The Allegorist - The Third Album: Hybrid Dimension II

I always look forward to releases by The Allegorist, and Anna's third album 'Hybrid Dimension II' continues the 'slow building' style of her previous releases, but this time the production value in the sophisticated mix of ambient, classical and electronica has jumped up several notches. This is all the more remarkable because when I first heard her work at an Ableton Loop in Berlin a few years ago, it was already stand-out, drop-dead, slack-jaw perfection. You could kind of tell it was exceptionally good, because Mandy Parnell simply didn't know what to say for several minutes after hearing it. 

The cover of 'Hybrid Dimension II'

There's a huge amount of finesse exhibited in this third album. The opening track has a drone that is gradually replaced by vocals in the invented 'Mondoneoh' language, and it's like an overture to set the scene for the whole album. But the vocals don't just fade in as you might expect - instead the drone's timbre gradually morphs into the vocals, rather like a curtain drawing open and letting higher and higher frequencies out. And this masterful slow building of almost visual soundscapes just continues in the rest of the album - this isn't your standard 'ambient' washes merely produced by using lots of echo and reverb. It is more like one of those 'continuous zoom' videos, where you keep diving deeper and deeper through breath-taking vista after glorious vista. And the blending is seamless, all the way down...

There's a huge variety of vocal, sample and synth timbres, and they aren't static. They wash over each other, like waves crashing on a beach, evolving and slowly changing... It is stories told in music on a grand scale. There's a host of superb pad sounds to die for, and the changes just keep going - just as you start thinking: 'that's a nice pad', then it has already changed into something else equally gorgeous but different, and it just keeps going. There are multiple techniques that are used to transform from one sound to another, and just as you think you have one figured out, along comes another one that does it differently. A sound designer's or synthesist's or producer's cornucopia of riches... Listen and weep.

Amazing. Immersive and confident. I wish I could make electronic music half this good.

I struggle for analogies, but if you go back over 40 years to Jean-Michel Jarre's seminal 'Oxygene', and replace the organ sounds processed by twin EHX 'Small Stone' phasers and VCS3 doodlings with modern sampling, synthesis and vocal processing and production techniques, and you change the genre to 'Ambient crossed with Classical crossed with Electronica', then you start to get a feel for just how much this connects to the very deepest part of me. Or maybe a more modern angle would be The Flashbulb... 

Recommended. Oh wow, is this recommended!

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Project Proposal seeks Student... (Keyboard Noises)

You know when you have a good idea, but don't have the time or resources to actually do it? Well, here's an example... Cue: picture...

Photo by Kelly Sikkema on Unsplash

You have probably noticed that just about every electronic musical instrument makes a different noise when you press and release the keys on the keyboard. Some are quiet, and you barely notice them. Some are noisy and intrusive in some circumstances. Some make springy sounds. Some make a sort of clacking sound. Some are so jarring that you have to use headphones or turn the amplifier up. Some are an intrinsic part of the instrument (acoustic piano, clavinet, harpsichord...). Some defy description. Everyone has their own preferences for what is an ideal 'noise', or lack of it.

But what is difficult to determine is what an instrument will sound like - in advance. In general, and especially in these 'online purchase' times, it is only when it arrives that you actually get to hear what the keyboard actually sounds like, especially long-term. Even a few minutes in a music shop may not give you a very good idea of how it will sound in your acoustic environment.

So I gathered some keyboard noises from a few instruments: high velocity key-down, and high velocity key release. This is what I got (time waveforms at the top of each set, spectrograms underneath in colour):

SY99, Montage 7, and CLP-930 key noises

These weren't perfect recordings. But what is intriguing are that they key-up is sometimes louder than the key-down (SY99), and that there are wide variations, particularly the Clavinova (echoes from the cabinet, or scrapes from the action?). You will eventually be able to get edited versions of my recordings here.

VI-Control Forum

This topic came up recently on the VI-Control Forum, a place where composers, musicians and technologists with an interest in virtual instruments (Kontakt et al) discuss a variety of topics. In this case it was specific to MIDI Controller keyboards, but the subject of keyboard noise is really a generic one across all electronic musical instruments with keyboards. I proposed that a crowd-sourced approach to gathering the sounds might be a way to get some useful material (by asking VI-Control Forum members to record their keyboards), although I acknowledged that there were some problems to overcome. There wasn't much feedback, and this often happens in forums - they aren't really places for marshalling big campaigns involving lots of people, unless there's a very strong reason that will  motivate people.

The Alternative

Since then, I've been thinking about it a bit more, and I decided that an alternative way of gathering the data could be an interesting experiment. I am sure that it is a good idea, and that the results will be useful, but I don't have the time to do it myself.

So I have written up some more of my thoughts and the background research that I did around the idea, and have produced a rough Project Proposal. That's what follows. My hope is that a university or college student on a music technology course, or maybe an intern at a manufacturer, sees it, and decides to do the research and then publish it. If you know someone who might be interested, then feel free to give them this URL or the project proposal below!

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Project Proposal

Title: Noises in the Keybed.

Author and Licence: This project proposal is written by Martin Russ aka Synthesizerwriter, and this work is licensed under a Creative Commons Attribution-ShareAlike 4.0 International License.

Abstract: The keys on the music keyboards (keybeds) on electronic musical instruments make noises when they are pressed and released. Data on the Sound Pressure Levels (SPL) of these noises, as well as subjective descriptions, and recordings of the sounds that are produced, are not widely available. As a result, it is difficult to make informed decisions when deciding which instrument to use for a particular application. This project aims to provide information on how relevant data can be captured consistently and reliably, and subsequently made available to anyone who would find it useful.

Procedure: There are several stages required to capture relevant data of the noises made when the keys on a music keyboard (on an electronic musical instrument or MIDI Controller keyboard):

1. Preparation of the keyboard, the recording/measuring equipment, and the performer. 

2. Key-down noise (pressing the key on the keybed, from the default 'rest' position, to the full depressed position, where it is stopped by the physical keybed)

3. Key-up noise (releasing the key on the keybed, so that it returns to its default position)

4. Processing of the captured audio files and SPL levels

5. Processing of data from the processed audio files

Stages:

1. Preparation.

The capture can be carried out by individuals acting as part of a crowd-sourced team, where the results are collated by an individual or a team. Alternatively, the capture can be carried out by a single individual or a team.

