Subtractive Synthesis
Subtractive Synthesis
• Osc - Oscillator.
• Kybd - Keyboard
• LPF - Low Pass Filter.
• HPF - High Pass Filter.
• Amp - Amplifier (not Ampere; the measure of electrical current).
• Env - Envelope.
• Mod - Modulation.
• LFO - Low Frequency Oscillator: usually used for modulation.
• Kybd.Trk. - Keyboard Tracking: aka KF (Keyboard Follow) or KS (Keyboard
Scaling).
• Vel. - Velocity : information on how hard an instrument note is struck.
• VCO or DCO - Voltage Controlled Oscillator or Digitally Controlled Oscillator
respectively.
• VCF or DCF - Voltage Controlled Filter or Digitally Controlled Filter respectively.
• VCA or DCA - Voltage Controlled Amplifier or Digitally Controlled Amplifier
respectively.
• ADSR - Attack, Decay, Sustain & Release: the typical components of an envelope
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In essence, this is subtractive synthesis. You start with a sound, subtract out the
unwanted bits and control its loudness over time.
There are many types of oscillators but the main "Oscillator" usually refers to a
"Keyboard-Controlled Oscillator". On a synthesizer, the Keyboard-Controlled Oscillator
may be labled as VCO, DCO, or sometimes as Waveform.
All this means is that when you play a note on the keyboard, the oscillator will generate
the waveform (or shape) at that intended pitch continuously (for as long as you hold
down that note on the keyboard). If you pressed "A4" (the note "A" just above middle
"C"), the waveform will be generated at a rate of 440Hz (ie 440 cycles a second). In
short, the keyboard controls the pitch of the oscillator (hence, keyboard controlled
oscillator)
A Waveform (or Wave) is a shape which the oscillator generates. The shape determines
the "timbre" or quality, characteristic or brightness of a sound. While pitch tells us
which note is being played, "timbre" tells us which instrument is being played. While
pitch is a basic frequency which identifys the note, "timbre" is made up of many other
frequencies or overtones which gives the instrument its overall character and identity.
WAVE CONCEPTS - The basic set of waveforms covered by this document are simple
mathematical shapes which are found on analogue synthesizers. The first waveform to
look at is the "Sine" wave. Although it is not always available on
• WAVE GRAPHS - When we graphically show a waveform cycle (eg Sine Wave), it
has two axes: Amplitude (vertical axis) and Time (horizontal axis).
Note - "Time" is very short. A wave cycle at 440Hz occurs in only 2.27
milliseconds.
Note - "Amplitude" is not necessarily in terms of perceived overall loudness but
it's relative changes over a cycle (ie a microscopic view) {Perceived loudness is
where we draw the graph 10 times taller (vertical axis) and say "Ooh! Yes, it is
louder now!"}.
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• HARMONIC SERIES - A Sine wave is the most basic waveform and is the building
block of harmonic analysis. This is because a Sine wave has no harmonics
(overtones) at all. It only has the tone of the fundamental frequency and no
timbre. The Fundamental frequency is the base or root frequency which we
identify as pitch (eg "A4" would be at 440Hz).
Since a Sine wave is pure (no harmonics), we can create other waveforms simply
by adding together any number of sine waves at different frequencies and
different volume levels (amplitudes). Mathematically, any sound can be created
using Sine waves at different frequencies and amplitudes. Similarly, any sound
can be broken down into discrete and distinct Sine waves at different
frequencies and amplitudes.
If we look at any static waveform (a pitched sound which doesn't change timbre
over time), it is made of sine waves which are Multiples of the Fundamental
frequency (F). This is known as the Natural Harmonic Series where the series
consists of F, Fx2, Fx3, Fx4, Fx5, Fx6 etc. If you analysed a bright violin sound
played at the note "A3" (where F.Freq = 220Hz), it is made of a series of
sinewaves at 220Hz, 440Hz, 660Hz, 880Hz, 1.1kHz, 1.32kHz, 1.54kHz and so on.
