybody has been here... in modular, the wheels fall off the wagon pretty quickly if you expect too much non-volatility.  And we also realise, if you've thought about the above workflow enough, that Finished recorded music is completely non-volatile.  It's fully baked.  It plays the same every time.  And Genesis is purely volatile.  Think about that some.  This crossfade of volatility vs non-volatility. So!  We take the best of what it means to be 'hardware' and make sure it's on the left and fun - 1) above - and then make sure it has a nice width so you can travel to a reasonable distance to the right - 2) above - and then make sure there is a mechanism for iterative development - 3) above.  Note the backwards arrow in the Creation Workflow diagram above.  Creation requires some movement backwards to refine/improve/change, etc.  Can fun hardware allow this?  (the answer is yes, if you do the HW properly!) .  This iterative part is actually where we had the biggest breakthrough with our sequencer.  Iterative hardware that is still WYSIYG (which means its super fun and intuitive for the human) was elusive for us for a very long time.  But we have seen the light and this high-level workflow has allowed us to reap the benefits of a glipse at these bright rays! Back to the widget! OK, sorry for the off-shoot there but it's important to know where are heads are at.  We can do a separate article on the creation workflow at some point with more detail but this is good for now.  We were talking about the drum module.  At this point in time, we had what we would consider old and dated philosophy in the prototype.  We'd already agreed that HW should be fun an inspiring.  It should also have suitable width.  Why?  Because fun-only HW can simply turn into just toys to have fun with.  But we need to go to the right.  If something fun isn't moving to the goal post, then it gets placed by the wayside in favour of something that is creatively productive - even if those tools are not fun or exciting.  And so we've lived most of our lives in a DAW and using the DAW much farther to the left than we feel is appropriate for us as humans. Well, the big news is that the drum unit will likely be released as a standalone box.  There are so many powerful reasons for this and we simply cannot ignore them any longer.  Here are some reasons: We have more room for making the ergonomics correct. We have more room to add completely patchable submodules - even more patchability of the submodules can be exposed than we could on a modular unit where limited size is a factor in the format. We can place the unit on extreme far left - the Genesis - of a musical beginning. We have come up with a simple, yet extraordinarily powerful method to make the widget 'move' over time as well as a way to curate sounds beyond just one at a time or in an entirely volatile way.  This makes the unit have much more width than a module-only version meaning you can go farther to the right while still keeping that fun anchor positioned at Genesis (before about 2 weeks ago, we had not figured out how to do this without also having the unit's starting point slide away from the Genesis left - but we had another breakthrough here). We can make the unit free-standing in the sense that it can create actual musical lines without requiring any other widgets.  This gives the unit a strong identity and we get to do more of what we are good at: instrument design. We are currently debating whether to release a modular version of the drum.  It's entirely possible and we have several proposals.  One benefit is we can make it more compact.  But since the standalone system has much higher musical and creative value for us - and frankly, much more modular potential than even a module oddly enough - we will finish the standalone design first (as it is more comprehensive), then determine what pieces of this make sense to put in a module. Big news, eh!?  It is for us!  We are darn excited about this and it also gives us a head start on the next unit - the Colour Workstation (name TBD) and even our full sequencer.  They will also all operate and play together nicely as a cohesive system. So!  Break's over!  Back on your heads! <3 RE Team

Hello!The drum module is about 90% the way there. The prototype many tried at Superbooth will get several changes.  It is only till recently that we have put the CV inputs through the ringer and we are coming up with some very interesting uses of the module beyond doing only kicks.  We are getting really good toms and other perc sounds when FM'd with another VCO so this is now something we want to keep in mind to get the last 10% ;)    We do think the prototype is a little bit too constrained and some of the controls do have quite a bit more milk to provide without sacrificing settability.We spent a lot of time at SchneidersLaden after SB trying all of the drum modules they had.  BLD, Entity, Noir, Erica stuffs, Hex, and so on.  It's the case that we work in a sort of isolated fashion, which is nice because it results in less bias.  And I think we've achieved the benefits of isolation in terms of what this module sounds like.  The biggest thing I noticed was how much of a sound design module it is.  A good character is surely also there, which I'm always paying attention to - especially for a module like this where the main point was not to do another 808 or 909 style thing but something new.  I also think some of the ranging is very good.  When compared to the other drum modules, much more constrained (some modules go way too far at the cost of being able to find the right spots).  These experiences give me a few ideas to further push this module.  But I would never give a knob a range that was too hard to dial in as kick drums, esp. with all of these tweaks does require some hand in subtlety to 'tune' it into a mix properly as opposed to having to do it in post (which was another design requirement going in).

We've been testing many variations of custom knob designs for our future products which also includes inclusion of dual-concentrics to achieve density without sacrificing ergonomics.  We are very close!  We have some prototypes being printed out of resin then we'll fire off an order with Kilo International, a company who makes extremely high-quality machined knobs.  And check out the render, courtesy of Mr. Hannes: And the resin prototypes:

We are currently working on a single-track sequencer but are playing with ways to make a multi-track machine that retains most of the fun of this WYSIWYG machine.  The realisation is that once you go multi-track and more importantly: if you have a directed musical approach (ie. you know what you want to program in), the DAW is extremely hard to beat. It's always faster in editing 'whatever.'  A HW UI to enter 'any kind' of data tends, really, towards the fast, generic computer mouse/KB/shortcuts once you need that hardware to enter in all kinds of data and not just one parameter per one hardware element.  We used 2 OTs for sequencing and even that was just wasting time once we reached that threshold of 'I need to do these specific things to the sequence.'  Even more importantly, this HW still had a ton of constraints - which you really don't want when you're getting closer to the completion of a song because of the directed nature of the work during this side of the process.I would say use hardware for the creative curation fun end of things. To do sequences you can't think up on your own or create your own sequencing paradigm via patching in a modular environment.  We've been down this road and done lots of testing and prototyping trying to do DAW sequencing in HW and it ends up being a compromise not worth putting in HW because you either make an enormously huge machine (to which point the UX begins to fail for ergonomic reasons) or you make a generic 'layered' input method.  We're still working on a way to do a multitrack seq but there are some severe restrictions you have to place on the machine - without those restrictions,  we'd rather just 'prototype' one up in Max to do what we like or just go right to the PRV in a DAW.  We are designing in a way that says the DAW is still a part of the song making process.  And this unburdens the design significantly.  The question is really how far into the workflow we can go with a HW sequencer before it falls apart trying to do too much or do things in a worse way than can be done in a DAW.  We also want to make sure this 'jump' is somewhat up to the user by making sure connectivity to a computer is quick n easy. There is a workflow crossfade most musicians need to go through: Creation/inspiration/fun <x>'Work' doing the nitty gritty to complete a full song. The former is where you want creative HW.  The latter is where, for us, the DAW kills it.  There is a similar crossfade layer (well many, really) from left to right in terms of the balance between: Open-ended 'suggested' ideas <x> "I want this exact thing." The nice thing is that with fun hardware, you listen and curate more purely because you are operating in a liter headspace - because it's fun to use creative HW moreso than a computer/mouse/screen.  We spend a lot of time on a computer anyways - esp with all of the non-musical CAD, simulation, and other technical design work.  So breaks from it are quite appreciated and inherently fun!  We keep this mental space sacred and our machine strives to have this reverence too!If you have to stay entirely in hardware (most folks want the potential reliability/simplicity of a non-computer-driven approach), have a look at more comprehensive solutions too!  Things like the Cirklon or even the OT (because what they do to audio is great on the creative end!).  Another trick we used to do is to play into these machines with a keyboard or even fire off MIDI from a DAW and record it in. Maybe you can think of your own, personal workflow crossfades <x> and see how you would like to work throughout the process.  For us, our own preferences and understanding them really helps not only design of new machines but also what works for what when making a song. Have fun!

