Tuesday, 27 October 2020

A Deeper Look at Single Cycle Waveforms - [Single Cycle part 2]

The revisit to 'The 30dB Rule' got me thinking. I realised that there is a lot that most sources don't tell you about waveform displays. Now, if this was a YouTube video, I would be using an eye-catching picture and a bold primary colour headline like:

'Hidden waveform secrets!'

But this is a blog, and I'm not a great fan of the hyper-sensationalism that YouTube seems to encourage. Instead, I've decided to extend 'The 30 dB Rule Revisited' into a few parts, so that I can cover the topic in more detail. (and in even more detail in the next edition (4th) of my 'Sound Synthesis & Sampling' book) 

The Waveform

Let's start with the sort of diagram that normally appears in textbooks (or on Pianobook videos!) and build from that.

This diagram shows a single cycle of a sine wave: a 'sine' waveform. The time axis is horizontal, and the amplitude (posh word for volume, size, voltage...) is on the vertical axis. The convention is that you show the positive part first, starting with the first time that the waveform crosses the zero axis, and then the negative part, finishing with the third and final zero crossing. This is just a convention! You could start anywhere on the waveform if you wanted - the waveform doesn't care. But you do need to show at least one cycle, of course. 

I normally try to avoid too much maths in this blog, but there's an important formula in the diagram: the relationship between Frequency and the time it takes to complete one whole cycle (this time is called the period, which harks back to pendulums and clocks and physics...). The formula is:

Frequency  = 1/Period

Which can be clarified as:

Frequency (in Hertz) = 1/Time to complete One Cycle of the waveform (in ms)

For most audio signals (posh word for sounds, audio voltages...) then the time is usually conveniently measured in milliseconds (thousandths of a second) or microseconds (millionths of a second). Seconds are too big a unit of measurement for the short time that it takes for a sine wave to wobble. For example, a 50Hz mains electricity (in some countries) takes 0.02 seconds to complete one cycle, but it feels much easier to say 20 milliseconds (200ms). In countries where the mains frequency is 60Hz then it takes 16.666... milliseconds (or 0.0166666... seconds) to wobble up and down (or down and up - remember it doesn't matter!)

The formula means that as frequency goes up, then the period (time for one cycle) gets shorter. So a note of 440Hz (an 'A') would have a period of 2.273ms. Up an octave to 880Hz, and the period is 1.136ms. Up another octave to 1760Hz and we have to move to microseconds for expressing the period: 568.2┬Ás or 0.5682ms. This is quite a wobble, if you think about it - the waveform is going from positive to negative in just over half a thousandth of a second.

'Wobbles' is actually a very useful word for describing sine waves. That smooth shape is not an accident - it turns out that if you make a ruler twang, or watch a pendulum swing, or pluck a string, or turn an amp up so that you get feedback from a microphone, then the basic shape of the time wavform that you get is probably a sine wave (or similar). This is because objects in the real world are lazy - they expend as little effort (energy) moving as possible, and the way of something moving back and forth (or up and down, or side-to-side) using the least energy is a sine wave. Look up 'Simple Harmonic Motion' if you want to read about maths and physics... It a bit like: 'the shortest distance between two points is a straight line' - it's one of those fundamental things about how the universe works. 

If you want, you can do a totally non-scientific experiment that kind of illustrates this 'least energy' thing. Hold out your arm, with your hand vertical. Now wave it from side to side smoothly counting 'one thousand, two thousand, three thousand...' so that you are doing one cycle every second (a frequency of one cycle per second is 1 Hertz (Hz), and has a period of 1 second!). Then try jerking your hand as quickly as you can between two stationary positions about 100mm apart at the same rate. Instead of a smooth, continuous waving movement, it should be just two brief movements interspersed with waiting. You should find that it feels like you are using much more energy to do the 'Square-shaped' wave than the smooth 'Sine' wave.

So the sine wave is interesting and important because it is least energy and a very smooth shape, and it turns out that it is pretty fundamental in other ways too - you can make any waveshape by adding together sine waves of different frequencies, amplitudes and phases. Phase is just the relationship between two waveforms. If they are 'in phase' (no phase difference) then they go up and down at the same time... Let's look at some diagrams:

Zero crossings are one of the standard places on the waveform that are used to determine phase differences. In the diagram above, the blue circles highlight the zero crossing as the descending sine wave crosses the horizontal zero 'time' axis. From left to right, the two sine wave are: 'In phase', 'slightly out of phase', and 'Out of Phase'. The 'Out of Phase' can also be called 'Anti-phase' - here the two sine waves are opposite: as one goes up, the other goes down, and vice-versa. Phase is normally measured in degrees, as if the time axis was wrapped around a circle. So 'In Phase' is a phase difference of 0 degrees, whilst 'Out of Phase' or 'Anti-Phase' would be 180 degrees. If you keep increasing the phase difference, then eventually you go around and end up at 359 degrees, then 0 degrees as the two waves are in phase again.   