• SPECTRUM GRAPHS - We analyse waveforms using a Harmonic Spectrum (It's a
lot like the spectrum analyser on a hi-fi). The Harmonic Spectrum has Amplitude
(vertical axis) and Frequency of the sine waves (horizontal axis). To simplify
analysis, Frequency is usually expressed as multiples of the fundamental
frequency while Amplitude is usually expressed as relative to the fundamental
sine wave's amplitude.
WAVEFORMS - Let's look at the harmonic spectrum of the basic waveforms available
on subtractive synthesizers. The graph shows Waveforms (left) and Harmonic
Spectrum (right). Top to bottom: Tri, Square, Saw, Pulse.
• TRIANGLE Wave - Or Tri-wave. It sounds (and looks) a bit like the sine wave but
has some hollow-sounding quiet overtones. It is made up of only odd numbered
harmonics : F, 3F, 5F, 7F 9F etc. The amplitudes of the harmonic series decreases
exponentially.
• SQUARE Wave - It has very sharp corners, has a hollow sound with quite strong
overtones. It is made up of only odd numbered harmonics : F, 3F, 5F, 7F, 9F etc.
The amplitudes of harmonic series decreases steadily.
• SAW Wave - Also known as Sawtooth, is available as RampUp or RampDown
(which sound the same). It has a very bright and rich sound with strong
overtones. It is made up of all harmonics : F, 2F, 3F, 4F, 5F,6F etc. The ampitudes
of harmonic series decreases steadily.
• PULSE Wave - Also known as Rectangle wave. Its sound depends on how far the
Pulse Width deviates from the Square wave. The Pulse Width (period when the
wave us "up") is usually expressed as a percentage (of the full wave cycle), so a
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• Square wave is a 50% Pulse. A 10% Pulse and a 90% Pulse sound the same (to all
intents and purposes).
As the Pulse Width deviates from 50%, it sounds increasingly brighter and richer;
but as the Pulse Width becomes very narrow, it becomes more thin and nasal.
Generally, it is made up of all harmonics : F, 2F, 3F, 4F, 5F etc. The amplitudes of
harmonic series decreases but the odd and even numbered harmonics have
different amplitudes depending on the Pulse Width.
Note - Some amplitudes have alternating negative values which means inverted phase.
To the ear, it really doesn't make any difference.
Note - The fundamental is usually labled "F". The first harmonic is actually at 2F and the
second harmonic is at 3F etc. We usually refer to Odd-Numbered harmonics as being
the series F, 3F, 5F, 7F etc; and refer to Even-Numbered harmonics as being the series
2F, 4F, 6F, 8F etc.
Filters
The Filter section of a synthesizer may be labled as VCF (Voltage Controlled Filter) or
DCF (Digitally Controlled Filter). A Filter is used to remove frequencies from the
waveform so as to alter the timbre.
LOW PASS FILTERS - The main filter in subtractive synthesis is the Low-Pass Filter (LPF).
Low-Pass Filters allows the lower frequencies to pass through
unaffected and filters out (or blocks out) the higher frequencies. This is controlled by a
parameter labled as "Cut-Off Frequency" (or "Cut-Off" or "Frequency").
CUT-OFF - The Cut-Off Frequency determines the position in the harmonic spectrum
where the filter will begin to filter (or block) the higher harmonics (ie frequencies). If the
Cut-Off is set to a high position, then the timbre will be bright. If the Cut-Off is set to a
low position, then the timbre will be dull.
What does it sound like? Try this! Slowly say the words "Wow, wow, wow". This is what
an LPF sounds like as the Cut-Off position moves up the harmonic spectrum and back.
The first sound was "wuuu" which is fairly pure and dull. The next sound is "aaaa" which
is bright and full of harmonics.
FILTER RESPONSE - The way in which filters block out frequencies beyond the Cut-Off
position is not abrupt but is instead a gradual rate (which is more natural). So, the
further beyond the Cut-Off position, the more the subsequent frequencies are filtered.
