Subtractive synthesis is easily the most important type of synthesis you will come across. Why? Because everything that falls under the subtractive umbrella term has gone on to directly influence other types of synthesis such as additive, wavetable, frequency modulation and nearly anything else you can think of.
The term is descriptive as it’s about sculpting sounds from harmonically rich waveforms using filters, envelopes, LFOs and amplitude shaping, and it’s the sound of quite literally thousands of records dating from the sixties right up until now.
Let’s start out by having a look at the front panel of a few common analog synths.
Where better to start than with where it all began – the Minimoog. Released in 1971, this was the first commercially available synthesiser that boasted portability. Previously, synthesisers were huge modular systems confined to the studios of scientists at the BBC Radiophonic Workshop. The Minimoog changed all that – finally keyboardists could compete with their noisy guitar-playing neighbours:
Can’t afford the thousands these go for on eBay now? Not to worry, Arturia make a nifty emulation, the Mini V, which we’re going to look at the front panel of:
I’ve highlighted two key features of subtractive synthesis in red and blue. Let’s start from the left: in red (or pinky red, whatever you want to call it) is our Oscillator Bank.
This section contains anything that generates a tone. We have three oscillators available to us. With oscillator one, we decide the range (synth-speak for octave, not enough time to go into exactly why they use this at the moment) and the waveform (more on these later).
Oscillators two and three can be tuned relative to oscillator one and we can also choose their waveform which, like oscillator one, have triangle, two types of sawtooth and three types of pulse wave.
Next, let’s look at our blue section, which the Moog calls Modifiers. This is the filter stage that uses the famous Moog Ladder Low-Pass Filter. We have cutoff frequency (where in the frequency spectrum our filter begins to do its job) filter emphasis (which you might be more familiar with as resonance or Q) and some envelope function, which we’ll get on to later.
With just these two sections, we can get most of the sounds we want out of the Minimoog – it really is that simple. All synths have a similar signal path – oscillators into a filter into the amp section – so it’s easy to navigate them once you know what you’re looking for.
In 1982, the Japanese company Roland released the now legendary SH-101. A staple workhorse synth in many a studio, the 101 is a monophonic synth known for it’s phat basses and squelchy acid leads:
Costing you around £500 for one in good nick, it’s not an unobtainable piece of gear but the guys at TAL made a free emulation (that sadly isn’t supported anymore but you can still get from their website).
Like our Minimoog, I’ve highlighted the oscillator section in red and the filter section in blue. Our oscillator section is named VCO, which you might come across in lots of analog synths. VCO stands for voltage-controlled oscillator as many of these machines were pre-MIDI (more on controlled voltage another time).
Our VCO section allows control of range, pitch modulation, the width of our pulse wave and how it can be modulated. Just to the right, highlighted in white is the Source Mixer where we can blend in our other oscillator waveforms: a sawtooth, sub-oscillator (tuned down one or two octaves) and white noise.
The filter section is highlighted in blue as and is named VCF. The quicker among you might have ascertained this stands for voltage-controlled filter. We have the familiar frequency and resonance controls (resonance is amplification of the filter cutoff position, giving a sharp, piercing squeal to the filter at higher values) and Env, Mod and Kbd modulation sliders.
Next, let’s look at the green highlighted section, ENV, our envelope generator. This can be routed to the amplifier and/or the filter, allowing us to shape our sound over time. I’ll cover envelopes in more detail further down the page.
Finally, there’s our Modulator, highlighted in yellow. This is what we may be more familiar with as an LFO, which stands for low-frequency oscillator. Here we can change the rate (frequency/speed) and waveform (sine, triangle, saw, pulse, sample and hold and white noise). We’ll cover most of these terms later on.
By now, you should get getting the idea, so I’ll skip through our last two examples a bit quicker. Next we’ll look at the Korg MS-20, a semi-modular monophonic bassline beast produced from 1978 to 1983 and responsible for Mr. Ozio’s Flat Beat.
The modular aspects of this will be covered another time but let’s skim through the interface. Our oscillator section allows for triangle, sawthooth, pulse and white noise waveforms, with oscillator two having an external input section for processing sounds from the outside world through the filters.
Our filter section in blue has two filters, a high-pass and low-pass filter. The peak rotary is what we know as resonance or Q. Underneath there are modulation options from either the LFOs or envelope generators.
The envelopes are in green: envelope one is a simple AR (attack and release, no sustain stage); and envelope two is a slightly more complex AHDSR (attack, hold, decay, sustain and release). Finally, our LFO (in yellow) has a control to morph between sawtooth, triangle and reverse sawtooth or varying degrees of pulse waves.
The last synth we’ll look at is the monster that is the ARP 2600, another monophonic semi-modular that was in production from 1971 to 1980.
Like the Minimoog, this is emulated by Arturia, and that’s what we’ll have a look at below:
The interface can be a bit disorientating compared to synths with a simpler layout. I’ve highlighted the oscillator, filter, envelope and mixer section – by now this should be ample information to acquaint yourself with it – but don’t worry if it takes longer because half the fun of the 2600 is stumbling across bizarre sci-fi sounds made by haphazardly turning things up and down.