The diagram above uses the highest positive peak of the sine wave, which is often easier to see on some waveforms. Again, the three examples are: 'In phase', 'slightly out of phase', and 'Out of Phase'. Phase difference can be measured with any waveform - sine waves are just used here as examples. Waveforms that have lots of similar height peaks, or lots of zero crossings, can be difficult to try and figure out by viewing the waveforms. There are electronic meters which can measure phase in audio signals, and these are used in filter design, loudspeaker crossover design, phase pedal effects, and more. There are filters called 'all phase filters' that only change the phase of signals that pass through them, and a phase meter is used to characterise them.    

Earlier, I noted that it was possible to make any waveform by adding together sine waves of different frequencies - well, there's an exception to this... The only wave that can't be made by adding together two or more sine waves is... a sine wave. A sine wave has only one frequency 'inside' it, which is why it sounds so 'pure' when you listen to it, and also it is why it is so smooth in shape. If you added any other sine waves then the shape would be less smooth. Circles and sine waves are the ultimate in smooth!

When you turn down the cutoff of a low-pass filter, then you can hear high frequencies being removed from the signal, and eventually, there is only the sine wave left. So the lowest frequency you hear is the frequency of the sine wave itself, and if you turn the filter cutoff down even lower, then even that sine wave will vanish and you get silence. 

Combining and extracting multiple sine waves will be examined in more detail in a future part of this series.

The Waveform - Deeper

Wrapping the time axis around so that it forms a circle is actually a clue to what the 'single cycle' diagram really is - a convenient abstraction that turns reality into something easy to visualise. Actually, a waveform is always moving up and down in time, and the waveform view just captures that up and down movement and makes it visible. If the sine wave is a sound, then the waveform shows how the air is compressed and rarified (not a word we use very much!). If the sine wave is a voltage, then it shows the change of voltage as the sine wave moves up and down. So a waveform is like a long exposure photograph that freezes movement. (or a stroboscope that uses a flashing light to 'freeze' movement...)

Let's dig deeper into a waveform:

The first thing that often surprises people is that the waveform continues both backwards and forwards in time. The waveform repeats over and over again. each time the same. (If the waveform changed over time, then there would not be a single waveform that we could use to represent it.) This means that the start of the waveform (let's start at the first zero crossing) is also the end - remember the wrapping round of the time axis into a circle. This means that the level at the start has to be the same as the end - but it also means that the slope of the line at the start has to be the same as at the end. 

One of the things that is often never explained by most sources is what you can and can't do with a waveform. The levels and slopes being the same at the b=start and the end is the first 'not immediately obvious' thing to note. But the diagram above shows some more.

The steepest slope is one that is almost vertical. It can't be completely vertical, because this would mean that there were several different values at the same time (the horizontal axis is time...), and this isn't usual in this reality. You might like to try thinking about a different universe in which an LFO could have several different values at the same time, and follow it up by Googling 'Schrodinger's Cat'. 

You can't have slopes that go over, because this would mean that the waveform was moving backwards in time. Equally, a waveform can't cross over itself, because this would mean that there were two values at the same time.

Finally, you can't have a gap in a waveform - there is always an amplitude value for each time. Now a digital waveform is actually just a series of sample values, but you can't have empty sample values either.  

One 'Ah, but...' FAQ at this point often revolves around sawtooth or square waves. They may appear to have gaps in them, but actually it is a very steep (nearly vertical) slope, followed by a much slower slope. On some screens, especially old oscilloscopes, the slow slope is all that is visible, but it doesn't mean there is a gap. In digital samples, a sawtooth can change from a big negative value to a big positive value with a single clock. Some text-book diagrams don't always show sawtooth waveforms as being continuous, and these are called 'idealised' (meaning 'not realistic').   

Square waveforms are mostly flat (remember moving your hand from side to side in the experiment, earlier?), but they again have almost vertical slopes joining the two flat portions, and again they are sometimes shown in text-books as idealised pairs of flat output values with gaps between them.

Any time that there is a nearly vertical slope or a sharp corner (like on a square wave or a sawtooth wave), then this indicates that a high frequency is present in the audio (more about this in a future part). If you filter a square wave so that some of the higher frequencies are removed, then you get ripples on the top and bottom portions of the waveform...