The keyboard should be placed on a solid surface, ideally one that is free of rattles, creaks and other noises. The 'performer' should checking for any variation of noises across the range of the keybed, and then choose the noisiest key as the one to use for SPL measurement. If the noise is consistent across the keybed, then a key in the middle of the range of the keyboard should be chosen. Whichever key is chosen, it should be marked with a removable sticker, and this will then be the only key used for the capture. 

If a microphone is used then it should be placed 1 meter vertically above the keybed. This makes positioning easier than close mic'ing. Because the source of the sound is typically inside the keybed, then positioning the mic above the keybed (and with the microphone) aimed at the keybed, should capture the sounds produced by the keyboard, and should not interfere with the pressing and releasing of the keys on the keybed by the 'performer'. This positioning of the microphone should be possible with most mic stands.

If an SPL meter is used then it should be placed in the same way as the microphone: 1 meter vertically above the keybed. 

Low-cost SPL Meters. You 'can' get cheap (about 20 USD) SPL meters from Amazon et al.., but their supplied calibration is difficult to determine, there is anecdotal evidence that indicates that measurements may be several dB different relative to a properly calibrated SPL measuring device, and proper calibration devices cost about 6x that (about 120 USD). This approach seems to be a recipe for making the results worthless because of lack of calibration - but may be the only apporach that is feasible when using crowd-sourcing.

Higher-cost SPL Meters. Properly calibrated and calibratable SPL measuring devices will typically cost over 120 USD, and will require the use of a calibrated sound source (about 120 USD). The same SPL measurement device should ideally be used for the whole of a session, and ideally for as many captures as possible. 

Capture measurements should be of SPL and the sound simultaneously, and so a microphone and SPL meter should both be positioned 1 meter vertically above the keybed. They should not touch. One possible alternative to using separate microphone and SPL measuring device (meter) many be a calibrated USB microphone. One example is: https://www.minidsp.com/products/acoustic-measurement/umik-1. These cost about 120 USD, but can be used to capture both the audio and a calibrated level for the audio (it may be possible to also derive the SPL).

The recording environment should be as quiet as is practical. A 30 second recording of the background noise in the recording environment should be made at the beginning and end of each capture session, so that the recording can have noise reduction applied, if this is required. The ambient temperature in the recording environment should also be noted at the beginning and end of each capture session.

The instrument, of which the keyboard is a part, should be set so that it does not make a sound. For electronic musical instruments this generally means that it should not be connected to an amplifer, or the internal loudspeaker amplifier should be set to minimum volume. For other instruments, then damping may need to be applied to any vibrating part that is not directly a component of the key mechanism. 

The performer, whose role is to press the key on the keybed at five second intervals during the capture session, should wear cotton clothing, a sleeveless tee-shirt, no jewelry, no rings, no bracelets, no ear-rings, etc. Ideally, they should wear no metal or other material that could make any extra noise. The only noise that should be recorded is the sound of the key on the keybed. The performer should breathe quietly and moderately during the capture session.

It may also be possible to find data on the SPL of keybed noises on the InterWeb, although the author has not found any such data. A search for this would be good practice before committing to a full project.

 2. Key-down noise 

The performer's task is to press the defined 'middle' key on the keybed, moving it from from the default 'rest' position, to the full depressed position, where it is stopped by the physical keybed. The start of recording should be indicated to the performer, who then waits 10 (ten) seconds, and then presses the key. Once pressed, the performer then waits for 10 (ten) seconds, and then moves to the next stage ('Key-up')

A range of velocities or strengths, of pressing the key should be used. The minimum set should be High (as fast and hard as possible), Low (as slowly and lightly as possible), and Mid (mid-way between the previous values). If MIDI is available, then MIDI velocity can be used to achieve consistency. Max would have the MIDI velocity value of 127, Mid 64 and Low should be below 30, but this depends on depending on the playing ability of the performer. 

Notes should be taken during the capture session as a subjective record of the 'sound' of the key noises. Words used can include: clicks, snaps, clunks, thumps, spring-buzzes, slides, sizzles, etc.

The three MIDI values:127, 64 and 10-30 should capture most of the variation of 'sound vs velocity', and a single key is not onerous to record: less than 30 seconds of WAV, and probably less than a quarter of an hour to accomplish. If it is discovered that there is a particular velocity or strength of pressing that produces a different opr louder noise, then this should be captured and noted.

Three key-down captures should be recorded.

3. Key-up noise 

The performer's task is to release the key on the keybed, so that it returns to its default position. Based on the techniques used to record acoustic pianos, then there are two extremes that should be captured:

a. Starting from the key being held down (the ending position of Stage 2), then the performer moves their finger backwards so that the key is allowed to rise up to its default position on its own. The performer should strive to avoid making any noise as the key is released.

b. Starting from the key being held down (the ending position of Stage 2), then the performer moves their finger upwards and off the key as quickly as possible, allowing the key to rise up to its default position on its own. The performer should strive to avoid making any noise as the key is released.

Once the key has been released, the performer should wait 10 (ten) seconds before the next Stage 1 key-down, or if this is the third capture, then the performer should wait 10 seconds and the capture will stop.

Three key-up captures should be recorded in total per chosen key on the instrument. If one key has been chosen, then three captures. If two keys have been chosen (noisiest and quietest, where there is a noticeable range of noise levels), then two sets of three captures should be recorded in total.

4. Processing of the captured audio files and SPL levels

The recordings of the key-down and key-up events stages (2 and 3) will be in pairs (down then up), and there will be three contiguous sets of the pairs. Each stage should last for ten seconds, with an extra 10 seconds at the start.

The audio files can be trimmed so that there are 10 seconds of near-silence before the first stage 1 capture, and 10 seconds of near-silence after the third stage 2 capture. The audio files should be stored in a linear, uncompressed format. WAV files, either 24 bit or 32 bit floating point, ideally at 48 kHz sampling rate. 44.1 kHz can be used if it consistent across all captures. Mixtures of 44.1 kHz and 48 kHz sampling rates should be avoided if possible.

If an SPL meter is used, then the peak measurement during the three captures should be noted.

Files should be saved with names that include the date, instrument name, a capture session ID, the sample rate, and the bit depth (24 or 32float).

5. Processing of data from the processed audio files

A spreadsheet should be used to hold the metadata noted during the capture session. Columns should include the date, instrument name, a capture session ID, the sample rate, the bit depth (24 or 32float), the measured peak SPL, the name of the performer, the name of the recording engineer or co-ordinator, any notes (quiet, noisy, clicky, thumpy, etc.) and any other comments. 