Katrina and the Waveforms
So we’ve mentioned in some detail a number of waveforms commonly found in oscillators. I’m going to concentrate on five: sine, triangle, pulse, sawtooth and white noise. Let’s start with the sine wave.
This is the most basic waveform available to us – it’s just a single frequency with no overtones or harmonics. Overtones are frequencies which appear above the fundamental (the note you play, or the lowest loudest note); harmonics are integer multiples of that fundamental, though we’ll cover this in another article.
I’m going to be using Voxengo SPAN, which is a great free spectral analyser, to monitor overtones. In addition I’m going to use an oscilloscope called s(M)exoscope (which is sadly 32-bit only) but is also a free download.
Here’s what our sine wave looks like:
A simple curved waveform which you might recognise from maths or physics (if you remember anything at all from them, of which I remember very little). Great for sub bass, g-funk style leads and pseudo-organ pads.
Notice it’s just a single frequency with no harmonics or overtones above it. The note I’m playing is A at 110Hz. Hz is cycles per second, so a cycle will be completed one hundred and ten times per second to sound that pitch.
Let’s have a look at the triangle wave:
As the name suggests, a triangle shaped waveform. Great for bass and leads as it’s a bit rougher than a sine. Can sound a bit computer-gamey to some.
I’m playing the same note as in the first examples but we can see other frequencies above the 110Hz. In-fact we can see 330Hz, 550Hz, 770Hz, 990Hz etc. Triangle waves contain the odd harmonics (remember harmonics are integer multiples of the fundamental).
Let’s move on to our pulse:
A pulse wave can be square (as above); this has a ratio of positive:negative of 50:50 and adjusting the balance of this is known as pulse-width modulation (PWM). Great for basses at more symmetrical values; the more the width is offset, the thinner the sound becomes.
Adjusting the balance a little:
Adjusting the balance further still:
You might notice that this has the same overtones present as the triangle (though we can see the ones in the upper frequencies more clearly), the only difference being that these are a lot louder: the second harmonic (330Hz) is almost the same volume as the fundamental, and the third (550Hz) nearly as loud again, and so on.
Next we’ll look at our sawtooth:
Imaginatively named as it looks like the teeth of a saw. Good for rich pads, lead lines, distorted DnB basses.
This contains all of the harmonics, i.e 110, 220, 330, 440, 550, 660 etc This is what makes the sawtooth good for richer, more complex patches.
Finally, let’s have a look at the white noise waveform:
A very complex looking waveform, as close to random as a computer can handle. White noise is every frequency at equal amplitude (as we can see from SPAN below). This makes it useful for synthesising hi-hats, snares and risers or transitional effects in dance music. It can also be a useful layer in a bass or pad patch to dirty it up a bit.
The next stage of our subtractive synth is the filter. I’m going to explain the basics of low-pass, high-pass and band-pass filters, but be aware there are other types such as notch, comb, formant etc. Most subtractives have at least a low-pass filter.
Low-pass filters work by only allowing frequencies below the filter cutoff to pass through. I’m going to use the Novation BassStation plug-in for our source sound, which will be a sawtooth. Here’s the unfiltered sound:
Here’s what the SPAN looks like (I’ve changed the colour to white so we can overlay the filtered sound on top afterwards):
I’m using Live’s Auto Filter for this demonstration, but really any multi-mode filter will do. I’ve brought our low-pass filter down to 1.5kHz – have a look at the frequencies we’re left with (which will be coloured red):
As we can see (and hear), frequencies above the cutoff become attenuated. The amount of attenuation is defined by our slope, typically measured in dB/octave (6, 12, 24, 32). The higher the dB/oct, the stronger the filter will sound. Now I’m going to drop the filter down to 250Hz:
Again, we can see the red (filter sounded) super-imposed over the white (original sound). Although we are hearing frequencies as high as 600Hz they will be very quiet, and it’s that response that gives this particular filter it’s characteristic sound. Before moving on to high-pass filters, I’m going to increase the resonance to maximum and move the cutoff to 400Hz:
The resonant point is now louder that our fundamental, giving the synth a new tone. Hear the cutoff being swept with this resonance value? This is what’s called self-oscillation, where the resonance is so high our ear perceives it as a new tone:
Now let’s look at a high-pass filter with the same source sound and staying with 400Hz:
Now, only frequencies above the cutoff are allowed to pass. One more example before moving on, this time at 2kHz:
High-pass filters are obviously not a great idea to use on bass sounds but work nicely on pads or leads as filtering out the bottom end clears space for other instruments. High-pass filters (also known as low-cut) are a key ingredient to EQing a good mix, too.
A band-pass filter is like a high-pass glued to a low-pass, isolating a frequency spectrum in the middle of the sound. Here’s a band-pass filter at 300Hz:
And here’s one being swept from 100Hz right up to 14kHz. At lower frequencies it could be mistaken for a low-pass filter and right at the top we’re left with just the upper frequencies of our sound:
With these waveforms and filter types we can do most anything, though having them static or manually affecting them could be tedious. Envelopes and LFOs are built in modulation sources that can do some of the work for us. Let’s first look at LFOs.