These ripples are caused by there not being enough high frequencies to draw in the flat top and the almost vertical edge. In the diagram above, can you imagine what frequency sine wave would be required to cancel out the ripples in the top and bottom portions?

One useful diagnostic technique works well on waveforms which are approximately square. If you imagine a vertical axis in the middle of the flat portion, then if the waveform is 'mirrored' around this axis, then this means that the audio contains mostly odd multiples of the fundamental frequency. If you think about this, then a square wave is the perfect mirror around that mid-flat axis, and it turns out that a square wave does contain only odd multiples of the fundamental frequency. 

In physics (and most science) you are often given a theory or Law, and then told when and why it doesn't always apply. In the real world there are often 'ah, but's! So it turns out that the mirroring effect only works when the harmonics are all in-phase... So, phase can sometimes be very important. Let's investigate this:

The above diagram shows the effect of changing the phase of the sine wave that it 3x the fundamental sine wave frequency. 10 degrees (there are 360 degrees when you go all the way round a circle, so 10 degrees is a tiny amount) puts some wobble into the top of the waveform, so it isn't flat any longer. 30 degrees makes the ripple bigger. By the time we get to 60 degrees (cutting a pie into 6 equal pieces gives you six pieces each with a 60 degree angle) then the square wave is almost lost and mostly what we see is a jagged waveform that has a lot of a waveform that is 5x the fundamental sine wave frequency.

So for the shape of a waveform, phase matters. But if you listen to all of these waveforms, then they all sound like a square wave. More on this in a later blog post...

One final thing that you can see just by looking at the shape of a waveform is the 'DC offset'. All of the waveforms so far have been centered on the zero axis, so there's no overall voltage present all the time - the positive and negative bits just cancel out. But the diagram above shows that if the area above that horizontal zero axis line is different to the areas below, then you can get a DC offset. DC offsets can make clipping asymmetric, which changes the sound, and they can cause clicks if you connect a cable carrying an audio signal that has a DC offset. 

Just to come full circle, the above diagram also serves as a reminder about 'The 30 or 40dB Rule' - if there is part of the sound that is lower in level by more than about 40dB, then it will be too small to be visible in a waveform. I am pretty sure that this aspect would probably get breathlessly emphasised in some YouTube videos!

Single Cycles

You should now know more about single cycle waveforms, what they are, how they work, and some of the terminology around them. You also now know that the shape of the waveform is related to the timbre (or sound), and that the shape can be produced by adding together different frequency sine waves - although it seems that phase complicates this. But essentially, a single cycle waveform is a little fragment of timbre - actually, it is the smallest fragment you can have that gives a specific sound. 

Single cycle waveforms were used in early analogue oscillators, and were chosen to provide a diverse set of timbres. Sine waves for simplicity and 'purity' of tone. Square waves because they sound hollow. Sawtooth waves because they sound sharp and bright. And Pulse waves because they sound thin and buzzy. All of these are fixed 'snapshots' of timbre - they don't change over time.

Over time, oscillators were extended to provide additional waveforms - and significantly, these can change over time: Pulse Width Modulation changes the shape of a pulse wave so that it sounds 'animated' or 'chorussed'. Oscillator sync resets one oscillator waveform using another, and so generates a distinctive, glitchy sound. 

Using wavetables instead of fixed singe cycle waveforms allows the timbre to be changed over time: either smoothly by interpolating from one waveform to the next, or simply jumping abruptly from one to the next, which can give a fascinating 'glassy' texture. The ultimate smooth and long wavetable is a sample, of course, with a long looped sustain and a long looped release. 

So there's a basic split between the single cycle, 'fixed' waveforms, and the multi-cycle, 'timbre changes with time' waveshapes (made from wavetables or samples). Of course, the ultimate 'timbre changes over time' is noise, where it is randomly different every time. 

There's a simple model for sound synthesis that has Controllers controlling sound Sources like oscillators whose outputs are then altered in timbre and volume by Modifiers to produce the final sound output. When the output of an oscillator can change timbre over time then it stops being a pure 'Source' of sound of acquires some of the functionality of a Modifier, but then models are only supposed to be approximations of reality.

The table above fills in some of the possible sound Sources and sound Modifiers. One common application of the Source and Modifier model is for subtractive synthesis:

The above diagram shows how the oscillators and noise are treated as Sources, but it also has PWM (Pulse Width Modulation), Oscillator Sync, and FM as sources. The mixer is also interesting - does changing the mix between Sources count as a Modifier?

The diagram above extends the 'Modifiers' section so that it includes PWM, Sync and FM. This effectively leaves the oscillators with only fixed single cycle waveforms as Sources, plus noise.