A cloud-based spreadsheet is recommended. Airtable is a sharable spreadsheet with advanced capabilities that offers a free service for small numbers of rows in the spreadsheet. https://airtable.com/

Stages 4 and 5 are just spreadsheets, and file management, and will appeal to a very particular type of person. 

The results should be shared as widely as possible. The VI-Control Forum is one possible place where a post could be placed.

Conclusions: I hope that this project proposal is useful to someone. Please feel free to share the URL or a printout of it to any researcher, student or intern that might find it useful or inspiring. This work is licensed under a Creative Commons Attribution-ShareAlike 4.0 International License, so that it is available to all under clear usage terms.

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Black Dust From An Old Keyboard Rack...

So, there I was, working away in the studio, when I noticed some black dust on the keyboard of my Yamaha SY99 on my keyboard rack. I looked at the Synthstrom Audible Deluge above it, and moved it a little, and more black dust appeared. What? 

The Black Dust!

The dust is dark black, gritty, and very fine. I have no idea what it has been doing to my synthesizer keybeds or key contacts... I have to say that this isn't a complaint - after over 30 years, then I would expect some deterioration of some components, and everything else about the rack is still perfection! 

After some investigation, it turns out that the rubber pads on the keyboard support brackets on my custom hand-built Ultimate Support Systems rack have perished over the last 30-odd years, and need replacing. (As an illustration of how things have changed over those intervening years: Ultimate now only seem to make those stage-friendly 'Z' and 'X' shaped stands... http://www.ultimatesupport.com/products/keyboard-stands.html )

Just removing the support brackets created dust!

Fixing the problem requires pulling apart a five-tier custom keyboard rack ( the modern equivalent would be something like https://www.jaspers-alu.de/aid-677-KR170-4-150B.html ), removing the keyboard support brackets, removing what's left of the pads, then removing the 30-year-old adhesive (nasty and difficult), then putting on new better pads, putting the keyboards/synths, drum machines, sequencers, mixers and stuff back, then cabling it all up again - not quite a five minute job. 

The rubber pads are now very fragile...

As you can probably sense, it's all a bit fraught here at the moment, and I have no idea how much damage that black dust (gritty, abrasive, horrible stuff) has done to my keyboard contacts and keybeds. Vintage gear, eh? You think that all you have to contend with are LCD backlights failing, or electrolytic capacitors drying out - and then you get awful black dust everywhere. I'm just glad that I have spotted it, that I don't move stuff around on the rack very often, and that I don't have any of those amazingly expensive synths that you see on eBay and Reverb. I'm also guessing that there may be more than a few other synthesiser owners who haven't checked the rubber pads on their more-or-less permanently-installed keyboard racks for quite a while... Now could be a good time to check. It did seem that, as with 19 inch racks, heat may be a major contributor to problems - the worst deterioration did seem to be where the synth/sequencer/drum machine/etc., got warm...

The support bracket, the gripper bracket, the bolt and an Allen key...

Some mechanical details are probably a good idea here - because the modern racks from people like Jaspers are different (Yep, in more than 30 years, designs have changed!). The basic tubing of the rack is black anodised aluminium: 38 mm in diameter.  the gripper clamps/brackets use an Allen key to tighten them in place by gripping the tube, and a pozidrive bolt holds the support bracket onto the gripper clamp/bracket. As you can see, there's a design problem with the gripper clamp/bracket - it is in one piece, and so can only be removed by pulling almost the whole keyboard rack apart (in most cases!). That is a lot of dismantling! A two-piece gripper clamp/bracket would be a much better design...

The black dust removed from a support...

After two days pulling my keyboard rack gradually apart, and then attempting to get the rubber pads off, I now have a much better idea of what is involved, and I also have a lot of black dust... 

Chemical failure...

The adhesive is the worst part - after more than 30 years it is almost part of the metal. I've tried scraping at it, Fairy liquid, various degreasers, orange oil (my secret weapon) and a scary adhesive remover that is covered in warnings and is apparently toxic, flammable, harmful and more. None of them shifted it. As usual, take care with chemicals: read the warnings, do it in a well ventilated place (or outside) if recommended, etc. 

So I stopped the arms race of ever-more-scary chemicals, and went old school. Time. 

After soaking overnight...

Yep, there are very few things that a little time won't degrade - as shown by the rubber pads! So I soaked them overnight in a hot dilute fairy liquid solution. Fairy liquid, or your alternative favourite washing-up liquid, is designed to do degreasing and a few other useful dish-washing tasks, but the important ingredients here are time, and water. Water is a strange liquid. It has all sorts of properties that are unusual, and it turns out that the more you investigate it, the more interesting it becomes. Anyway, the recipe, again, is water, time, and heat to help reactions along a little. Let's pause here overnight, and maybe another overnight...

Scraped off and allowed to dry...

So, when the adhesive remnants have changed colour (paler, less saturated colours, and a milky appearance), then things have probably got a lot easier. After two overnights, then the remnants of the adhesive can be scraped off very easily using a screwdriver or an old credit card (another secret weapon). This leaves just a few bits, which I soaked again overnight. 

Finally! Clean to my satisfaction!

I was now starting to see some progress, and after drying them, then one of the 'not quite as alarming warnings' spot cleaner got me very close to a clean metal surface. And a final wash and dry got me to a final look that I was happy with, and one that was suitable for the new clear rubber stick-on pads that should apparently last for 50 years. 

Just before reassembly and sticking on the new pads...

If anyone had told me that the 25mm square rubber pads on my keyboard stand were going to be such a major problem, I would never have believed them. Amazing. I'm dreading opening up the keyboards to see what I find inside.

Brackets with new rubber pads...

And that was that. Several days of work to remove the gear, remove the brackets, figure out how to remove the remains of the rubber, then how to remove the adhesive, and finally clean the bracket ready for the new self-adhesive pads. Oh, and then put all the gear back, rewire it, etc. In the process, you can't help but reappraise the gear and start thinking about modernising, modding...

What is interesting is that I realised that I also have a 'vintage' keyboard rack' for my 'vintage' gear!

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This tool sucks! (which is good!)