We did some noise testing on our Low Loss Distribution Boards.   We've created some high performance distribution boards for Eurorack which feature 5x 1/16" thick copper bars that run the full length of the boards.  We feel we've struck the perfect balance of portability/compactness and performance.  We are also solving the power cable issue in Euro with these as well.  They host 26 'Rabid Elephant' Samtec connectors which are properly gold-plated, keyed, shrouded, latching, and just awesome, in general.  They are fully compatible with Eurorack systems. The Rabid Elephant Low-Loss Distribution Boards intend to reduce or completely eliminate the issues of noise in a modular system.  And they do just that.  Here's some proof: THE TEST Since throwing around theory doesn't mean all that much for the average user, I decided to do what I would call a real-world test. To do this test in this manner would be difficult for many because (1) it was a PIA, and (2) you'd need a solid control for the test and that is not possible in many cases (changing from a ribbon cable to, let's say, a PCB system means other things would need to be changed too... and then our control isn't all that great). Since the boards I've made can be populated with and without the copper bars, I thought it a perfect situation to do a before and after test... without then with the copper busbars installed.  Keep in mind that even without the copper bars, these boards are already much better than most distribution systems you'll find.   There are 4x Low-Loss Distribution Boards ('LLDB' from here on out) in my 18U 114HP 'suitcase synth' that I used for this test.  The power supply is a dual linear system based on some modified Power One supplies. I wired up the whole synth with 2 LLDBs without the copper busbars installed.  This is the 'before' test. For the 'after' test, I soldered in the 10x copper bars into the two bare LLDBs and reinstalled everything exactly as it was before. I put the same modules in the same case locations, same header locations, same patch, same PSUs (linear), same recording levels, same wiring to/fro the busboards, The results of the test can be heard in clips below. They are recorded right from the synth, into an RME UFX. The first half of the clip is without the busbars installed (just the PCB) and the last half is with the copper installed. The recording is boosted a whopping +70dB to make the noise very obvious.    Below is the noise test before and after but with no boost... (this is how it would normally be recorded).  Not as bad, of course, but it is definitely there.  Even at this low level, the high-pitched bleed is indeed problematic and would require a notch filter to remove in mixing... which I did have to do because I recorded something in the time before I put the additional bars in.  I can hear the bleed noise *easily* when playing through a PA, headphones, or studio reference monitors.  Try it for yourself!   Here are the spectra using Ableton Live's spectrum analyser.  The same analyser you'd likely use when mixing a tune inside a DAW: Without the bars installed: And with the bars installed: Here's an A-B video with the spectrum. TURN DOWN YOUR VOLUME IF YOU JUST LISTENED TO THE UNBOOSTED CLIP!: We plan to manufacture the LLDBs though timing is unknown.  We may do a Kickstarter to fund these to get them done sooner as they are not in the standard product pipeline at the moment. Have fun and be sure to spread love and creativity! Copy blog RSS feed url here

Introduction I love this VCO and that's really something for me to say... if I'd built this oscillator, I'd be very proud.  The WR manual's are a bit abstract and that is fine.  It allows users to decode it into our own ways like I'll do here.  A scope is your friend with this module... or maybe your ear is enough if you're very coordinated in that way.  I like writing things down.  I have shit for brains so it also helps me to read this again in the future when I've forgotten something. More to come... I just got this thing!  And now have 2x... so maybe I'll have some tips using both together at some point. Let's have some fun... Overview This module is a VCO with 'play' in the retriggering of the pulse generation.  Because of this, you get interesting 'resets' or glitches and even divisions for interesting effects. The actual waveforms produced (with exception of the simple square OP) should likely be thought of as a pulse-train instead of a typical saw/tri/sin, etc.  This helps make more sense of some of the controls and what they do.   Constant Formant & Constant Wave Modes Users may be unsure what the switch CONSTANT WAVE and CONSTANT FORMANT actually does.  Just flipping the switch with a constant pitch or no FM modulation may seem like it does nothing in certain scenarios.  And rightfully so.  You have to change the pitch to see the differences... so that's the first bit to realise. I like to think of it like this: Do you want the wave pulses to 'run into one another' or not?  If you do, put the switch in CONSTANT FORMANT mode.  This locks the wave shape's pulses to a fixed width. On the other side, if you want the waveshape to look the same across any frequency input, put the switch in CONSTANT WAVE mode.  This effectively 'squishes' (as you increase frequency) the whole thing in the time dimension... just like a normal function generator or VCO would as you increase pitch. An Experiment to Demonstrate the Difference Constant Wave Set to CONSTANT WAVE and set BARREL to 12 o'clock, set FORMANT to just a hair CW of 9 o'clock (9:30 let's say), no inputs, start with PITCH knob fully CCW.  Patch output from the FORMANT output. Look (and hear) the waveform.  It will have some shape.  Now slowly increase the PITCH knob CW.  Look at the wave shape... looks the same doesn't it?  Keep going CW with pitch, rescale scope if you need.  Same waveform. Constant Formant Now put PITCH knob back full CCW and flick the switch to CONSTANT FORMANT.  Do the same experiment by increasing PITCH knob CW.  You should also hear the 'smear' almost like a little sync when the wave form shapes or pulses start to collide and combine.  Be very observant of this and go slow increasing the PITCH knob. Got it!  Great! Air Air is a clipping stage and VCA.  Nothing too crazy here but keep in mind that this oscillator isn't inherently very 'squarey.'  But this is just a matter of chopping the tops of those tris/saws/ramps off and you'll get a decent enough sounding square.  Use the AIR for this purpose to great effect.  Abuse it, in fact.  Don't think of it simply in terms of something trivial like "do I want distortion."  Think of it more of a waveshaper and you'll likely use it more.  But that's just mental  I note soft clipping starts happening around the 2 o'clock mark of the dial. And since it's a VCA, it will reshape based on external modulation signal shapes.  This input alone adds a very wide variety of shapes when coupled with external VCOs especially those that are harmonically related to the fundamental.  The gray capped dial is an attenuverter for the AIR CV input.  0 gain is dial centred at 12 o'clock. The AM of this module via the AIR jack is really nice sounding.  Try audio rate modulation of this input for sure! Formant & Barrel I've got these two listed together because at some point, you just cant separate them. I'll explain what they each do first independently, then how they interact.  To start, put the VCO in CONSTANT WAVE mode. Formant (independent explanation) FORMANT controls the width of the pulses.  The more CW you turn FORMANT, the thinner the pulse all the way to an impulse train.  This increases the brightness of the spectra.  When learning this control independently, be sure to set the Barrel control to full CW.  You'll know why later. Barrel (independent explanation) BARREL controls the 'tilt' of the waveform pulses from a ramp at full CCW, to triangle at midway, and to a saw at full CW. All Together Now The first observation that you'd want to make can be found by doing the following experiment: Blanking Count 'Slope' Experiment Set PITCH to the line between '80' and '320', set FORMANT to 12 o'clock, set mode to CONSTANT WAVE, set BARREL to full CCW.  Make sure AIR is set to 2 o'clock or less as to not clip the waveform. You should have a nice ramp waveform (this is important).   Now Slowly rotate the FORMANT control CCW while looking at the waveform on a scope until you hear the pitch jump down.  Stop there!  OK.  Something weird has happened.  Let's talk about it.   Note that it did this pitch jump as soon as the rising part of the cycle (the ramp) touched the vertical falling edge of the previous cycle.  Just queue this observation... it's not enough to fully understand what's going on... right now we just know that the rising part of the cycle touched the falling part and we got a pitch division... let's continue! You're probably wondering why I called this the 'Blanking Count Slope Experiment'?  Soon!   Back to more experimentation: Set the FORMAT control back to full CW and BARREL to full CW. Now rotate the FORMANT control CCW and observe.   The waveform pulse's decay gets longer as you go CCW.  At some point, it will take so long to decay, that it turns into a DC level and you wont hear it.  For kicks rotate it full CW as well.  It will turn into shorter and shorter pulses.  We see that FORMANT acts as it was described above in the 'independent explanation' and there are no pitch divisions like before!  Why not!? Let's experiment MORE! Set the FORMANT control back to 12 o'clock. Remember, the barrel controls the 'tilt' of the waveform.  This is another way to say that it changes the ratio of the slopes of each sides of the pulse.  At 12 o'clock, it is more or less a triangle which means the slope of the rising portion equals the slope of the falling portion of the cycle.  Turning BARREL more CCW, it gets more rampy and more CW from 12 o'clock, it gets more sawey.   Let's start with BARREL set to 1 o'clock.  This gives us a waveform whose pulse has a just a slightly faster rise time than fall time. Rotate FORMANT CCW very slowly. No pitch division.  