This occurs up to a point where all further frequencies are completely filtered out.
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In Low-Pass Filters, the gradual rate of filtration (beyond the Cut-Off) is usually available
as 4-pole or 2-pole filters (which are 24dB/octave and 12dB/octave respectively). So a 4-
pole filter has steeper rate of filtration and will filter out more as compared to a 2-pole
filter.
The graphs show Filter Response with Amplitude (vertical axis) and Frequency
(horizontal axis). The horizontal dotted lines indicate the "unity gain" level, where the
timbre is unaffected. Below this dotted line, the harmonics are filtered (cut); and above
the line, the harmonics are boosted. Where the graph is at zero (horizontal axis), the
harmonics are completely filtered out.
Top graph = Low-Pass Filter with 4-pole and 2-pole response (4-pole is the steeper
sloped).
Bottom graph = LPF with Resonance and shows the filter response as Resonance is
increased.
HIGH-PASS FILTERS - Another common (but less used) filter in subtractive synthesis is
the High-Pass Filter (or HPF). A HPF does the opposite of a LPF. A High-Pass Filter allows
the higher frequencies to pass through unaffected and filters out (blocks-out) the lower
frequencies.
Top graphs show the filter responses. Left to right: Low-Pass Filter; Resonance; and,
High-Pass Filter. Cut-Off is indicated by vertical dotted lines.
Middle graphs - harmonic spectrums of the original waveform.
Bottom graphs - harmonic spectrums of the resultant waveforms (affected by the
filters).
Envelopes
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ENVELOPE PARAMETERS - Nowadays, there are many variations of envelopes available.
However, the most common envelope has 4 parameters: (1) Attack time, (2) Decay
time, (3) Sustain level, and (4) Release time. This is usually called ADSR Envelope.
Apart from controlling the Amplifier, envelopes can also be used to control other
sections of a synthesizer. But for now, let us just look at controlling the Amplifier.
Basically, an ASDR envelope only has 3 loudness (amplitude) levels; zero (off), full (the
maximum) and the Sustain level.
- Attack time it the time taken from zero to full.
- Decay time is the time taken from full to Sustain level.
- Release time is the time taken from Sustain level to zero.
Let us look at what happens to an ADSR envelope when a key is pressed on the
keyboard {Note - Before any key of pressed, the envelope is at zero and so there is no
sound}.
• ATTACK - Immediately upon pressing a key, the envelope "opens" from zero to
full. This is controlled by Attack time. An Attack time of zero means the envelope
goes from zero to full instantly (ie sharp attack). Increasing the Attack time
means that this will happen more slowly.
• DECAY - While the key is still pressed, the envelope then drops from full to the
Sustain level. This is controlled by Decay time. A Decay time of zero means the
envelope goes from full to sustain instantly. Increasing the Decay time means
that this will happen more slowly.
• SUSTAIN - While the key is still being pressed, the continuing envelope just
remains at the Sustain level. This is set by Sustain level.
• RELEASE - When the key is released, the envelope then drops from the Sustain
level to zero. This is controlled by Release time. A Release time of zero means
that this will happen instantly. Increasing the Release time means that this will
happen more slowly.
ASDR EXAMPLES - For amplifier envelope, the graphs show Amplitude (vertical axis) and
Time (horizontal axis).
Top - shows key-pressed and key-released information
(also called "Gate", equivalent to ADSR = 0%, NA, 100%, 0%).
Second ADSR = 0%, 80%, 50%, 60%. Perhaps for Piano.
OTHER ENVELOPE TYPES - Apart from the ADSR envelope, there are many other
variations but they are usually based on the same principles.
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One variant with 2 decays is the ADBDSR envelope; Attack, Decay-1, Break-Point, Decay-
2, Sustain, Release [where Decay-1 is from full to the Break-Point level; and Decay-2 is
from Break-Point level to Sustain level].