One of the downsides of having vintage gear is that time takes a toll on old electronics. Quality counts, but the ravages of time can affect even the highest quality of components. I have covered fitting replacement displays (LCDs) with backlights previously (TX7, RM-1x), but I've recently been replacing electrolytic capacitors and switches, which are also sometimes the unfortunate casualties of time. Capacitors can dry out because of heat, from CPUs and power supplies, mostly. Switches just have finite numbers of cycles, and the front panel of a synthesizer can require a lot of button-pressing depending on how it is designed - 'Enter' buttons, or cursor buttons, or increment/decrement buttons, or 'Play/Stop' buttons are all candidates for repeated presses over the lifetime of a piece of gear. Some switches fail because of metal fatigue, some because of dust, and some because conductive plastic loses its flexibility over time. The switches that you get on older, vintage equipment from the 1970s and 80s tend to be 'proper' metal contact switches, and so the failure mechanism tends to be just plain old 'wear and tear' - but at least they are individually replaceable. (More modern gear can use conductive rubber switches which are made as sheets, and have to be changed all at once, trying not to get any dust underneath them. And no soldering required, usually!)

Removing thru-hole components from printed circuit boards (PCBs, but not Poly-Chlorinated Biphenols) can be tricky. You want to avoid damaging the pad or tracks, and yet you need to heat up the joint sufficiently to be able to remove as much of the solder as possible - which are a difficult pair of requirements to juggle. The key words here are:

'Removing the solder...' 

Ideally, I would have invested in two soldering stations: one for surface mount components (with a tiny soldering iron bit, a heat gun, and strong magnifiers - something like this or this or this), and maybe one for thru-hole (with built-in suction). But this isn't an ideal world: I used to have an Antex soldering iron back in the 1970s, and I have had a very basic (and a bit industrial) Weller soldering iron for many years, that really needs replacing with a more modern TS100 model. 

But at the moment, I've been struggling with an antiquated classic; the almost-all-metal RS components 544-516 model solder sucker. The sucker itself works perfectly fine, but I have been totally unable to find any replacement plastic nozzles for it - and the RS website, and Farnell, and others. all laboriously lead you down a series of labyrinthine pages that never actually lead to a replacement nozzle. 

So I have the classic problem where the end of the metal rod, sticks out of the nozzle, and reduces the suction enormously - but no obvious way of fixing it... 

I did try one of the 'affordable', part plastic, part metal alternatives from Amazon, but that has a wide nozzle, a small metal rod, and doesn't suck very well. It also has the unobtainium problem with replacement nozzles as well. 

I even reminded myself that I don't like solder wick, although I know some people who like it, and I also know some people who mis-use it to create solid ground links on digital audio boards. 

Finally, I decided to stop messing around and do what I should have done in the first place: buy a decent solder sucker (and defer the surface-mount soldering/heat gun workstation to another time). 

The Engineer package - without the solder sucker!

After some research, I found the Engineer SS-02 from Japan: an all-metal, guard-less solder sucker with a bright red plunger button, and a replaceable silicone plastic nozzle (and a spare length of tubing included) and where additional tubing sections can actually be found and bought on Amazon! 

The included tube (top), plus an extra pack of two tubes

I have to say that the combination of a flexible plastic nozzle and huge amounts of suction, is just wonderful! For removing thru-hole components, then you heat up the joint until the solder goes shiny as it melts. Then you let the molten solder spread to all of the joint (you can see it go reflective and start to move), then position the nozzle as close to the soldering iron tip as possible and press the button on the side. The plunger is pushed out by the internal spring and the red button jumps upwards as air (and solder) is sucked through the nozzle inside the sucker.  

It is an interesting balancing act. You want to heat up all of the solder in the joint so that it can be sucked out, but you do not want to over-heat the pads and the tracks they are connected to. You want to get the solder sucker nozzle as close to the soldering iron tip as you can, and yet you don't want to stress or damage the silicone rubber nozzle by over-heating it - although it does seem to withstand a lot of heat!

Then there's the longer term question of what happens to the solder? It ends up inside the solder sucker, either as tiny round spheres, or as thin streams. When you push the red button to reset the plunger, then the metal rod pushes down through the nozzle and sometimes pushes some parts of those thin streams of solder out. Some solder stays inside the sucker, and so you have to unscrew the end every so often and shake/tap out the bits of solder. The inside of the sucker also tends to get coated with remnants of flux from the joint, and this goes sticky and dark over time. 

You can't see what is happening inside the solder sucker, and so emptying it tends to happen when you can't push the plunger back down until it clicks. The internal entrance to the nozzle tends to get clogged with solder that the rod can't push through, and so then you have to empty the sucker. The more solder you remove, the more often you need to empty it, and so a good solder sucker tends to need emptying more often - which is a sure sign that it is working well!

Ideally, enough of the solder has been removed from the joint, so that the component (capacitor, switch...) is now loose. To help this process, with that third hand that you probably don't have (soldering iron in one hand, solder sucker in the other) you should be applying pressure to the component so that it comes free as soon as there isn't enough solder there to hold it in place. In practice, as soon as the solder sucker has done its suction thing, you drop it, move the iron away, and nudge the component so it comes free (the component is on the other side of the PCB, of course, which just makes it all the more challenging!). 

It's a complex ballet of movements to try and remove the solder and loosen the component, and you really need an extra hand. Trying to co-ordinate two people's hands and associated tools in the tiny space near a solder joint is very difficult, but can work with larger PCBs. Another technique is to use the soldering iron tip to jiggle the component lead as the solder is sucked away, which works better with capacitors than switches, because switches tend to have more solid connections rather than flexible leads. Flat-bladed screwdrivers or tweezers can be used to lever the component as the solder is sucked away, so that any remaining solder can't freeze and lock the component in place again. 

Some people advocate twisting the cans of electrolytic capacitors axially so that the leads break, which makes the remnants easy to remove - sometimes the lead just drops out when you heat the joint and suck the solder away. But this can damage solder resist and pads, and is difficult to do if the capacitors are tightly packed together, which is often the case. 

So the solder sucker is only a temporary home for the solder. Unfortunately, it doesn't magically disappear - the sucker moves it from the joint to inside the sucker, and you then have to empty the sucker. The next step up are solder suckers which have heated tips like a soldering iron, but they also have suction through a hole in the middle of the tip, and the solder ends up in a glass tube. Again, over time, the tube fills with tiny spheres and streams of solder, and needs to be emptied - the glass tube just makes it easier to see when you need to empty it.

In conclusion, the Engineer SS-02 is especially perfect for removing switches and capacitors from PCBs... I may have put off buying a proper suction station for another day...

Name choice

I just have to comment on the choice of brand name by the Japanese manufacturer of this solder sucker. 'Engineer' is a totally brilliant, deceptively descriptive (and tautological), choice for a brand name: these are obviously tools for serious engineers, hence the name. But there's no doubt about what the purpose of the tool is: 'Hobbyist' or 'Bodger' wouldn't be very good choices, whilst 'Professional' and 'Expert' have been so over-used that they have become mostly worthless. 'Serious'? - it just sounds like irony. So 'Engineer' is a near perfect name.