Hmm.   Set FORMANT back to 12 o'clock.  Let's now nudge BARREL to just shy of 12 o'clock... maybe 11 o'clock.  This changes the waveform now such that the rise time is slower than the fall time.   Now rotate FORMANT CCW very slowly.  And there's our pitch division again. NOTE: In this video, I think I said 'reset'... don't think of it as a reset a la oscillator sync! Now we have enough data to make a reasonable explanation.  I'll explain it in simple terms (no circuit or too much technical lingo!) If a previous cycle's pulse collides with the next cycle's pulse AND the previous pulse (at this collision point) is falling FASTER than the rise time of the next cycle's pulse, then a division event is instantiated. And for the observations with no pitch division, we may say: If a collision happens but the previous cycle's pulse (at this collision point) is falling SLOWER than the rise time of the next cycle's pulse, then there is no pitch division. So basically, you can only divide pitch when the pulses looks more ramp than saw.  This is why you don't get pitch divisions when the BARREL control is past 12 o'clock ish. But what about these pitch divisions... we need to investigate these a bit further, I think. Elaboration on the Pitch Division The pitch divisions can be viewed as 'counts' where the FORMANT pulse train will 'wait' n cycles of the master clock (square output) before firing the next pulse where 'n' is mostly an integer*.  The number of counts to wait, 'n,' is incremented +1 when the pulse length exceeds the length of 'n' cycles of the clock. If the pulse is in length '2.78 cycles' of the master clock, then 'n' is 3 and you'd see a pulse every 3 cycles of the master clock. If the length of the pulse exceeds 1 cycle of the master clock, then n = 2.  You'd see a pulse every 2 cycles of the clock. When doing experiments with all of the pitch dividing that is occurring, do yourself a quick check from the FORMANT output to the SQUARE output.  The SQUARE output is the main core frequency.  The FORMANT is after any processing/retriggerring.  Put them on a scope at the same time and you'll see about this whole blanking counts thing. *There is some 'wiggle' between these seemingly 'binary' zones of division and you can spot if you are very careful with knob movement.  Here you get lots of interesting Thomas' English Muffin things that happen.  I'm guessing this is some 'loose' comparator and when the pulse duration is too long (and of correct polarity), the pulse train cannot retrigger and so it seems like it is waiting master clock pulses.  Curiosity!  I think the fact that this transition zone ends up being 'semi-snappy' is why the analogue domain is so wonderful. BTW, the panel notes: utone.  I'm guessing that's 'mirco' I don't know.  I don't think this term is required and adds unnecessary vocabulary.  It's also on a weird spot of the control making it seems like you dime the control to get that 'mode.'  I'm assuming this is supposed to describe the pitch division phenomena we are observing.  Either way, I'll not be using that term here. If anyone, including the maker of this VCO, wants to correct me, just contact me (PM33AUD) over at muffs! Details/Blurbs from Muffs Think of it as a primary oscillator which feeds the SQUARE out. Then drives a formant generator where the panel controls create interaction between the sections. Sync is soft. Output stage is waveshaped VCA. FM w/ VCA is linear input. Exponential FM via the PITCH jack.  A further note on the linear FM input- it's AC coupled for inter MANGROVE connection. With high FORMANT settings there can be a large DC offset in the FORMANT output so AC made sense for best pitch tracking. For those after DC shifts or linear LFO input there's a tiny solder jumper on the back to go DC. Easy to mod and easy to revert.The FM INDEX vca is normalled 100% level. Designed to attach an envelope on CV, it drops to min volume when 0v is present, and has exponential response for snappy harmonic braps.  Copy blog RSS feed url here

Disclaimer: I am just a user of this device.  It is a rather complex module with tons of useful information scattered throughout 50+ page threads, the manual, patch examples/demos, and tutorial videos.  It is a bit overwhelming to find and sort through all of this data.  So here is my reference for this module hosted on Confluence here, which is a decent enough tool for this purpose.  I've provided 'in my own words' explanations of many things from the point of view of a user.  I've tried to credit the original source as much as possible.  If I've missed a reference, just let me know and I'll add it in!  Also, I try not to put too much stuff that's in the manual.  It's not intended as a replacement for the manual.  Read that sucker first! Phase & Frequency Modulation The heart of the oscillator is a digital counter that increases its count once every clock cycle. The higher the pitch input the higher the amount the counter steps. When it reaches its maximum value, the counter gets reset back to zero, to start a new cycle. So, the counter output looks like a sawtooth waveform.  The output of the counter is what we call the "phase". Modulating the phase means adding or subtracting to the counter output. Frequency modulation amounts to changing the step size that counter increases on each clock.  The "phase" is used to address the wavetable memory. This is the shaping process, where the sawtooth shape of the counter output is converted into the various other types of waves. As the phase goes from zero to the maximum value, the shape of the selected waveform is traced out (e.g. a sinusoid could be stored in the wavetable and would be read out as the phase goes through its cycle).  In effect, what is happening is that the phase sawtooth is being "shaped" by whatever is stored in the wavetable. But we could use some other signal than the counter phase signal to address the wavetable. For example, we could use the MODA signal to do this. If we do that then the effect would be to "shape" the MODA signal with whatever is in the wavetable.  This is exactly what you would get with a waveshaper module (such as the uFold) when you input an external signal as compared with inputting a sawtooth. The advantage of using the shapeshifter as a wavefolder in this way, over say the uFold, is that you now can have 1024 different wavefolding effects instead of just one.  ~jjclark Some background on PM/FM in the digital domain: http://electricdruid.net/direct-digital-synthesis/ http://electricdruid.net/phase-distortion-synthesis/ https://moinsound.wordpress.com/2011/03/04/frequency-modulation-or-phase-modulation-synthesizer-technologies/ https://en.wikipedia.org/wiki/Phase_modulation http://msp.ucsd.edu/techniques/v0.11/book-html/node87.html (in pd) https://www.muffwiggler.com/forum/viewtopic.php?t=126309&sid=eb00196381374bdd718daa4051c17a76   Tuning, Ratio, Quantize Coarse/Fine knobs adjust the pitch of both oscillators.  Ratio knob controls the pitch of OSC 2 relative to the Coarse and Fine controls.   The Quantize button is not for quantizing in the sense of quantizing to musical 'notes' but rather for low integer ratios useful for providing clean FM. Combo SYNC Patch Examples & Tips - reverse sync can make harsh sounds smoother  - hold sync gives nice gating effects  Hold Sync Rhythm Here's a nice industrial rhythm patch example. It uses holdsync to give the rhythmic stuttering. A constant rate lfo is fed into the sync (about 100Hz). Osc1 shape is shifted manually between Noise2, Noise3, Noise4 with some knob twiddling on the Shape1. The pitch input is playing a little (non-sequenced) melody.  https://soundcloud.com/cylonix/ss-industrial-driving-beat https://soundcloud.com/cylonix/shapeshifter-21sync-song This is an example of using the 2=1sync mode in the Shapeshifter. The entire track is made just by wiggling the shape2 knob (and the Coarse knob at the end), and selecting different wave banks. Osc1 is set to LFO and bank1 is mainly set to Basic2 and BasicRec. The output is taken from Osc2, and bank2 is mainly on Grain1.  1=2 and 2=1 sync modes are some of my favorite sources of fun!  [edit - there is reverb on this track, that is not from the Shapeshifter - everything else is!] Pulse MOD A MOD B This is a bit weird... multiple things can be modulated by the MOD B jack and it starts modulating when you enter the menu.  How to turn off modulation for certain things though?!  OK, figured it out... if you want to control multiple things with MOD B (these are the buttons on teh right BTW when you click them twice), you click the button you want twice - you'll see the +/- mod number change based on IP - then while this menu is up!!! click another button twice and it will start modulating that too.  To have it not modulate, you have to go to that modulation destination (click the button) and get it out of the MOD B mode (the 2nd page).  Then you can click another button and repeat as needed.  Really weird! Chord Selection using MOD B To have MOD B choose chords, go to the last chord option in the chord menu.  It will have a little * on the screen. Morph Mode Holy shit, this is where things can get really squirrelly (from a usability perspective and also the actual sounds that come out).  So I'd recommend skipping anything 'MORPH' until you've gotten around the other features - in particular, how to save/load presets, MOD A and MOD B inputs, combo modes, etc. Fundamentally, MORPH is very simple.  You can do PARAMETER (!!!) morphs between the 'panel' and the preset BUT ONLY WHEN MORPH MODE IS ON!  Now, turning MORPH mode 'on' requires some explanation: The simplest is when the encoder is pressed and you choose the MORPH button on the left.  That's easy.  But there are some modes that do not require this to be 'on' and therefore, it's hidden from the panel.  That is where things are very confusing, or can be.  The SS will do weird modulations and you won't know why.  