Here's a table which emphasises that the only non-Modifier sound Sources that don't 'change timbre over time' are single cycle oscillators and noise generators. 

Simple oscillators with just a few single cycle waveforms (plus noise generators) are the main source of sound in synthesizers, but so far the only waveshapes that have been mentioned here are sine, square, sawtooth and pulse. This might appear to be a major limitation, and so the only area worth looking at might seems to be the Modifiers... But it turns out that this is not the case at all, and the next blog post will examine what you can do to exploit the possibilities of single cycle waveforms as much as possible. 


Pianobook videos (This one describes an ADSR envelope...)

Pianobook.co.uk (Free user-created samples (crowd-sourced samples?))

Schrodinger's Cat (A sideways route into quantum physics...)


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Monday, 19 October 2020

The 30 dB Rule Revisited - [Single Cycle part 1]

Technology changes all the time, sometimes for the better. One of the background tasks that tends to get forgotten is to revisit and re-assess old assumptions, and to update them when necessary. The mains electricity wiring in a house is one example - the insulation in cables does not last forever, and rubber, PVC and plastics all degrade over time. Recently, I have been blowing the dust off some of my archives of audio samples, and it reminded me of one of the first articles that I wrote for Sound On Sound magazine... It is still available on the Interweb:

SOS - The 30dB Rule   

It set me thinking: how has time changed this 'rule of thumb' that says that if you look at a waveform, you can only see the 'top' 30dB of whatever is in it? I was curious to see if advances in technology meant that this needed a revision...

Into the past...

May 1986 is over 35 years ago, and a lot of the things that you now probably use everyday didn't exist in anything like their current form: the World Wide Web, the Internet, HTML, cheap domestic microwave ovens, cheap laptop computers, LCD video monitors, mobile phones, MP3, DAT, DVDs, DAWs... and high streets full of little more than charity shops and coffee shops. 

Digital audio was possible, in a limited way, on a hobbyist computer. If you were in the know, then researchers in places like George Lucas's Sprocket Systems were working on prototype DAW-like technology, and if you had lots of money, then New England Digital's Synclavier was shipping direct-to-disc recording of digital audio, or you could sample at 8-bit resolution on a Fairlight CMI Series II, whilst you saved up for the recently-released Series III with 16-bits! Most ordinary hi-tech musicians were limited to just using computers for simple audio file editing, or for another relatively new innovation: MIDI. 

So, my samples from this time were mostly 8-bit, sampled at 8, 16, or maybe even the insanely high rate of 32kHz! They were mostly kept on 3.5 inch floppy disks (From Sony, which were encased in plastic and so weren't actually 'floppy' at all...) or on a hard drive which would be a few hundred Megabytes in a case about twice the size of a modern hard drive with a few Terabytes. To look at the files, CRT (Cathode Ray Tube) monitors would be used - VGA resolution (640x480) monochrome LCDs didn't appear until 1988, and colour LCDs didn't become affordable until the 1990s. You might like to create a graphic image sized 640 x 480 pixels on your current computer to see just how small it really is - on my 27 inch 5K (5120 x 2880 pixels) monitor it covers about the same area as a credit card.

So here's an 8bit sine wave, displayed more or less 1:1, so there are 256 pixels from the highest to lowest peak (except it isn't - no matter what I do, my browsers won't show the graphic actual size. Strange...). Anyway, just imagine that the following graphic image is 256 pixels in height: 

Now on a VGA monitor, that 256 pixel high sine wave is going to be taking up just over half of the screen height, so it is going to be pretty large. If you looked at the same sine wave on an Oscilloscope (You can still get them, although now they are digital, have LCD screens, don't get hot, and weight very little!) then you would probably set the controls so that it occupied about the same sort of percentage of the screen height - especially since 'scopes often have all sports of readout on the screen for frequency, voltage, range, offset... Notice that despite the 8-bits and the small image, it looks like a sine wave!

To 2020...

If we now do this with 16 bits and a bigger screen, then we can fast forward to 2020. We can take the normalised output as our maximum output level (let's call it 0dB), then we can compare it with a higher frequency (4x freq) sine wave attenuated by 30dB (i.e at -30dB). Because these are going to show relative levels, I'm not going to align the bits to pixels here, especially because I can't show 16 bits on a sensibly-sized screen (the screen would have to be 65,536 pixels high, which is more than 20x the vertical resolution of my current screen (2880 pixels).