And as a final thought. It turns out that my favourite, 'go to', special purpose pliers, that I use to adjust/tweak/remove awkward small bolts/objects in tight locations, are also made by Engineer. and it is only just now that I noticed... They are PZ-57s if you are interested. 

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MIDI Pitch Bend - A Tiny Inconsistency

Sometimes 'The Bears' really are lurking, ready to get you if you step on the cracks between the paving stones... 

One of the things that has been beaten into me, over many years of working with hardware, firmware and software, is a rule that has many forms, but which boils down to something like:

"Question everything. Measure everything at least twice. Always ask: 'Why?"

It is an expanded version of the 'Never Assume Anything' rule. It has served me well. But you must never let your guard down...

Ever...

Photo by Synthesizerwriter

The MIDI Pitch Bend Inconsistency

As with all unexpected things, it crept up on me silently, unannounced, from a direction I wasn't expecting. When you have spent a long time with something, then you think you know about it. Since I got my first copy of the original MIDI Specification back in the mid 1990s, then I have read it carefully and repeatedly. I spotted some of the things that were put in there by knowledgable hardware people who really knew their stuff, like what a MIDI Clock message actually looks like 'on the wire' of a 5-pin DIN cable, and why it was defined like that. And since you are now intrigued, I'm going to leave that until another post...

So the original MIDI Specification 1.0 (1996) has a section for Channel Voice MIDI Messages, starting with Note On (0x8n in modern formatting, but shown in mid 90's style as 8nH, where 'H' means Hexadecimal and 'n' is the MIDI channel (0x00-00xF or 00H-0FH for 1-16)), then Note Off (0x9n, 9nH), through to Pitch Bend (0xEn, EnH). After that you have the System Common MIDI Messages, which all start with '0xF'. So all of the 'highest bit set' values are specified, from 0x9 to 0xF.

The Pitch Bend message is the last of the Channel Voice messages to be specified, and the specification  contains just two paragraphs - the second of which is just two sentences and is just clarification about sensitivity. Here's that first paragraph:  

This function is a special purpose pitch change controller, and messages are always sent with 14 bit resolution (2 bytes). In contrast to other MIDI functions, which may send either the LSB or MSB, the Pitch Bender message is always transmitted with both data bytes. This takes into account human hearing which is particularly sensitive to pitch changes. The Pitch Bend Change message consists of 3 bytes when the leading status byte is also transmitted. The maximum negative swing is achieved with data byte values of 00, 00. The center (no effect) position is achieved with data byte values of 00, 64 (00H, 40H). The maximum positive swing is achieved with data byte values of 127, 127 (7FH, 7FH).

There are quite a few important take-aways in this paragraph. Firstly: Pitch Bend messages are ALWAYS 14 bit resolution. Now I've done quite a lot of Max and MaxForLive devices, and Max is a very useful general purpose tool for exploring MIDI... In Max, there are two basic objects that are used specifically for receiving Pitch Bend messages (there are other, more generic MIDI 'parsing' objects...): 'bendin' and 'xbendin'. 'bendin' is the 'basic' object, and it returns 7-bit values for pitch bend of 0-127 (a single MIDI data byte), whilst 'xbendin' is the 'extra precision' object, and it returns 14-bit values from 0-16,383 (two MIDI data bytes)). 

7-bit and 14-bit Pitch Bend objects in Max

The next important thing here is that the 'bendin' object is throwing away the second byte, the Least Significant Byte (LSB), so the values that you get are just the raw 7-bit values (0-127) that are in the Most Significant Byte (MSB). As I'm sure you know already, individual MIDI 'bytes' only have 7 bits available for data, which is why the value doesn't have the range of 0-255. You need multiple MIDI 'bytes' in a message to get extra resolution. In the 14-bit-oriented way that MIDI represents higher resolution numbers, then for a value represented with two 'bytes', the MSB is the top 7 bits, and the LSB is the bottom 7 bits. So the range covered by the LSB is from 0x0000 to 0x007F (0 to 127) in steps of 1, whilst the MSB is from 0x0000 to 0x3FFF, in steps of 128. Now 0x3FFF is 16,383, so that's where the full MIDI Pitch Bend resolution of 0-16,383 comes from.

Note. I need to point out that Pitch Bend messages, by design, should include 14-bit 'extra precision' values - as noted by the MIDI Specification - because pitch bend is a 'special purpose' controller. Max and MaxForLive provide access to the 7-bit lower resolution value only because that value can then be used for other things, anywhere in MIDI or Ableton Live, or even externally if you convert it to a Control Voltage. For the control of pitch, then 14-bits are a much better idea, because this will give you nice smooth changes of pitch.

Ok. All sorted.

Not quite. There's a problem. 

Pitch Bend is bipolar: it can be positive or negative. In MIDI, the 'no bend', middle, detented position is defined as being a value of 8,192 (0x4000 or 4000H), which would be output as a value of 64 from Max's 'bendin' object and as a value of 8,192 from Max's 'xbendin' object. Max does provide another special object, called 'xbendin2', and this outputs the two 7-bit Bytes separately, so you can see the actual MSB and LSB if you want to. 

So negative Pitch Bend is the 64 values from 0 to 64 when we are talking 7-bit resolution, and the 8,192 values from 0 to 8,192 for 14-bit values. All perfectly fine and reasonable. But the positive Pitch Bend is slightly different. it can only go from 64 to 127, which is 63 values, because the highest 7-bit value MIDI allows is 127. yes there are 128 possible values in 7 bits, but if you start at 0, then you end up at 127. There are 128 values between 0 and 127. Max's 'bendin' object only provides 7-bit values for controlling other 7-bit parameters, and you would not use it for actually bending the pitch of a note - you would hear the steps! But the smaller numbers do make it very clear what is happening...

In 14-bits, then it goes from 8,192 to 16,383, and there are only 8,191 values, because 16,384 is ever so slightly larger than you can represent in a 14-bit number. 

The MIDI Specification 1.0 doesn't hide this. That final sentence of the first paragraph says:  

The maximum positive swing is achieved with data byte values of 127, 127 (7FH, 7FH).

The previous two sentences in the paragraph define the centre position and the maximum negative swing - but most people don't notice that 0->64->127 and 0->8,192->16,383 aren't symmetric. There is one less positive number than negative, and it is not hidden, it is in plain sight, printed in the specification. Unfortunately, the big numbers (16,383, and 8,191) tend to obscure what is actually happening...