Another variant would be something like Rate-1, Level-1, Rate-2, Level-2, Rate-3, Sus,
Rate-4 where the envelope is very flexible (Rate is the inverse of Time). Rate = High (a
fast rate) is equivalent to Time = Low (a short time).
Modulation
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selection would be to use either Tri or Sine waves because of their regular rises and falls
without any sudden jumps (Square and Saw waves have sharp jumps).
• LFO ~MOD~ OSC.PITCH - Using an LFO to modulate the pitch of the keyboard-
controlled oscillator causes the oscillator's pitch to "wobble" with the LFO. Using
a Tri or Sine on the LFO will produce "vibrato" (a smooth up and down varying of
the pitch). Using a Square on the LFO will produce a "trill" (a sudden varying of
pitch). Using a Saw on the LFO will produce something like an alarm siren.
• LFO ~MOD~ FILTER CUT-OFF - Using the LFO to modulate the Cut-Off of the filter
causes the Cut-Off to vary with the LFO. Used in extremes, it will sound like a
"Wah-wah" sound.
• LFO ~MOD~ AMPLIFIER - Using the LFO to modulate the Amplifier causes the
loudness to vary with the LFO. Using a Tri or Sine on the LFO will produce
"tremolo" (smooth varying up and down of amplitude). Using a Square on the
LFO will produce a sharp "gated" sound (ie loud, quiet, loud, quiet etc).
KEYBOARD TRACKING - The keyboard not only controls the pitch of the oscillators but
can itself be used as Modulator. Keyboard Tracking (KT) is where the position of the
notes being played are used to influence other parameters. KT is sometimes labled as
Keyboard Follow or Keyboard Scaling. Keyboard Tracking can best be viewed as either
off or a positive amount or an negative amount.
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• VEL ~MOD~ AMP.ENV. - If Velocity was used to modulate the Amplifier Envelope,
then the harder struck kybd notes would result in louder notes (ie larger
amplitude envelope). This the most common setting for velocity.
• VEL ~MOD~ AMP.ENV.ATTACK TIME - If Velocity was used to modulate the
Attack Time of the Amplifier Envelope, then the harder struck kybd notes would
result in faster attack times.
• VEL ~MOD~ FILTER CUT-OFF - If Velocity was used to modulate the Filter Cut-Off,
then the harder struck kybd notes would result in overall brighter notes.
• VEL ~MOD~ FILTER CUT-OFF ENV. - If Velocity was used to modulate the
Envelope of the Filter Cut-Off, then the harder struck kybd notes would result in
making the filter envelope more pronounced. This usually means brighter notes
(unless inverted) but the effect would depend on the envelope shape.
Advanced Waveforms
control over the resultant harmonics. Below is the harmonic series for Square and Saw
wave mixed together at 100% and 75% respectively:-
HARMONIC F 2F 3F 4F 5F 6F 7F 8F 9F
Square 100% - 33.3% - 20% - 14.3% - 11.1%
Saw 75% 37.5% 25% 18.8% 15% 12.5% 10.7% 9.38% 8.33%
RESULT 100% 21.4% 33.3% 10.7% 20% 7.14% 14.3% 5.36% 11.1%
TWO DETUNED OSCILLATORS - Where two detuned oscillators are mixed together, the
result is still the sum of the waveforms. But this time, the oscillators are not
synchronised (ie not perfectly in tune).
The diagram shows the resultant waveforms. The top and middle waves are the two
individual detuned waves, while the bottom wave is the added result. The left shows 2
detuned Square waves added together and the right shows 2 detuned Tri-waves added
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together.
The resultant waveform is quite dynamic. It changes over time, due to the interference
of the detuning. This evolution of waveform over time gives a sense of movement to the
timbre because of the "phase" differences. Sometimes the two waves are in-phase (ie
both going up at the same time) and sometimes they are out-of-phase (ie one going up
and one going down). Mostly, it's something in between.