So, from left to right, we have the sine wave at 0dB, a 4x frequency sine wave at -30 dB, and the result of mixing them together. The waveform on the right looks distorted, and is obviously not a sine wave, and if you listen to it, then it is very easy to hear the 4x frequency sine wave because it is only 30dB down, and your ears are good over a much bigger dynamic range than that. 

But the middle screenshot is particularly interesting. A signal 30dB down is just about visible on modern screens, but you can imagine that if this was an oscilloscope with a slightly fuzzy line of light as the display, then it might be possible to see the sine wave. But many 2020 synthesizers feature waveforms shown on small OLED displays that are not even 256 pixels high, and so the vertical resolution is worse than a VGA monitor, and worse than the 8-bit example shown above.

So let's try the same process at -40dB.

As before, from left to right we have the sine wave at 0dB, then the 4x frequency sine wave at -40dB, and then the result of mixing them together. The middle screenshot is now much smaller, and the sine wave on the right looks like...a sine wave. If you listen to it, then you can hear the 4x sine wave. but it is not obvious from looking at the screenshot that the sine wave is impure at all.

Finally, how about adding noise instead?

This time the middle screenshot is noise at -40dB. The right screenshot is the result of mixing the sine wave and the noise. It looks pretty much like a sine wave to my eyes, although when you listen to it, you can hear the added noise (it is only 40dB down). On an oscilloscope, the width of the line is going to hide the noise even more effectively.

Let's simulate that:

So anything smaller - like below -40dB - is not going to be visible to your eyes at all...

The 40dB Rule...

Over 30 years of progress has given us better displays, and cheaper, lighter oscilloscopes. But it seems that just 'looking' at waveforms still only tells you about the top 40dB or so of the signal. Anything lower than that is not going to be visible. 

"You can only see the top 40dB or so of a waveform..."

and a useful pair of corollaries:

"Your ears are much better at hearing than your eyes. Don't trust waveforms."

"On a small OLED screen, you may only see the top 30dB of a waveform, or less."

At one time, it was quite popular for synthesizer manufacturers to provide the ability to draw waveforms (usually using light pens, but these days you would probably do it with a mouse)... Hopefully, you now know why this is not a good idea if you want to have control over anything other than the very loudest component parts of the sound.

Only today, I saw a Facebook post where a person was describing how an analogue synthesizer software emulation VST had been prepared with great care, emphasising that the actual and emulated waveforms had been compared on an oscilloscope 'very carefully'. Unfortunately, you now also know that this is not a good technique to base comparisons on. Instead, spectrum analysis of the waveform would show the frequencies that were present down to levels much, much lower than -40dB!  


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Wednesday, 30 September 2020

Taming LEDs - a practical guide to muting overly-bright indicators

Light Emiting Diodes (LEDs) started out 50-odd years ago as little red solid-state indicators that were expensive but soooooo much cooler than filament bulbs, and over time have replaced those old inefficient incandescent bulbs in lots of applications, particularly lighting. For indication purposes, though, there's a trend which seems to be growing, and that is using LEDS as power indicators that are way too bright, yep:

LEDs that are TOO bright!

The worst offenders are clear LEDs which can give very bright beams in some directions. There's a word for it: Glare! Quite a lot of my gear has LEDs that are far too visible, and my studio isn't dark! Some equipment provides control over the brightness of the UI (user interface), but this is quite rare, and doesn't always cover the power indicators. An interesting example is the awesome Synthstrom Deluge groovebox (and more), where there's a shortcut that allows the LED brightness to be set (the default is bright), but it works for all of the LEDs except the power LED, which glares yellow or green at the same eye-catching/distracting brightness regardless of the setting...


One approach would be to open up the piece of equipment, find the series resistor that provides current to the LED, and increase its value. The problems with this quickly make it impractical for many reasons:

-   Voiding the warranty (and even just opening some modern gear is not easy!)

-   Requires basic reverse-engineering of the relevant circuitry to find the resistor.

-   Requires skill and tools, and for surface-mount resistors; a proper desoldering station.

Your circumstances may vary, but I tend to restrict my customising to old/vintage gear where I'm fixing a problem like dead capacitors, faded display backlights, or adding extra functionality. Messing around with new gear just to fix an LED that is way too bright doesn't feel like a good idea...

Practical (Well, quick and dirty, anyway...)

So are there any practical solutions that don't involve diving into the internals of the gear? For a long time, I've used Blu Tack as my quick fix (My grateful thanks to Bostik, who manufacture it!). Just put a small blob of the slightly sticky blue stuff over the LED, and then put your fingernail through the Blu Tack to expose just enough of the LED so that it is visible rather than exposing the normal 'Woah: That's way TOO bright!' mode. Rapid, and 'fix & forget', although it does, of course, look rather like, nope, look exactly like, a blob of Blu Tack.