In other words:

If no pitch bend at all has a value of zero, then the most negative pitch bend value is -8192. But the most positive pitch bend is 8191. (14-bit values are used here because these are what pitch bend applies to!)

Yep, The Bears just got us. 

The MIDI Pitch Bend Message doesn't allow us to bend up by the full amount. We can bend down and produce 64 7-bit messages or 8,192 14-bit messages (assuming our MIDI Controller outputs every value as a message, but that's another story). But when we bend up, then there are only 63 7-bit or 8,191 14-bit messages that can be output. That final value (8,192) is just outside of what MIDI allows. 

This means that if you set your PitchBend sensitivity to be 1 octave, then you can bend down by exactly one octave, but you will only be able to bend up by slightly less than one octave. The Owner's Manuals for MIDI Controllers, synthesizers and any other devices that output MIDI Pitch Bend messages generally say it exactly like it is - they say what the maximum positive output is. What they tend not to mention is that this is slightly less than what you need to do a pitch bend up that has the same range as a pitch bend down. And with 14-bits of pitch resolution, then the difference is very tiny. Miniscule.

In fact, I would guess that you've never noticed it...

So if you want exact pitch bend that utilises the end-stop of the Pitch Bend wheel or lever, then you should only bend downwards. This applies to any device that uses MIDI 1.0, regardless of age, firmware, operating system or manufacturer. Oh, and MIDI 2.0 is... different, because it has even higher resolution available.

Actually, there's another solution, and that is to not use the limits of the Pitch Bend wheel or lever (or push pad, or however it is implemented on your device), and that is to set the range to one note more than you require, and then to only move the wheel, lever, etc. by the amount required to get the bend you actually require. So for an octave, you might set the range to 13 notes up and down, and then only ever bend up or down by 12 notes. This gives perfect pitch bending, albeit with slightly less than the resolution of 16,384 values that were intended by the MIDI specifiers (but only very slightly less!). It does mean that you can't use the end stops of the wheel, lever, etc, but that's a minor inconvenience, and Pitch Bend by ear is so much better than relying on mechanics...

Oh, yes, and if you are thinking that this is a tiny difference in the pitch bend, and that it doesn't matter, then re-read that section in the MIDI Specification 1.0. It says that the MIDI Pitch Bend messages always use 14-bits resolution BECAUSE '...human hearing... is particularly sensitive to pitch changes.' I will gloss over the fact that it then goes on to define positive MIDI Pitch Bend so that it isn't perfect in precisely the place where human hearing is particularly sensitive. 

Not an Error

Actually, there is no error at all here. Nothing to see. This isn't a mistake by the people who wrote the MIDI Specification 1.0. It is nothing more than a consequence of the way that number work in these particular circumstances. Image the simplest pitch bend controller: three positions, No Pitch Bend (in the middle), Full negative (at one end of the travel of the wheel, lever...), and Full positive (at the other end of the travel). So these could be represented by -1, 0 , and +1. But this gives a jerky pitch change, of course!

If we increase the resolution by 5 times, something interesting happens. The range is now -5 to 0 to +5, and there are 11 values instead of the 3 values that we had for -1, 0, and +1. So if we start with the most negative value (-5) and assign it to 0 on the pitch wheel, then the middle (zero) value will be at 6 on the pitch wheel, and the most positive value will be at 11. Aha!: We have a Marshall Amplifier 'goes up to 11' situation. Unfortunately, no matter how the range and the resolutions are set, there will always be an odd number of values, consisting of the negative numbers, plus the negative numbers, plus that zero in the middle. So we need a controller with an odd number of values (which would also be very useful for that Marshall amp!)...

The binary world of computer is even. The basic counting system (binary) is based on two values: 0 and 1. So if you have just one bit to represent a number then there are two possible values: 0 and 1. Two bits gives four values: 00, 01, 10, and 11, which are 0, 1, 2, 3, and 4 in decimal number form. Any number of bits used will always give an even number of possible values. MIDI's 7-bit numbers have 128 different values, which are normally shown from zero: 0 to 127.  MIDI's 14-bit numbers have 16,384 different values, and if we show them from zero they go from 0 to 16,383. 

(Decimal numbers are even as well! So are pairs, dozens...) 

When we take these even numbers of possible values and try to map them on to a Pitch Bend wheel, lever, etc. then there's a problem, because we now know that the total number of pitch bend values are always odd since there has to be a zero in the middle. The positive and negative values are symmetric and have the same range, but the need to have a zero position in the middle adds an extra number and we get an odd number of values. No matter how hard you try, if you have an even number of drawers and an odd number of things to put in those drawers, there will always be at least one empty drawer or thing left over (each drawer will hold only one thing, of course, in this scenario - which matches the way that numbers work very nicely!). 5 drawers and 4 things? One drawer will be empty. 4 drawers and 5 things? One thing will be left over, because all the drawers will be full. 

One possible solution is to have two zeroes! If you assign two of the values in the middle to zero, then you can have perfect matching! 

In the case of MIDI Pitch Bend, the design puts a single 'no pitch change' zero value at 8,192, full negative at 0, and full positive at 16,383. So the negative range has 8192 different values (0 to 8,192), and the positive range has 8,191 different values (8,192 to 16,383, which is 8,191). There is no value of 16,384 because that would require a 15-bit number, and we only have 14-bits. 

So the biggest negative pitch bend message value is -8,192, and the biggest positive pitch bend message is +8,191. The biggest possible positive pitch bend is always going to to be 1/8192th smaller than the biggest negative pitch bend message value. And it is a teeny, tiny value! Nothing to worry about. It is a minute pitch difference.

But it is an inconsistency!  

  

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Emulating Turning a Circuit On and Off...

On the Discord channel of the Music Hackspace, @MeLlamanHokage asked about simulating power failure in a circuit using Max. I replied:

For audio circuitry then you would need to implement things like: a low-pass filter with a falling cut-off frequency, an increasing noise floor, rising (or falling) gain, falling clipping levels (eventually to zero volts), rising intermodulation distortion, a changing dc offset...

(Here's a sneak peak of what was in my head - scroll down for more details):

And so I idly wondered if it would be possible to do this in Max - actually MaxForLive, because I realised that this might be a perfect addition to the Synthesizerwriter M4L Tape (etc.) Suite. In much the same way as you record release 'tails' or 'triggers' on a piano, so that you capture the sound made when a key is released and returns to its 'resting' position, then I realised that an M4L emulation of a circuit turning off (and on again) could be used to add in the sound of Guitar Effects Pedals being switched in and out of circuit, or having their power removed. 