As mentioned previously, LFO stands for Low-Frequency Oscillator which means that it’s like an ordinary oscillator but operates at sub-sonic ranges (typically below 20Hz, although some LFOs can run into the audio rate – that’s a topic for another day).
LFOs are free running, meaning that by default they are not re-triggered by gate information, although some synths do allow this option. It’s also noteworthy is that they contain both positive and negative information, so they oscillate above and below zero.
What does this mean? Well if, for example, we modulated our oscillator’s pitch by an LFO, the pitch would increase and decrease either side of the original pitch. So if our original pitch was C3 and we applied two octaves if modulation, the pitch would increase to C4 and down to C2.
The most common parameters LFOs are assigned to are pitch (vibrato at lower amounts, laser sounds at higher), pulse symmetry (pulse-width-modulation), filter (typical dubstep wobble) and amplitude (also known as tremolo).
There are two key controls on an LFO – the rate and amount. Here’s an example of an LFO modulating a sine wave’s pitch (we’re increasing the amount of modulation):
Here’s the same LFO but this time I am increasing the rate of the LFO:
Let’s modulate the symmetry of a pulse wave. Observe how the sound thins as the LFO cycles. At faster rates you can create the classic Reese bass sound:
Now, let’s add a low-pass filter and modulate the cut-off of that with our LFO. I’ve changed our oscillator to a sawtooth now and I’m adjusting the cutoff manually, too – watch how it affects the oscilloscope:
Finally, here’s the same saw, however this time we’ll modulate the amplitude:
The last thing we’re going to look at is envelopes. Envelopes have two key differences from LFOs. Firstly, they only contain positive data so if we we were to modulate the pitch of an oscillator with an envelope, the pitch would only rise past the start position (whereas an LFO would swing above and below it).
Secondly, envelopes are gate-triggered. They require a Note On signal to be triggered. Ordinary envelopes contain four stages – attack, decay, sustain and release – commonly referred to as an ADSR envelope.
Every synth will have at least one ADSR hardwired to the amplitude, and every time you press a key you engage this ADSR. This is just simple example but envelopes can vary in complexity, especially synths such as the Native Instruments’ FM8 – more on this another day.
The attack stage is the amount of time the envelope takes to reach the maximum value after note on information is received so, if the attack time is 1ms, then the envelope will engage instantly. If it’s 250ms, it will fade in over that time. Decay is the amount of time the envelope will take to reach the sustain stage after the maximum value is reached.
Sustain is different to attack, decay and release in that it isn’t a time duration but the value a note will sustain until a note off message is received. So if the sustain is zero, the sound will tail off to nothing after the decay time has elapsed. If the sustain is maximum, the decay stage will have no effect.
Release is amount of the time the envelope takes to reach zero after a note off message is received. No release means the sound will tail off very quickly whereas a longer release will make the sound fade out.
The simplest type of envelope is on/off, like a binary message (actually, some synths allow you to disengage envelopes all together and have the sound constantly sustain but nevermind about that for now).
An AD envelope is the next most simple type, there being virtually only three ADs you can have:
- No attack, some decay
- Some attack, some decay
- Some attack, no decay.
The x-axis is time in all three examples; attack and time are represented by blue and decay by yellow.
N.B sometimes, as in the case with the MS-20, you might see AR instead of AD – this has virtually the same application.
ADSR envelope are by far the most common type of envelope you will see. To visualise this, let’s look at the next diagram:
Aside from amplitude, envelopes can be wired to control pitch and filter cutoff. Let’s look at some examples of each of these. Firstly, we’ll start with an amplitude envelope with no attack or decay, full sustain and a little release:
This sound has no real movement to it so let’s add some attack and decay and drop the sustain to half its value:
The sound fades in, drops quite slowly, sustains and then fades out abruptly. Let’s increase the attack further and add more decay and release:
Now, I’ve reset the amp ADSR to it’s starting point. Let’s have a look at a second envelope, the pitch ADSR, I’ve added some decay, no attack, sustain or release:
The sound now has a sharp blast of high-frequency energy at the beginning. This sort of effect is useful when synthesising leads, pads and even drums. I’m going to increase the attack to match the decay value:
Now, the sound’s pitch rises and falls before settling at the original pitch determined by the sustain. It’s important to understand that sustain is a value so if we add some, the pitch doesn’t not fall to the note we’re playing but a note or two above it (depending on how much sustain was added).
Now let’s reset the pitch ADSR to its starting values and look at our filter ADSR. I’m using a resonant low-pass filter for this example. Using a short decay, we can make the synth almost sound like a ‘pluck’:
Finally, adding a lengthy attack with some sustain:
And there you have it, those are the foundations of basically every synth. If you can get your head around LFOs, ADSRs, waveforms and filters there’s not really much more to it.
Of course, different types of synthesis employ other techniques but, to give two examples, the way wavetable synthesis morphs between waveforms will likely be just LFO modulation and the way frequency modulation works is by using simple sine waves and envelopes. So it really is worth having a good grounding in subtractive before moving on to more complex types of synthesis.