The red power LED on an old Roland M-240 mixer bought from Turnkey in London... 

The blue power LED from a 'no-longer-supported' M-Audio Firewire audio interface...

Over time, the Blu Tack method does tend to attract dust, and it doesn't look any less like a blob of Blu Tack. If you happen to catch it with your hand, then you get a sudden bright LED and you need to reshape it and try to expose just enough of the LED with your fingernail to restore it to 'how it was before'. Curiously, getting back to how you remember it isn't as easy as you might think, although what is very simple is making it too bright or not bright enough. Maybe the Blu Tack method is better described as:

Semi-practical: imperfect in several ways


Now, you are probably expecting something a bit more nuanced from me, so here are the two solutions that I have developed, plus a third that may suit some specialist situations.

First, there's what I call the 'Yogurt pot' method. You cut a piece of the white plastic out of the base of a yogurt pot (where the plastic is thickest), and fix that in place over the LED. You can use Blu Tack, although I find that a tiny blob of the clear flexible glue often described as 'general purpose' is better! You can see this in use on my 'Tuner' - a chromatic guitar pedal that I use for tuning up analogue gear and even digital gear that requires it (FM sounds on the Deluge are one example...). The designers at Donner drive the LEDs very brightly, and whilst I can see that this might be good on a very bright stage, auto-brightness or a bright/dim selector switch could well be a good sales feature!

Then there's the 'Envelope' method. I cut a small disc out of the two layers of plain white paper from an old envelope, and then glue that over the LED. If I feel like ignoring the warranty, then I might open the case up, push the LED inwards a bit, and then put the paper on the inside. On my Deluge, I put the paper underneath the Mxpand overlay (a very useful and informative add-on that augments my poor memory very nicely!). 

Finally, there's the 'New Age' method, which twists the 'Quartz crystal lamp' meme (that you see in many studios) to suit my nefarious purposes. You must have seen them in those depressing 'Is your studio ever THIS neat and tidy?' publicity photos of other people's studios: big lumps of yellow quartz with LEDs inside them, that glow reassuringly, but which don't need watering like plants or cactii. So my variation is to go to the opposite end of the spectrum and get those tiny little lumps of quartz that have been polished smooth and are sold as decorations. A size of 5-10mm across seems to work well for glueing on top of an LED. 

'Tumbled' is the adjective often used to describe the polishing method, and you can find them in various sizes on Amazon and from craft shops. Rose quartz is good for red LEDs, but you can get purple Amethyst or green Aventurine or... My recommendation is to go to a craft shop because that way you get to see the size and smoothness of the polishing. Of course, you might want to get unpolished lumps if you prefer. Either way, the stone diffuses the light from the LED and prevents glare - and it looks rather natural and connected, if you are into that kind of thing. If 'Grand Designs' did studios, then the power LEDs on their gear would definitely be obscured by bits of tumbled quartz!

Just one more thing...

LEDs being too bright might seem to be trivial, and not worth considering at all. But anything that distracts can kill creativity. It is not an exaggeration when I say that the very first thing that I noticed about the Synthstrom Deluge was the power LED, and the very next thing I did was put a piece of Blu Tack over it and start to create music with the Deluge - which is totally wonderful, by the way. But as soon as I could, I used the 'Envelope' method to turn that bright distraction into a plain yellow or green dot that now does its job of telling me if the battery is charging or fully charged - and that's all that it does. 

Sometimes it isn't software, or firmware, or even fancy hardware.  


In a connected world, there are often fascinating consequences and misinterpretations of mundane actions. I bought some tiny tumbled rose quartz fragments from Smile.Amazon... (the bit where you force a charity donation) to cover up some glaring LEDs, and since then, I have had some intriguing emails and 'You may be interested in...' suggestions from Amazon. You may want to cover your tracks by using the 'I prefer not to use this for recommendations' tick-box...

Much more interesting, and very relevant to music gear, is what tends to happen when you buy stereo in/out, hi-end or boutique guitar pedals for processing synths in live rigs (and I'm definitely no stranger to this type of activity!). When you do this, then online music shops seem to assume that you must be interested in hi-end or boutique guitars as well...  (I'm not another pseudonym for the amazingly talented Benn Jordan, btw!) I did wonder if I could nullify the implied linkage by buying a hi-end or boutique synth from them - but that type of synth tends to come from specialist suppliers that rarely sell guitar pedals...  So, to help you (and me) I have added a few of the hi-end or boutique synth suppliers in the 'Links' section for your to browse.