This was hugely reinforced when Christian Henson @chensonmusic (of Pianobook and Spitfire Audio fame) did a performance using just a piano sample loop and his pedal board with 23-ish pedals in a recent YouTube video (there are three Moog pedals off to the right of the screenshot, btw):

https://www.youtube.com/watch?v=yF6VKqpC-nA

Christian Henson @chensonmusic 'plays' his pedal-board...

That's two nudges in the same direction, so I stopped dithering, and started coding...

Now I know that the Effect/Bypass switches in guitar effect pedals are designed to not introduce any glitches or noise into the audio stream, and I know that they don't turn the power on and off to the pedal (well, not usually)! But not all of them will function perfectly, and power supplies do fail! I have had several synths, groove boxes and drum machines that made interesting sounds as you turned them off, and I would venture that this may be an obscure and under-utilised area of sampling. See the 'Reminder...' section near the end of this blog post for more thoughts on this...

Are the sounds made by foot-switches on guitar pedals part of a performance? That's an interesting question... So how can I make it easier for everyone to explore this aspect as well?

Free Samples...

Whilst there may be little actual sonic effect on audio signals from a modern foot-switch, there's also the very characteristic sound that guitar pedal foot-switches make in the real world: definitely something to sample and use as a drum sound. (I notice that the BBC overdubbed the sound of a gun being cocked to replace the sound of a seat belt clicking in a recent 'edgy' trail...) So I did some sampling of switches from a variety of sources*, and this turned into an editing session, and you can get the results here for free:

Free Footswitch Samples

These samples were created by me, and are released into the public domain with a CC0 licence. There are the sounds of foot-switches, and lots more. 'Footswitch14' and 'Footswitch Toggle 14' are what I think of as the classic guitar pedal foot-switch sound, but you may have your own preference...

As always, let me know if you want more of this type of content.

*Sources: Eventide H9 Dark, EHX Oceans 12, EHX SuperEgo+, Poly Digit, Donner DT-1, MIDI Fighter Twister, Rebel Technology OWL, and some other obscure stuff...

ON Off Emulation

But back to the main topic! I hasten to say that my original reply wasn't based on having already seen something that emulated a circuit being turned on and off. Instead it was me quickly thinking about what happens when you remove power from a circuit. So now that Christian Henson had inspired me to  actually design something, I jotted down some more concrete ideas about what 'turning something on and off' actually meant, as well as what sorts of things would happen to the circuits.

The first thing I thought about was a state diagram. At first sight, this is obvious. There are only two states: On and Off. Duh! (And there's the trap, sprung...)

But, with a little more analytical thinking, it is slightly more complex than this. There are actually four states: Off, Turning On, On, and Turning Off. The 'Turning On' and 'Turning Off' states normally happen so quickly that we don't notice them, but the longer they are, the more you notice them. One example where I've done this type of thing before was in a few MaxForLive devices that I programmed where the volume can be set to rise or fall slowly. The 'generic' version is called '3rd Hand' because it almost gives you a third (and very steady) hand to control rising or falling volume...

http://blog.synthesizerwriter.com/2018/07/slow-fades-in-live-performance.html - 3rd Hand

What was really interesting was that when I included an automatic volume control in a device: https://maxforlive.com/library/device/5258/instsineatmosphere then the only feedback comment that I received was that it didn't make any sound, which was exactly what happened with an earlier 'Generator' device where you had to click on a button to get a sound, so since then I have not included this feature in many devices. Please let me know if you would like it to be included!

But back to 'states'...

This '4 states' approach is very useful for live performance where you want something to evolve slowly: like an audio drone that gradually evolves over minutes, or tens of minutes, or longer. If you've ever gone to a 'Drone' performance by William Basinski (there are other audio drone specialists) then you will know the power of a slow rising, changing sound. I saw/heard him when he performed at Ableton Loop 2017...there's something about live performance!

In a drone performance, then there's a lot of slow evolution of sounds, timbres, volume... (the 'Turning On' and 'Turning Off' states) and very little time when the soundscape is static ('the 'On' or 'Off' states). To revisit the old saying about music typically being made up of 'sound and silence', then a better recipe might be that music is effective when it contains interesting/evolving/changing sounds seasoned with  sprinkles of silence.

In this case, the change was all about what happens when circuits get turned on or off. And so I jotted down as my first proper approximations:

- Mains breakthrough from PSU

- Clicks, Crackle and Crunches

- Noise (Rising then falling?)

- Clipping of audio

- Loss of high frequencies in audio

And then I programmed a custom 'synthesizer' in MaxForLive, that used a 4-state slow On and Off generator to drive 5 sections producing each of the 5 features that I had noted. I'm sure there are more things happening, but this was a first attempt, and I'd never seen anything like this before... Now when I say 'I programmed' then that can make it sound like something trivial and rapid, but that isn't quite how the process works... Anyway, some hours later...

OnOff Emulation mr 0v01

There are five processing sections, plus a sixth 'control etc.' section on the right hand side. The layout is much the same as the Ironic Distortion and the Ferrous Modulation M4L devices:

Ironic Distortion - blog post.                   Ironic Distortion - M4L.com

Ferrous Modulation - blog post.              Ferrous Modulation - M4L.com

Each section has a 'Mix' slider, which sets the level of the output of that section, with a big 'Mute' button.

But because this device is all about controlling the 'Turning On' and 'Turning Off' states, then there are extra controls in each section which are devoted to setting how time affects each section. The big grey bar is the current On/Off stare: 100 (all of the bar is grey) is On, whilst 0 (all of the bar is black) is Off. The movement of the bar (up or down) shows how things change during the 'Turning On' or 'Turning Off' states. To the right of the bar are four rotary controls, plus an 8-button switch selector.  The screenshot above is for the I (Impulse) section, and so each starts with that letter. So I-On is the time for the bar to move from Off to On (the 'Turning On' time), whilst I-Off is the time for the bar to move from On to Off (the 'Turning Off' time). 

The 8 position switch in the middle controls how the bar moves. L is for Linear, and so the bar just moves steadily from top to bottom (a straight line graph). S is for Sine, so the bar slows down as it gets to On. Z is Sine-squared, so the slow-down is sharper. P is for Power, so the bar slows down as it gets to Off. Q is for Power-squared, so the slow-down is more abrupt. B is for Bipolar, so the bar slows down as it gets to On and Off. D is for Bipolar-squared, so the slow-down is more abrupt. Finally, the light purple S that is linked to the two rotary controls on the right is for 'Smooth' (or Log) and this provides two additional controls that let you change the time of a logarithmic slow-down as the bar gets near to  the On or the Off. 