I've been down this road... trying to figure out why MOD B was changing the sound after going through the common MOD B destinations (combo, tilt, etc). Chord You can create progressions using different chords by setting the same patch but with different chord voicings/types stored on separate presets then step/change presets for each transposition.  NOTE: the newest firmware doesn't require this workaround (though still useful for some situations)... you can now select chords directly with MOD B input.  See MOD B. MULTI Patch Examples & Tips Try different MULTI settings with the wavebank set to one of the Chip or 2Tone banks. Play with (or modulate) the Shape controls.  - jjclark Libra wrote: How are people using the multi mode? try putting VCO2 in LFO mode (becomes LFO2) and the multi to something greater than 1. Now LFO2 will be a very complex but repeating shape, like a complex multistage envelope. You can use sync to trigger this on beat.  Have it modulate pitch or patch it out to modulate MOD B or the Fold parameters. Even more fun is to have both osc1 and 2 in multi8 mode, and set the combination mode to ring mod. Set ratio quantize on and turn the ratio way down. Tilt Patch Examples & Tips Acoustic Bass With osc1 set to a sine wave (Basic1 bank, shape1 fully ccw) use an envelope to modulate the TILT parameter. This gives a nice acoustic bass.  jjclark Delay Patch Examples & Tips Looper When using echo, plug a cable into the pitch2 input and leave the other end unconnected. This will break the pitch normalling to osc1 and keep the osc2 freq constant. Set osc2 to lfo. Set the echo level to 25 and double click the echo button to enable modulation by modb. Then while playing stuff turn the modb knob fully ccw then quickly to full cw. This will hold whatever was just played and endlessly repeat. To get more stuff turn the modb knob quickly ccw and then back to full cw. This lets you overdub and loop. The loop length is set by the period of the osc2 lfo. jjclarke Stutter Effect I've just started using the delay, and its quite a lot of fun. Put it in percussion mode, with OSC2 in LFO mode, and the delay becomes very long. Modulating the delay depth with MOD B can get you stutter effects. So much fun, and it's great to have bonus echo if you need it.  Percussion Mode Folder Patch Examples & Tips Yes, adding an offset to the Folder input will make the folding asymmetrical. -With OSC2 in LFO mode patch it into the FOLD input. Now adjust the FOLD amount and take the output back to the FM1 (or any other mod input). It creates really complex modulation that can vary in interesting ways depending on the amount of folding. Some of the LFO wavebanks work really well for this.  Vocoder Patch Examples & Tips Vocoder mode is endless fun: set ModA in to be vocoder modulator and you can send it all kinds of stuff, filtered noise, and even basic waveform sweeps will create filter sweeps and phaser sounds. You can even feedback Osc A out back in and with enough gain get weird vocal sounds Preset Saving/Loading OK, read this, for sure, since it's not super clear.  "When you press SAVE while in preset step mode it will just take you out of step mode. You have to press SAVE again. This will give you the confirmation display (Yes or No). Select Yes or No and then press SAVE once more." Patch Examples & Tips Saving Hi, sorry you are having so much trouble. If you are able to save a preset to one of the slots from 13 through 64 then you should be able to save a preset to one of the slots from 1 through 12 in exactly the same way.  For example, to save to preset #3, first adjust the various parameters and knobs to the way you want, then press the rotary encoder to enter preset mode (red led on). You should see the display saying "UserPr 1" (or some other number if you had changed this earlier). Then use the rotary encoder to change the preset number to 3. The display should read "UserPr 3". Then press the SAVE button (top left button). The display should now read "Save? NO". Turn the rotary encoder, and the display should change to "Save?YES". Press the SAVE button again, and if successful, the display should change to "Saved! 3".  Preset Advice When playing with the preset morph or stepping, make presets that are either very similar to each other, or very different from each other.  jjclarke Unique Preset Stepping Is it possible with clever preset programming to get the pulse out to step the presets? Yes, you can do this. The FOLD out is also interesting to feed back to the sync input while stepping presets. In this case turning the FOLD knob changes the pattern. Making it sound more 'analogue' The SS is very precise, out of the box.  But it also has extensive modulation capability.  Exper rightfully states: ...add some subtle modulation to your Shapeshifter. A little movement in waveshape and a tiny amount in external fm makes it a little more 'analog' if you need it to.  The designer, jjclark also continues: To get the shapeshifter sounding a bit more analog try taking the output from the FOLD output, with the FOLD control turned up just a bit, so that there is no folding but just a touch of saturation.   Also, when using the chord mode, turn up the Overdrive setting. This provides a nice saturation which thickens up the sound.   Another tip is that if you are wanting square wave or pulse type waveforms use the PULSE output instead of the OUT1 or OUT2 signals. The PULSE output has a higher output bandwidth than the OUT1 and OUT2 signals, so it gives a brighter sound.   Calibration Is your VCO tracking 1V/Oct? If it was shipped without being tuned (by accident) then the base pitch might be higher than it should be.  Here is how to callibrate:  If you have a uScale or Quantizer:  You need a variable voltage source (ch2,3 from maths, triatt, etc,) and plug this into the uScale. Select only the first note of the uScale (so it will only output 1V octaves) and turn your voltage source all the way to zero. The output of the uScale should also be zero. Patch OUT A to PITCH1 of the Shapeshifter.  If not:  Patch something that generates voltages like a Triatt or Maths etc. and turn it all the way to zero.  On the back of the shape shifter press button#1.  Now turn up your voltage source just enough so that the uScale is putting out 1V (the next octave up from 0) (You will hear it on the Shapeshifter if you are listening to OUT1). Then press button #2 on the back of the Shapeshifter. You should hear the Shapeshifter instantly drop in pitch. Now sweep your voltage source through several octaves and you should hear them in tune.  OR unplug your voltage source and connect to a voltmeter. Adjust it so that you get exactly 1V, plug it back into PITCH1 of Shapeshifter and then press Button #2. Firmware Update sdfsdf - with osc2 in LFO mode, turn up the MODA attenuator knob. With nothing plugged into the MODA jack this phase modulates osc2 with osc1, and makes out1 and out2 into a nice stereo mix.  - put osc2 in lfo mode and use out2 as a modulator for everything and anything.  - with osc2 in lfo mode use the ring mode combo mode to get cool envelope effects (good in perc mode - the lfo is synced to the attack so osc2 becomes another envelope for osc1)  ——————— It can be really rewarding to start with a simple, concentrated patch that focuses on particular capabilities. e.g. just focus on nice chords or clean FM (both banks in BASIC1) or exploring the Mod B modes etc.  Some patches I like:  -With OSC2 in LFO mode, patch it into Mod A while in Vocoder mode. Activate Chord mode.  Set the OSC2 Multi value to something greater than 1 and play with the ratio to get a nice complex modulation source. It can sound even better if OSC1 is also in LFO mode.  -With percussive mode active you get an overall amplitude decay on OUT1.  You can use OSC2 in LFO mode as a second envelope if you select wavetablebank:LFO3. It will cycle so make sure to dial in a frequency that works with your decay value of the percussive envelope. Patch OSC2 into FM1 to modulate the pitch. Use a trigger at the sync input to trigger and sync both percussive envelopes (pitch and amplitude).  It makes great kick drums which you can morph into other percussive sounds. ——————— The pulse output was something i most looked forward to, and it doesn't disappoint.  A great way to get unquantized loops for odd drums or for clocking sequencers.  As stated above, the combo modes are super in LFO mode. Working in this setting along with the pulse output make the shapeshifter almost more interesting to me as a modulator than a sound source. ——————— Here's how to do those euro synthpop bass sounds with the Shapeshifter:  - Chord mode on, with chord type set to unison  - Detune set to 02  - Perc mode on  - Decay set to 62  - Overdrive set to max (99)  - Echo set to 35 (or as high as 45)  - Osc 2 set to LFO, with an unconnected cable plugged into Pitch2  - Adjust the Ratio knob to match the echo time to your song tempo  - set Quant off  Almost any waveshape will give an interesting sound, and they are all quite different. Changing the bank and Shape1 controls give a wide range of different sounds. I like to use Grain3, Misc1, VidGm4.  Here's an audio clip of various waves with a simple synthpop arpeggiation:  https://soundcloud.com/cylonix/shapeshifter-arpeggiated-bass-demo ——————— I posted this in a separate thread a few months back, but here's a solo Shapeshifter cranking out grains:  https://soundcloud.com/the-february-thaw/shapeshifter-self-patched-solo-granulator It's not the most musical example, but I think it shows off just how flexible this module can be! The basics patch is to set the Shapeshifter into Perc. Mode, and plug the Pulse output into the Sync input. Again, nothing else was used in this patch. ——————— Also, when I go to the MODA menu, I am confused by the difference between Phase 1 and Phase 2, Combo 1 and Combo 2, and Shape 1 and Shape 2. If, as the manual states, Phase 1 has the MODA signal modulate the phase of OSC 2, then does Phase 2 have the MODA signal modulate the phase of OSC 1? Or am I completely misreading this? Make sure you download the latest manual revision. It will explain the differences in MOD A destinations. Targeting Shape 1 or 2 basically turns the Shapeshifter into a Mega-Wave, although it will only use audio-rate signals, whereas the MW can process slow CVs (The Shapeshifter, meanwhile, has a much higher bit-rate...).  ——————— Daisuk wrote: Is there any trick to get oscillator 2 to behave more like oscillator 1 soundwise? If I unsync the two, set combo mode to osc 1, mult on both oscillators to 1, oscillator 2 still sounds far more gnarly and noisy than oscillator 1.    Is that just the design of the Shapeshifter, or have I missed some setting? I love it either way, but just found the gnarly output of oscillator 2 to be a bit odd. I don't know if this answers your question, but I did notice that if you have nothing plugged into the MODA jack but the MODA input attenuator is up (anything past CCW), then there will be distortion in OUT 1 (not sure about OUT 2). You might try putting the MODA input attenuator to full CCW and see how that changes the OUT 1 and OUT 2 signals. ——————— The reg Shapeshifter Thread: ——————— Something to try in regards to SYNC: jjclark wrote: Denis Goekdag wrote:   The only thing I really do want to re-iterate is that it would REALLY open up the potential of the module to have a sync mode that also switches the oscillator to one-shot (so hard-sync + oneshot). With all the cool waves and their interpolation in the SS, it would really allow for excellent sound-sculpting if there was such a "clean" sync mode! I've just been doing this trick again with my CBII and it's a world of difference.  This should be do-able, but you would have to drop something, perhaps the 2=1 sync mode. I will try to work it out and send you a config file if successful. It's not something that I will make as a general update, at least not right away. i was always wondering how/why people used the one shot mode on the cb   do you set it to oneshot and hit the sync with a square wave ? ....how does that differ from hard sync ?  grex wrote: i was always wondering how/why people used the one shot mode on the cb do you set it to oneshot and hit the sync with a square wave ? ....how does that differ from hard sync ? As Denis said, a good trick is to set the CB to oneshot mode and turn int.sync on. Then set osc2 pitch to be more than osc1's pitch. What happens then is that osc2 when synced will go through a complete cycle and stop and wait for the next sync pulse. If osc2 frequency is more than osc1 (actually more than 1/2 osc1's frequency) it will always go through a complete cycle. The advantage of this is that osc2 will always start from where it left off, so there will not be any glitch when the sync occurs. Thus the sound is smooth and not harsh like you usually get with hard sync. The other advantage is that by changing the pitch of osc2 (keeping it above 1/2 osc1) you get formants appearing. For this you want the osc2 pitch to be decoupled from osc1, so plug something into the ratio input.  The other main application of the one-shot mode is to make voltage controlled envelopes. For something similar to the "one-shot" mode being discussed, I think you could send something like a triggered envelope into the Mod A "Shape in" mode. This is a favorite trick of mine with the Megawave to get some interesting CV shapes.  My only complaint is what Denis Goekdag mentioned, that the shape input doesn't handle fast modulation (including envelopes) gracefully. I'm assuming this is a hardware/design limitation, and considering the wide range of timbres available, I don't think I'll miss this capability. tonepanic wrote: For something similar to the "one-shot" mode being discussed, I think you could send something like a triggered envelope into the Mod A "Shape in" mode. This is a favorite trick of mine with the Megawave to get some interesting CV shapes For making envelopes etc yes, that would work. For clean sync sounds not.  Edit: That said, using OSC 1 as Shape source via Mod A and listening to OSC 2 is a *goldmine*. Insane.  Edit #2: I stand corrected. Some aspects of the one-shot sync can indeed be replicated using Mod A to modulate Shape, then triggering a super-short envelope at audio rate and using that to drive OSC 2. It's obviously nowhere near as precise as a one-shot sync, but it does produce very nice results :-) ——————— Denis Goekdag wrote: OK, I now have one unit running up-to-date firmware. Seems that this issue is now significantly improved. The sound will still "break" when modulating faster than say 40-50Hz (and I'd say this is normal unless you resolve the crossfade stupidly high in DSP), but the crackling I was hearing earlier even at lower rates is completely gone. Any wave bank that has smooth transitions when using the knob to interpolate through the waves now also sounds smooth when modulated. And I'd go ahead and say that if you need faster modulation than that, using the Mod A to manually scan the wave bank is the way to go.  Interesting. A lot of the wavetables sound smooth when I turn the knob slowly, but I can get the same artifacts if I turn the knob too fast. I'm assuming it's just a problem with the interpolation not being able to keep up, maybe due to more drastic changes in the waves throughout the table? I'm getting artifacts even with relatively slow LFOs (much slower than 40-50Hz). It also seems worse (more noticeable?) at lower pitches. But there are some wavetables which are much smoother than others.  ——————— Royalston wrote: Sounds great in chord mode...anyone got any tips for getting non ear-killing noise out of the different combo modes (other than 'osc1') and using the fm? I haven't worked out how to get clean signals out of OSC 2 yet either.... Those other modes become very interesting when you set both VCOs to LFO mode.  I also like using Ring mode and selecting waveforms from OSC2 (in LFO mode) that make envelope shapes. You get cool amplitude modulation that sounds like a looping envelope. If you also have Percussive mode active then you essentially have two forms of amplitude control simultaneously. ——————— cephalopod wrote: One thing I noticed that doesnt seem right is that in chord mode the output gets distorted whenever I go above 2 voices. Whenever it is turned up to 4 or 8, it sound almost like the increase in volume from the additional voices is overdriving the output. Has anyone else experienced this? Chord mode is always 8 voices, there is no control over the number of waves mixed together.  Perhaps you mean the "Multi" setting? The sound will always be rather harsh in Multi modes of 4 and 8 because of the changes in the waveshape.  You should also check the setting of the "odrv" parameter. Reduce it to zero to minimize the saturation effect. ——————— I have been caught in horrible noise patches and been unable to get out (had to turn it off and on again). I think this is because some knobs have soft pickup so you have to move past the spot on the knob where the preset setting is at before it does anything. So many factors modify the sound that finding why your sound sounds like white noise is not always easy.  ...  Chord mode is cool. I wish it could output in stereo. Noisy patches are usually due to (a combination of) having FM turned up, Multi set to something other than 1, combination modes other than osc1 or ring.  If you want to quickly get to a peaceful sound you can load preset #13.  Note also that soft pickup only occurs while in Preset Mode (red LED on). If you leave Preset Mode the values will jump to the current knob settings.  As for stereo, a quick hack is to set osc2 to sine and lfo, then turn up the MODA attenuator. This will cause out1 to phase modulate osc2 giving a nice stereo sound. ——————— Also cool for going stereo when doing internal FM based sounds: use OSC 2 to modulate a panner with OSC 1 as input. Audio-rate pan at harmonic intervals sounds really good, especially when using sine waves. ——————— Also of note: TILT stays active when switching to CHORD mode, even though pressing the button brings up the OVERDRIVE display. Very cool!  I think the reason that MULTI modes exhibit some HF noise is that if the phase accumulator stays at the same frequency but its output is multiplied to index a larger section of the wave memory, you get steps in the output wave; as for example at MULT == 2 only every second value in the table will be read. Just guessing, but that would kinda make sense. To avoid that, the phase accumulator would actually have to run at twice, 4 or 8 times the time resolution and just count to higher values. Which would be awesome, of course...*grin*  Pretty easy to get rid of that noise with a filter though, and it's only really noticeable when using darker waveforms. ——————— Funky40 wrote: after having 3 little check in sessions with my SS i have some things that is unclear to me: (No external CV sources patched)  1. "Int Sync LED" goes sometimes on alone, even after i turned it off.  then depending on some knob turns, just right now "IntFM" for example, it goes off again....... ??  i´d expect it to stay off after i turned it off. whats the cause or logic here ?  2. Main Coarse Tune. It seems not low enough on fully CCW. ( on my own sound )  .......But just right now i loaded a preset and it is as low as i like to have it go.  Is this a question of the loeaded wavetable ? What parameters affect this ?  3. I have "IntFM" closed, but when i turn "Ratio" the sound changes, on "Out1".  How can Ratio affect the sound when IntFM is not applied ??  I just recognized here too, its depending on preset.  Whats the parameters to check ?  4. When i like to sync ...lets say a Dixi from the SS, can i do this with the pulse out, is the +01 setting doing it ?  I´m confused here.  1. / 2. / etc. are just to allow quick answer  i find the store procedure counter intuitiv, .......especially at first.  to have to confirm while storing, but not when loading is IMHO inconsistent.  Confirm "screens" are a help at first to prevent from mistakes but later only PITA.  in the computer, within other synths, and even more here in modular. /1/ - It shouldn't do this at all. Are you running in preset mode when this happens?  /2/ - How low is low for you? With 0V plugged into the pitch input and the coarse knob fully ccw I get about 6 Hz on my unit. If you want lower than that you will have to input a negative voltage or use LFO mode.  If you have multi set to 2,4 or 8, the pitch may appear to be lower. This might be why you hear different pitches from different presets. Also, the ratio setting might be different in different presets.  /3/ Remember that out1 is not necessarily the output of oscillator 1. It is coming from the nonlinear combiner, which combines osc1 and osc2. If the combo mode is set to osc1 then you should just hear osc1, but if it is set to anything else then out1 will depend on osc2 as well, and so the ratio setting will affect the sound. Some presets set the combo mode to other than osc1, so these will give out1 sounds that are affected by the ratio setting.  /4/ Yes, you can use the PULSE output to sync other oscillators. You could also use the out1, out2 or fold outputs to sync as well. The pulse mode (+o1 etc) determines how the pulse output is computed, and will therefore affect whatever it is you are syncing.  I find the store procedure intuitive enough. We didn't put a confirm on loading because it seemed to be an unnecessary step which would just slow things down. It was felt that a confirm was needed for storing, to avoid accidentally overwriting a preset slot. For loading, nothing in the memory is being changed, and even if you loaded the wrong preset you can always just reload another one. ——————— There are 3 different conditions to note:  - if a cable connected to 0V is plugged in to PITCH1, with the Coarse and Fine knobs fill CCW, the osc1 frequency is 5.8 Hz.  - if no cable is plugged in to PITCH, the osc1 frequency is 7.14 Hz (a slight increase over the 0V frequency)  - if a cable is plugged into PITCH1 but not connected to anything, the osc1 frequency is 26 Hz (about 2 octaves above the 0V frequency).  The reason for this is that the PITCH1 input is normalled to ground, but through a resistor. The resistor is there to minimize the momentary short circuit that occurs when plugging in a cable. The input then goes to a voltage divider and then to a buffer opamp then to the ADC.  When nothing is plugged in, the normalling brings the effective input voltage close to zero, but a little above. Hence the pitch rises a little bit when unplugging a cable that was connected to zero. If you leave the cable plugged in, but disconnect it from the other end (i.e. leave it floating) the pitch will jump by a little more than 2 octaves.  So, if you want to get the lowest range in frequency, just don't plug anything into PITCH1. ——————— Funky40 wrote: next thing i don´t understand and can´t see the logic:  i loaded a preset, assoon i left "presetmode" by pressing the Encoder the sound changed.  not touching anything else  now same, this very same preset was then altered by me and safed as preset3.  just turned my modular on, loading my preset3, leaving presetmode, the sound changes.  ok, now i turned every knob from CW to CCW, stored the preset again, now when loading th sound remains same when leaving preset mode.  still, its not exactly clear to me, even if a preset has morph enabled ( not exactly clear at this moment about that )  is has to be with morph settings right ? Hi Funky, there is a logic to this. When you are in preset mode, after loading a preset the controls are in soft takeover. This means that the values for those controls won't change until you turn the knob past the preset's setting. So when you are in preset mode, the sound you hear when loading a preset is the sound you heard when you saved the preset.  But! When you leave preset mode, the controls change to hard takeover. This means that the values for those controls jump immediately to the panel settings. If these are different than the preset values then the sound will change.  Why did we do it this way? After all, isn't it bad that the preset sound gets changed once you leave preset mode? Maybe. But consider the alternative - that we leave soft-takeover active when leaving preset mode. This means that turning the controls won't do anything until they are moved past the preset settings. This gives the impression that the front panel has been deactivated if you didn't understand what was happening.  We decided, after testing both approaches, that having soft-takeover in preset mode is OK, since people treat preset mode as a special mode, and pay more attention to details such as having to move the controls a bit to change the settings, but that outside of preset mode the soft-takeover felt like the module was malfunctioning and inactive. So we made hard-takeover outside of preset mode.  The moral of this story is that if you want to hear the preset sound as it was when you saved it, you should stay in preset mode.  ——————— LeFreq wrote: I'm thinking that there is yet another function of the Shapeshifter I need to learn. It seems (and reads in the description) that he's scanning multiple banks in preset mode w/ a randomize feature.  You can step between presets in random order, and each preset can have a different bank selection. ——————— tiny333 wrote: Few questions  If i press comb mode twice it bring the mode up and a number  ? Whats that all about then and can i change that number ?  I cant pass audio thro the shifter like a piston Honda can i ?  Or am i missing something ?  Cheers me dears  And the vocoder is just a carrier for the main sound ?  I cant actually use it as a vocoder ? Or can i ?  When you press combo twice (or any of the other right hand buttons), this adds that parameter to the MODB bus. This means that the parameter is now being modulated by the MODB control and input, allowing voltage control. The left numbers you see are the settings made with the rotary encoder, while the right numbers are the offsets to this setting made by the MODB modulation.  Yes, you can pass audio through the shapeshifter, using the MODA input.  You CAN actually use it as a vocoder. You can!  All this is in the manual... ——————— To reiterate what Danjel said, this is not a crash. The module is not based on a microprocessor, so there is no software to "crash". It is a collection of hardware submodules, like counters, adders, etc, implemented in a gate-array.  The "Easter Egg" occurs when you get all of the LEDs to light up by pressing the respective buttons. In the original firmware release of the module if you revealed the Easter Egg while in Preset Mode, the settings would be randomized. This is a (undocumented) feature, not a bug, as it gives a way to do patch randomization. waveglider wrote: I'm curious why you removed the randomization feature, sometimes those modes are great for instant inspiration. Its still there, the only change was that it only randomizes when you enter the preset mode while the "Easter egg" is being displayed. ——————— A couple of points:  The FM1 input is AC-coupled, so for low frequency inputs this actually phase modulates OSC1 (because frequency is the time derivative of phase, and the highpass filter on the input looks like a differentiator at low frequencies).  Even at high frequencies FM and PM have similar effects (frequency modulating with a sine wave is equivalent to phase modulating with a cosine wave). The MODA input is also AC-coupled, so using it to phase modulate OSC1 would not give results much different than frequency modulating it, which you can do already via the FM1 input.  Phase modulation is most interesting when the modulator is DC-coupled. In particular, it is interesting when you do self-phase modulation. In fact, I thought this is such a useful and interesting effect that I included it already in the Shapeshifter. It is the TILT input, which is a controlled-level phase modulation of osc1 with itself. In my mind this is much more useful and rewarding to explore than using OSC2 to phase modulate OSC1 over and above what is gained by the currently available internal FM.  People should spend some time exploring the TILT function. Putting an envelope into the MODB input tied to TILT gives nice results.  In fact, the whole MODB capability of the Shapeshifter is the key (in my mind) to unleashing the performance aspect of the module. I like to keep my PLANAR joystick module patched in semi-permanently to the MODB input. Using it to control the PRESET Step is an awesome experience and you will quickly realize what an expressive beast this module can become.  And yes, as Denis mentioned, it is straightforward to update the FPGA configuration EPROM. I will be posting a HOW-TO for this sometime in the not too distant future. It is also possible, although less straightforward, to load your own waves into the wavetable, and I will explain how to do this in the upcoming HOW-TO.  [edited to change MODA to MODB] ——————— kisielk wrote: CV control over chord type would be awesome, then you could actually sequence chords. I haven't got my SS yet, but that is one thing I was hoping to be able to do. You can sequence chords by using the step preset mode, using the MODB input to change the presets. It takes a bit of work to setup the presets, but you can do it. ——————— adh82 wrote: I was hoping that too! Best part of the braids chord mode.  I know you can save chords as presets and step through them which is great but for more immediacy and experimentation cv control over chord type would be ideal. 8_) One of the Step Modes (called "ModB") will allow you to CV select a preset (quantized to the sync pulse input). You could store a sub set of chords into the presets and then select them this way.  If you had CV control over all the chords you would be selecting from 64 different chords which might be difficult for a user to dial in (too big of a range).  ——————— jonkull wrote: A couple of things...  1. Is there a list of which parameters work in morph mode? For example morphing from chord mode on to chord mode off or morphing between different wave bank settings.  Which leads to my second question...  