Grand Designs (TV Programme)


Mxpand (Overlays) 

Synthstrom Deluge

Donner DT-1

Benn Jordan

LEDs (lighting)

LEDs (indicators) etc...

Hi-end or boutique synth suppliers, a biased non-comprehensive list:

Signal Sounds (Glasgow, Scotland) (Bastl, Doepfer, Motas, 1010, E-RM, Cyclone, Polyend, Squarp, Twisted Electrons...)

Rubadub (Glasgow, Scotland) (Soma Labs, AVP SYnth, Modor, Dreadbox, Studio Electronics...)

Elevator Sound (Bristol, England) (Critter & Guitari Organelle, Make Noise,  Erica Synths...) 

Synthstrom Audio (New Zealand) (The only source of the fabled Deluge

Expressive E (France) (Osmose, Touche, etc.)

Haken Audio (Illinois, USA) (The Continuum Fingerboard! (Plus the synth inside the Osmose!) and more...)

Black Corporation (Japan) (Deckard's Dream in Rack or Modular forms, plus more goodies...)

MFB (Berlin, Germany) (Tanzibar, Synth Pro, Dominion-1...)

Isla Instruments (Florida, USA) (The S2400 (wow!), Kordbot...)

(I'm happy to consider adding more entries to this list - just contact me!)


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Tuesday, 29 September 2020

Selecting different effect paths in an Ableton Live MIDI Effects Rack...

The 'Racks' in Ableton Live are very neat solutions that contain a lot more functionality than you might think at first glance. The obvious application for the Audio or MIDI Racks is to put several effects in series into a rack, and then assign Macro Controls to specific important parameters in those effects, and then hide the effects leaving just the macro panel visible.

A simple example first...

Here's exactly this process for a very common set of MIDI Effects. 

The first effect is the 'Random' MIDI effect, which is being used to add an offset of -12 or +12 semitones to the incoming MIDI note number, with a 50% probability (so half of the time, the incoming note will be unchanged, and half of the time it will be transposed up or down by 1 octave). 

This is followed by the 'Velocity' MIDI effect, which is adding a random number (up to 29) to the MIDI velocity of the incoming MIDI note. 

The overall result of these two MIDI effects is to produce random inversions of chords (or transpositions of individual notes), with random 'voicing' (relative differences in timbre or volume caused by the velocity). This is very useful for string or synth pads, making it sound like the part, and the playing of that part, is much more complex than the contents of the source clip! 

To store these two MIDI effects as a single device, you just insert a blank MIDI Effect Rack into the track strip:

The MIDI Effect Rack will initially have no effects in it (just a grey bit in the middle), so you just add the two effects to the empty 'Drop MIDI Effects Here' area in the middle.:

Which gives us a neatly encapsulated effects 'chain' that can be named and saved, and re-used whenever pad 'sweetening' is needed. 

On the left hand side, there are some button that control which parts of the MIDI Effect Rack are visible. If we turn on the top button:

...then the left hand side opens up and we see the Macro Controls:

The Macro Controls have all been mapped here. So let's look at how that works. Some of the controls in the effects on the right hand side have a small green dot, and this means that they are actually controlled by one of the Macro Controls. 

All that you do ito set this up s click on the control on the right hand side (four little corners appear around the control) and then right-click on it. A pop-up menu appears, and you just select the appropriate Macro Control:

Because the mapping to the Macro Control is already set up, the highlighted option here says: 'Unpam from Chance'. If you were doing this from scratch, then it would say 'Map to Macro 1', much like some of the other menu options shown.

Once you have mapped the effects controls to Macro Controls, you can colour and name the Macros. If  you click on the middle button on the left hand side, you will see this:

The middle button unhides the middle section of the window, and it contains some buttons, then a large box called 'Chain', which represents the chain of effects over to the right, and then another empty grey space with 'Drop MIDI Effect Here' - just like where we added the two MIDI Effects right at the beginning of this blog post. Clicking on the middle button again will hide this centre section.

Finally, you can click on the lowest little button on the left hand side, and the effects will be hidden:

(You can see that the lowest button on the left hand side is no longer yellow!). 

And that's it! Two MIDI Effects turned into a useful composite effect that you can grab from Live's browser and insert on any pad track that needs inversions and velocity processing.

There's more...

But there are more advanced ways to use Racks, and this is where people seem to have problems. The metaphors used in the Racks appear to have been strongly influenced by the way that samplers manage samples: velocity switching ('Vel'), and the note range that a sample is assigned to ('Key'). But the 'Chain' button uses a different mind-set - kind of like multiple Racks that you can switch between. 