The best thing to do is to listen to the effect that the controls have! It is a bit like the Flutter waveforms in Ferrous Modulation: sine sounds boring, whilst the narrow spikes sound jerky... You may find that the Smooth setting gives good control, but remember that the S, Z, P and B positions don't have the slow-down at one extreme, and so have a very different effect. If you find yourself over-using the Smooth setting, then deliberately choose one of the more abrupt options. 

Separate controls are provided for each sections because the final sound works best when each of the sections has different timing! If you have the same timing for all of the sections than it will sound boring... One of the settings that I like is to have the Clip section happen last, so that you last thing you hear is the distorted audio...

The Sections

From the left hand side, the sections are in two parts. The first two sections process the incoming audio signal: 

Freq. 

This section just applies a low-pass filter to the audio input, and lowers the cut-off frequency as the bar moves from On to Off.  (Or raises it as the bar moves from Off to On). This emulates that way that some audio circuitry loses high frequencies as the power supply is reduced.

Clip.

This section has optional Compression and Filtering, but the main effect is to apply clipping to the audio input, where the clip limits reduce with the bar, so there is no Clipping when the device state is On, but more and more clipping as the state approaches Off. Note that the volume drops to zero when the state is Off... so you don't hear the effect of extreme clipping when the limits are zero! 

The next three sections are generators, and so do not process the incoming audio:

Noise

White noise filtered to give it various colours. The 'Gain' control and the mixer slider are effectively the same control...

Impulse

Crackles, pops, crunches, and other 'Impulse'-type sounds are generated in this section. Because the source is random noise, then these will not repeat - every time a new and different sound will be produced. 

Mains

This is slightly strange - it introduces the sound of mains hum, which is weird if the power supply has been removed. But, curiously, as with many sound effects, nonsensical often seems to work. It seems that having mains hum fade in and out implies something happening with the power - a bit like the way that sparks always seem to jump out of control panels on spaceships and submarines in the movies whenever there's an explosion or a collision. 

None of these sections is particularly complicated. Each does just one generation or processing function. Feel free to look into the code to see how each works - there's no magic used. 

The Big Button

One the far right hand side is the Control section. This has the big 'On/Off' button, text that shows the current state (1 of 4 possible states), and a generous set of memories for your own creations - just shift-click to store, click to recall. 

Reminder...

It is probably worth noting that although this might appear to be just a generator of On/Off sounds, it is also a processor - the Freq and Clip sections process audio, so if you don't input anything then they won't produce any output. The ideal setup is to have a sound that you want as the basis of the final result, and then use OnOff Emulation as a way of 'bracketing' it with an On and Off emulation.

This is slightly different to what the Release Trigger or Release Tail samples are doing in a piano sample - they deliberately do not include the piano note itself. (You press down on a key, and then release it, so you get the sound made by the key itself and the associated 'action' mechanics, but you don't do this with a string vibrating. ) OnOff Emulation processes any audio that is sent to it, and then layers the mains, impulse and noise sections on top of it. 

And this set me thinking... What about a release trigger/tail generator? I've never seen a dedicated one, because all of the release triggers/tails that I've ever used have been just samples that are part of the sample set of a sampled piano... The only exception that I can think of is for Harpsichord-type sounds on a DX7 (or other FM synths,  etc.), where the sound of the jack hitting the string when the note is released, is synthesized separately using an envelope where the attack and decay are short, the sustain level is zero, and the release rises to the final/initial level to give an envelope that only affects the release segment of the note. Oh, and acoustic guitar sounds, where string buzz is also synthesized this way... There's probably more now that I'm thinking about it this topic...

So a release trigger/tail generator would have different sections, and actually might have some aspects in common with a 'Riser' synthesizer, although they don't seem to be as 'in vogue' as they were a few years ago. I remember suggesting that a custom riser could be made by processing a 'reverb'ed section of a track in an Ableton Loop Studio Session back in 2015, and got a tumbleweed reaction from the room, and the 'Name' Producer then made one using filtered noise and everyone else nodded their heads. I have always swum against the tide, by the way...

I have added this task to my 'ever-expanding' (as Loopop says on YouTube) list of 'things to do'.  Don't hold your breath, though: it's a long list and I'm very busy. 

And Finally...

I haven't really seen a device like this before. Well, actually, that's not exactly right, because I have - this is just a synthesizer, but not your common variety. This is a purpose-built 'custom' synthesizer made to produce just one type of sound.  What I haven't seen before is a synthesizer that is dedicated to making the sound of a circuit being turned on or off. But now I have. And so do you, now! I think it shows the power of Max that you can use it to make arbitrary sounds and sounds that haven't been made before (or maybe sounds that I think might not have been made before...).

My grateful thanks to @MeLlamanHokage for the original question about turning circuits on and off, and to Christian Henson for using a pedal board as a performance instrument at exactly the right moment to get me to turn speculation into reality. Thank you, sirs!

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Getting ONOff Emulation

You can get OnOff Emulation here:

    https://maxforlive.com/library/device/7074/onoff-emulation

And yes, I realise that it should be called AUDonOFFemulation if I was to use my own naming scheme, but that just reads crazy!

Here are the instructions for what to do with the .amxd file that you download from MaxforLive.com:

     https://synthesizerwriter.blogspot.co.uk/2017/12/where-do-i-put-downloaded-amxd.html

(In Live 10, you can also just double-click on the .amxd file, but this puts the device in the same folder as all of the factory devices...)

Oh, yes, and sometimes last-minute fixes do get added, which is why sometimes a blog post is behind the version number of MaxForLive.com...

And no, I haven't had a chance to test it in Live 11 yet... To much to do, so little time...

Modular Equivalents

In terms of basic modular equivalents, then implementing OnOff Emulation is a mixer plus a VCF plus a Clipper plus a noise source, an envelope follower and a State-Variable Filter, plus a trigger and 10 AR envelopes. Nothing is complex, but there's quite a lot of separate bits to deal with, which can be tricky on a modular... 

Overall, I reckon that OnOff Emulation would require an ME of about 20. Alternatively, you could just take any modular patch and see what happens when you power it down and up. (Caution: Turn your amplifier volume down, and use a limiter on the input. Increase the volume slowly and carefully - with caution. Not recommended with headphones! Do not turn modulars on and off repeatedly and quickly!) 

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