2. Has anyone noticed morphing behaving more like an on/off switch where the change between panel and preset settings is very abrupt and doesn't actually morph? Morphing isn't always smooth and I'm trying to figure out if it's due to certain parameters not being morphable or something else (user error for example). All parameters work in morph mode, except for INT. SYNC and PERC. MODE. These follow the panel setting at all time.  The apparent lack of smoothness in some cases is due to the way the parameters are changing as you change the morph. In all cases it is a smooth morph, but the sonic effect can be quite abrupt as you go from one value to another. Remember that it is a parameter morph, and not a cross-fading between two different sounds. ——————— 1) Out 1 is normalized to the input of the folder as well as to the Mod A input, so if you patch nothing else that connection is used. Override by inserting a patch cable.  2) any input anywhere in the modular. They're regular audio outs.  3) perc mode applies to output 1 only. What the output carries is defined by the combo setting. the perc envelope is IIRC applied to a) Int FM amt (starts at set value and goes to zero) and output 1 level.  I think all of this is in the manual BTW...  ———————   I did the update yesterday, It works great.  I will mention one thing.  There is just one thing that should be changed or noted. For the USB-Blaster Driver install. It's located in the C:\altera\13.1\qprogrammer\drivers\usb-blaster folder for version 13.1 The instructions have for installing 13.1 but you the instructions for 12.x for the blaster driver.  It's a pretty simple procedure, just follow the instructions in the manual. ——————— As a followup, in case anyone else has trouble, my issue was solved by using a powered usb hub. Once I used that, the computer recognized it and I was able to install the proper driver and load the firmware.  Actually besides that hiccup, it's pretty easy to do! Have no fear!  Thanks for that suggestion jjclark. ——————— Cata wrote: Question, how does the delay in the Shapeshifter work? I read through the manual and it said that once the sound is sent to the VCA it is then sent to the delay. Does this mean the module actually has a built in VCA that I can control just like a uVCAs VCA or is it something different? If this isn't the case I don't understand how the Delay works since you would just be delaying a straight tone without interruption which seems pointless to me  Yes, there is a VCA (actually a DCA) which is only active in Percussive mode. However, the delay is useful even outside of percussive mode when you are changing various parameters (such as pitch, FM, etc) which causes changes in the sound which gets delayed. Also, when the delay time is very short you get a comb filtering effect. You can also change the delay time under voltage control (through either PItch2 or Ratio) which will change the sound. ——————— Madchiller wrote: So since I updated to 1.03. When I turn up the mod a attenuator past about 10 o'clock I get what sounds like white noise. Nothing is plugged into the mod a input jack. Mod A is set to phase mod. Is this normal?  I checked Control's v1.02 shapeshifter today and it did not do this.  I saw the update note talked of a 4fold increase for mod a into phase mod. Maybe this is the cause? Is the mod a input normalized to something?  Anyone else confirm or deny?  Thanks in advance! Yes, this is due to the increased sensitivity of the phase modulations. The MODA is normalized to Out1, so when there is nothing plugged into MODA, OUt1 will be doing the phase modulation (or whatever). When you are phase modulating OSC1 then you have self-modulation and chaos results when the mod attenuator is high enough. This is a feature! To avoid the chaotic noise just keep the attenuator turned down. ——————— Note that you do not necessarily need to enter 0volts and 1volts. You just need to enter two voltages that are 1volt apart. So don't worry if your CV is giving you 0.1 and 1.1 volts. It is the range that is important. ——————— cannonball swandive wrote: Just got a shapeshifter. Forgive me if this is posted somewhere obvious but I couldn't seem to find it in search. Is independant control over osc 1 and osc2 possible? I got osc 1 down and it is clear how but I'm confused as to how I do the same with osc 2. I'm trying to strip everything away and start from the ground up in learing the ins and outs of the module. The manual didn't seem to clearly state this. Guess I'm just used to the DPO. Yes, it's possible. A couple things: 1. the Pitch 1 input is normalized to the Pitch 2 input, so you'll need a separate input to Pitch 2 (or plug in a disconnected patch cable if you just want a drone) and 2. the Ratio control is really the osc 2 pitch control, and will be independent as long as you don't have the Quant. button set.  I'm assuming you're referring to pitch when you say "control" here, but there are obviously separate shape controls for each osc on the panel. Some of the other functions involve both oscs (like mod A and combo mode), so it doesn't make sense to have separate controls for each osc. Hit the Wave Bank multiple times to switch between osc1 and osc2 wave selection. ——————— Hanz wrote: I was playing around with my SS (v1.04) today - trying to wrap my head around the modulation structure...  -Contrary to the manual description, MOD B input does not flash the PEAK/LFO LED upon clipping, like the MOD A and FM1 inputs would do with exactly the same voltage input. Perhaps a small software bug?  -When modulating Combo Mode (using MOD B input and/or the modulation offset knob) - what does this actually do?  Logically thinking, I figured it would be something like dry/wet ratio (for example when applying ringmod) or amplitude modulation (when using OSC1 straight through) but that didn't turn out to be the case.  As far as I could find, the manual does not explain.  With OSC1 'straight through' mode, for example, I'm getting something that looks like a strong wavewrapper at low input / turning into a 'bit destroyer' later. Something similar with the Ringmod combo mode.  Anybody understand this better? The Peak/LFO1 led only indicates clipping for the MODA and FM1 inputs, as these are the audio rate inputs. I couldn't see in the manual where it says that it should also indicate clipping for MODB (and I wrote the manual!). On page 3, bullet item (13) states that the led indicates clipping just for MODA and FM1.  As for the MODB modulation of the COMBO mode, think of the COMBO mode as a parameter that ranges from OSC1 at one extreme through RING, MIN, etc up to gLcH at the other extreme. These settings are blended together as you move through the list. The purpose of this is to allow the MODB to select which of the COMBO modes you want to apply. So, for example (depending on the attenuator setting of course), 0 volts would give you RING, 3volts gives you PONG and 5volts gives you gLcH. MODB actually acts as an offset to the value you dial in with the rotary encoder. In this way you can use a bipolar LFO for example to sweep around one of the settings, such as MIN.  So its not a wet-dry adjustment, but a scanning selection of the 8 possible combo modes.  Put a square wave into MODB, set the combo mode to OSC1 and set the attenuator so that on each high level of the input square wave the combo mode gets set to RING, and use one of the LFO wavesets for OSC2. Also connect the squarewave to the SYNC input. Dubstep happens. ——————— Hanz wrote: I haven't really been able to figure out any patches where MULTI really 'works' for me (looking for melodic use rather than glitchy sounds).  Especially MULTI=8 tends to be harsh-sounding.  Was wondering, are there any waves where higher MULTI settings come recommended / intended? I somewhat naively thought that using them on the 'instruments' banks (Sax, Piano, Guitar, Flute etc.) would give more 'natural' results but far from it...   Anyway, one thing that I would like to do is apply the LFO sequence (especially on MULTI) as a 'one shot' thing that stops / goes to 0v after completing the wave.  Would there be a way to achieve that, perhaps by using Sync options in a creative way? There aren't any wavesets specially designed to work with the Multi. The LFO waves and the TwoTone waves are probably the best for smoother sounds. The MULTI is there to give some different textures. They can also be fun in LFO mode.  You can use OSC2 as a one-shot, by setting the sync mode to Hold/1-shot. ——————— Clicking the Int. Sync button next to the Int FM potentiometer puts both OSC in sync also helps reduce the chaos. ——————— Regarding the INT. FM depth, this was set to 'aggressive' since the Cyclebox was criticized by many as being too tame in its FM modulation. People often wanted to use an envelope feeding the INT FM input, and the Cyclebox did not sound as bright as some FM synths when doing this. The Shapeshifter sounds much brighter when pinged with an envelope.  The external FM1 modulation curve is different than that for the INT FM. The FM1 is slightly less aggressive, so you can try that (plug a cable from OUT2 into FM1).  It is would be easy enough to tweak the code to reduce the aggressiveness, but I don't want to do this. I could add another mode that would switch between tame and aggressive, but - you know - more menu diving... ——————— The shapeshifter internal FM is linear.  As such, the only free parameter is the modulation index.  The shapeshifter has a higher maximum modulation index than the cyclebox.  A higher modulation level means higher harmonics. That is why the shapeshifter can sound brighter than the cyclebox. It has nothing to do with having the 'right' envelope.  ——————— Royalston wrote: Can someone explain modB destinations and modulation... if I have decay set to 25 (and it sounds good) and press it twice I get the decay mod b modulation range. Its always then 25-99 ---the upper limit, rather than 0-25, and it starts modulating straight away like mad. I probably dont understand it right and need to read the manual again, but I've found it unuseable/ unintuitive so far.  