Now I have already used the word 'chain' to describe those two MIDI effects placed one after the other in the Rack - 'in series' is how techies describe this arrangement. But MIDI Effect Racks allow you to have more than one chain inside them...

Remember how the middle section had a 'Drop MIDI Effects Here' underneath the large 'Chain' box? Let's see what happens when we do that...

I'm not going to repeat the previous effects. Instead let's start by just making some chords:

This chain of two MIDI Effects uses the 'Chord' MIDI effect to generate a triad of three notes to make a Major chord based on the root note input from the Clip, followed by the same 'inversion generator' effect as before, using the 'Random' MIDI Effect. The 'Random' MIDI Effect is polyphonic, so when you pass the three note from the chord through it, it will apply the random inversion to each of the three notes separately. After the Rack, there's one of my MaxForLive note monitors, showing an inverted C Major chord, and then a Collision Instrument to make a sound.  

If we now click on that middle button, the middle section unhides, and we get the 'Chain' box with the 'Drop MIDI Effects Here' grey area underneath:

I have renamed 'Chain' to 'Major' because that box represents those two MIDI Effects that are producing inverted Major chords.

Next let's alter one of the MIDI Effects:

Above is another chain showing the same two MIDI Effects, but this time with the 'Chord' MIDI Effect set so that it produces a Minor chord. If we drop these two MIDI Effects into the grey 'Drop MIDI Effects Here' area, then we get this:

A second large box appears underneath the 'Major' box that used to be called just 'Chain'. I have renamed this new box 'Minor' because it contains the 'Chord' MIDI Effect that creates minor chords. I have also mapped across the Chance and Sign controls to Macro Controls on the left, and aded some colour. I have also added a 'Chord Maj Min' control because that seems like what we want to end up with... but what do we map to it?

Before we look at that, let's see what is happening inside this MIDI Effect Rack... 

In the above diagram, I have shown the two large chain boxes (Major and Minor) and how they map to the two chains of MIDI Effects. You don't actually ever see a view like this, but it shows how there are two chains in parallel.

What do those two chains do, and what are those controls to the right of the large boxes with 'Major' and 'Minor' in? There seems to be a loudspeaker button, a 'Solo' button, a 'Refresh' button, and a bar-graph like the ones in MIDI tracks. If the Major chain is soloed:

Then the output of the Rack is Major chords.

If the Minor chain is soloed:

Then the output of the Rack is Minor chords.

So only one chain can be active at any one time, which makes sense! But how do we control these two chains. Mapping the solo buttons doesn't work, of course!

The answer lies in the 'Chain' button. As I noted earlier, the 'Key' and 'Vel' buttons seem to be influenced by samplers, since in other Racks they deal with key ranges and velocity switching, but clicking on the 'Chain' button opens a similar looking section to the ones for 'Key' and 'Vel'...

Now it isn't immediately obvious what to do, but it looks a lot like the 'Key' range control, so if we assign the two ranges to 'Major' and 'Minor' then it looks like this:

Unlike the 'Key' section, the numbers at the top go from 0 to 127, which looks like a MIDI range, and there's a light blue highlighted box for the '0' (zero) position. If you right-click on this box then you get  a pop-up:

'Chain Selection Filters MIDI Ctrl' tells us that this is a control for that 0-127 range. Also notice the 'Distribute Ranges Equally' - you can use this to set the ranges like this if you want... If we map this to that pre-defined Macro Control (Macro 5), then we get a result that looks like this:

Here are the mappings that have been added:

So the 0-127 horizontal bar in the 'Chain' view is for a 'Chain Selector', which means that when the value is from 0-63, then the Major MIDI Effect chain is active (soloed), and when the value is from 64-127 then the Minor MIDI Effect chain is active (soloed!).

So now we have a MIDI Effect rack which produces Major or Minor chords, with random inversions. But how can we automate the Major/Minor selection? You could map an LFO to Macro 5, but another way is to use a Clip Envelope (My Favourite!):

This Clip Envelope is mapped to 'Chord Major Minor', which is the mapping name for Macro 5, that controls the Chain Selector. So this clip envelope plays a Major chord first, then a Minor chord. You can, of course, change the clip envelope to suit when you want the chord type...

More chains!

Adding more chains just requires more dropping of MIDI Effects in the 'Drop MIDI Effect Here' area, and the setting of separate ranges in the 'Chain' view. Here's an example I prepared earlier:

Again, this isn't what you would actually see on screen!


Downloads for the example Racks featured in this blog post (plus previous Racks) are available here.


If you find my writing helpful, informative or entertaining, then please consider visiting this link:

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