Again mod B leads very quickly to noise in most of my patches and it would be great to have some thing far more subtle It will go lower than the setting if you use a negative voltage input to the mod b. Sounds like your modulating signal is unipolar (positive only).  It is pretty simple, actually. As someone else mentioned, the mod B input just adds (or subtracts) an offset to the value set by the encoder. So in your example, if decay is set to 25 by the encoder, then a positive input to mod b will increase the decay value (from 25 up to a max of 99) while a negative input will decrease the decay value (from 25 down to a minimum of 0).  You can use the attenuator on the mod b input to tame the modulation if it is too 'mad' for you. That's what it is there for. ——————— sushiluv wrote: is it possible to use the ringmod combine mode also with an external signal?  don´t have a shapeshifter yet, but i´m curious because i´m on the fence to buy one  Yes, it is possible. Please get off the fence.  The audio rate external input MODA can be used for 8 different purposes:  - modulate the Phase of oscillator 1  - modulate the Phase of oscillator 2  - input to Waveshaper 1 (takes the place of the output of oscillator 1)  - input to Waveshaper 2 (takes the place of the output of oscillator 2)  - input to the Combo circuit, taking the place of oscillator 1  - input to the Combo circuit, taking the place of oscillator 2  - Vocoder Carrier  - Vocoder Modulator  The audio rate external input FM1 is hardwired to modulate the frequency of oscillator 1 (through-zero).  Note that both MODA and FM1 are AC-coupled so slow LFO waveforms don't work so well. But audio rate signals work great! ——————— Modulating the phase of Osc One with a bass guitar whilst changing waveshapes is endless fun ——————— Vocoder uses mod A input. You need a modulator source, I.e. Microphone. You will need to take the microphone to a preamp and then into mod A input. Select mod A as voc modulator. Enjoy.  ——————— realshafer wrote: Been digging into SS a little more and have some questions:  - Can Mod A be routed to more than one destination? Would be nice to modulate say phase and shape at the same time. Speaking of...  - Mod A is capable of modulating the rate at which the wavetable is read and this is called "Shape" in the Mod A menu. However, this seems confusing as there are two front panel controls for modulating which wavetable is being read labeled "Shape. Does anyone else find this confusing?  - Similarly to the first question, can Mod B be routed to multiple destinations?  The MODA input can only be routed to one destination. However, of course you can always use a multiple to duplicate whatever is plugged into the MODA jack and connect it to one of the other inputs (including the MODB) if you want to modulate more than one thing with the same signal. - I should probably rename the SHAPE 1/2 destinations in the MODA destination list. It doesn't modulate the shape (as in selecting which wave to use as the shape). Instead, it passes the MODA signal as the address of the oscillator's wave shaper instead of the oscillator phase. This gives a way to waveshape external signals with whatever wave shape is currently selected. This could then be modulated with an external signal connected to the SHAPE input, giving a very dynamic waveshaping effect.  The MODB input can be routed to multiple locations (all of the parameters listed on the buttons to the right of the rotary encoder - COMBO MODE, TILT, DRIVE, DELAY, DECAY, DETUNE, as well as select the preset in preset step mode). ——————— maudibe wrote: 3/ OK, so did my first save... after I had spent an hour creating a monster sequencer voice. Re-loaded it (user slot one) and was rather disappointed to see that lots of stuff was *not saved*. So yes, the front panel core knob positions had been saved, but the perc. mode, chord switch, quant and internal sync had not been saved, or the delay and drive settings.  Have I got something wrong?  The presets DO (should) save the perc. mode, chord mode, quant, drive and delay settings. Perhaps you actually didn't save? I don't know what could cause it to act as you said.  But the internal sync setting is NOT saved in a preset. We did it this way to avoid problems when preset stepping, which uses the sync input to step presets. If internal sync was on then stepping would happen at audio rates. That could be cool, but not all the time. You can get a similar effect if you really wanted by connecting out1 (or the pulse output) to the sync input.  As for having a switch to bypass oscillator 2 I am not sure what you mean by this. You can already get rid of any connections or modulations from out2 back to osc1. Do you mean can you just mute OUT2? Most VCOs don't have such a thing (are there any that do).  Actually, now that I think of it, there is a way to do this muting - connect OUT2 to the FOLD IN, then use the FOLD CV to do the muting. When it is zero the output of the folder will be muted. Then you can bring up the FOLD CV just a bit, since there is a range where the signal is not distorted before all the folding begins ——————— shootingtigers wrote: I got one a couple of days ago and it's great but I have a really stupid question, is there a way to get both oscillators to come out a single output so I don't have to use an external mixer?  You need to check out the combo settings, which will give you a mix of both oscillators. ——————— mqtthiqs wrote: - Similarly for MOD A: why not have a simple VCA option? Each MOD A sample would be multiplied by the oscillator sample. I know you can set it to Combo 1 and Combo to Ring, but that seems so convoluted for such a simple function!  We could add that in a future revision. Thank you for the idea. Although, as you say, it can already be implemented.  On second thought keep in mind that MODA is AC-coupled, so you wouldn't be able to use it with an envelope generator as a VCA. So it would really be acting as a ring modulator, which as you noted is already implemented. So you wouldn't gain much.  If you want to implement a proper VCA you can already do this with the FOLDER. If you keep the FOLD CV attenuator down low so that the folder is always in its linear range then it makes for a nice dc-coupled VCA on whatever is fed into the FOLD input. It also has the nice feature that if you turn the attenuator up a bit the FOLDER starts to saturate a bit giving a nice slight high-end distortion which can add a bit of crunch to whatever you are feeding in to it. ——————— Battagiovi wrote: Guys, i'm wondering if i can sync osc2 (lfo mode) to external clock. I can't figure out how...   This would be the patch:  Osc1 to Vca in  Osc2 (lfo) to vca cv in  Clock to osc2 to sync with song's tempo Use the SYNC input for this. It will sync both osc1 and 2, however, which might be a problem for you. Try it! Press the SYNC button (2nd button from the top on the left) and select Hard Sync.  You can also use the PERC mode to do your patch. The SYNC input will then trigger the note as well as SYNC the oscillators. Then, use the RING combo mode. This will multiply the osc2 (lfo) wave by the osc1 wave, just as you would do with your VCA. The only downside is you are stuck with the limited envelopes provided in PERC mode. ——————— Pampalini wrote: Shapeshifter, fantastic as it is, doesn't offer PWM, or at least I haven't seen anything like t on mine. Morphing through the BiPuls bank is the closest I've gotten, and it's still quite far from PWM. Have you guys found a way around it, or do you rely on other oscillators for PWM? One way to get PWM on the Shapeshifter:  Set the Pulse Mode to XOR, set SYNC Mode to HardSync and turn INT SYNC on. Take the output from the Pulse output. The RATIO knob/input acts as a standard PWM control over the lower 1/4 of the input range (from 50% to 0% PW), with higher values giving more of a sync sound.  There are variations on this idea, and there are others ways to get PWM-like sounds. Copy blog RSS feed url here

Here is a guide I've written for the FH-1.  It is mostly focused around generating the extremely powerful custom scripts with some examples for our pretty elaborate dual Octatrack rig.  Click here for the document!:

We have made a nice reference for the lovely Elektron Octatrack.  View it by clicking here: Copy blog RSS feed url here

I've tried many quantisers over the years and it's the lil bit loosey-goosey, but still awesome A-156 that always tickled my fancy.  The trigger in/out is essential and the guy just works without crazy menu diving or reading manuals.  The scale limitations may be too limited for some but I find with a transposition input, the quantizer is a perfect fit for any mood I'd like to set within this tt stuff we'd likely be doing with a quantiser anyways. One thing that always bothered me was that 1) I have to suck up an offset voltage from somewhere in order to set the transposition (and correspondingly, the mode) and 2) there is plenty of blank panel space to put it.  So I put a normalled transposition control right on the front panel! The Mod A 27k resistor, 10kB pot, and a new hole, and you're done.  If you want more range, decrease the value of the 27k resistor.  I know what range I like so this works for me   I chose a range just slightly higher than 1 octave at 14st.  This allows me to get decent settability with the control but still be able to get my m2 an octave away if need be.  If you start setting the range really large (several octaves), a single turn pot gets really, really touchy.   The 27k is connected to the OB 5V regulator to have some isolation from the rail voltages.  Since it's transpose and this is already quantised, very roughly, the error here isn't too much a concern. The wiper of the pot is connected to the already hangin' free switch pin of the transpose jack. Done! Video In-Action: Make sure you set the jumper to the upper-most position to have the same transpose and scales applied to both sections of the 156. Copy blog RSS feed url here