Note that, when they are in-phase, the resultant amplitude is strong (ie it is loud) and,
when it is out-of-phase, the resultant amplitude is weak (ie it is quiet). This will cause a
bit of "tremolo" or beating of amplitude (The greater the detuning, the faster the rate or
speed of the tremolo). The tremolo effect can be reduced by making one oscillator
louder than the other.
LFO ~MOD~ PULSE WIDTH - When a LFO is used to modulate the Pulse Width, then the
Pulse Width will change over time, which means the harmonic amplitudes vary too. This
is usually called Pulse Width Modulation or PWM (although not necessarily modulated
with a LFO).
As the LFO varies, so will the Pulse Width. The top left graph shows the original Square
wave, the middle left graph shows the modulating Tri-wave LFO, and the bottom left
graph shows the result. PWM looks and sounds a little bit like two detuned oscillators
and is an excellent way to create some movement in a timbre.
SYNC - Sync is where the cycle of one oscillator (the synchroniser) is used to synchronise
or "reset" the oscillator of another (the slave). The top right hand graph shows a Saw
wave before sync, the middle right graph shows the synchroniser (in this case a Square
wave at a lower frequency), and the bottom right graph shows the effect of
synchronisation. As the Square wave completes each cycle, the Saw wave is "reset" to
start again. This results in a new static wave whose frequency is that of the synchroniser
but whose shape is a portion of the original. Note that the choice of wave for the
synchroniser is irrelevant and only it's frequency matters.
RING MODULATION - Ring Modulators are circults which output the "Sum and
Difference" of its inputs. If the inputs are two Sine waves at 440Hz and 420Hz, the
output would be 860Hz (sum) and 20Hz (difference). Depending on the harmonic
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content of the input, the output can be very complex and it is generally used for gong-
like sounds.
With sampling and digitised waveforms, there are now more alternatives to the simple
mathematical waveforms. Let us look at how the subtractive synthesizer process applies
to these:
OSCILLATOR ---------> FILTER ---------> AMPLIFIER
If sampled or digitised waveforms are used, the only real difference is the waveform
used by the oscillators. The process is unchanged. The difference in approach would be
knowing the harmonic content of the waveforms. And the best way to learn this is to
experiment with the waveforms and let your ears be the judge. Things really haven't
changed that much.
DIGITISED WAVEFORMS - A digitised waveform (sometimes referred to as Pulse Code
Modulated [PCM] waveform) is a waveform stored in a memory chip. It may originate
from a sample (ie real sound) or it could be created. Typically, the waveforms stored
would be single cycles only. Put tons of these digitised waveforms on a memory chip
and you have the makings of a wave-table synthesizer.
The quality of the digitised waveform will depend on the Resolution and Rate (as well as
the inherent quality of the sound itself). Compact Disk quality uses 16-bit resolution and
44.1kHz sample rate.
Resolution of 16-bits means that the sound "amplitude" is captured and stored in levels
from 0~65,535 (2^16). On the waveform graphs, this would be the vertical axis. The
horizontal axis (time) is captured by the sample rate. A sample rate of 44.1kHz means
that 44,100 of the amplitude "levels" (0~65,535) are taken per second.
PHASE DISTORTION - This is a synthesis method popularised by Casio's CZ-101 and other
CZ-synths. Although it is not a subtractive synth, it can actually be treated as one. A CZ-
synth basically has 3 envelopes for DCO (pitch), DCW (distortion), and, DCA (amplifier).
Phase Distortion relies on "reading" the digitised waveform in a non-linear way.
Normally, when you play a digitised waveform, the processor "reads" the data at the
intended rate (eg 44.1kHz). A Phase Distorted "reading" would be where the part of the
data is read at one rate and the remainder at another rate (eg first half at 22kHz and the
second half at 66.2kHz [average = 44.1kHz]). This distorted data-reading in effect
changes the waveform. In theory, you get a brand new waveform. In practice, it sounds
a lot like a LPF at work.
OTHER SYNTHESIS ENGINES - In a lot of cases, the principles of subtractive synthesis will
still apply. Many new synthesizers will still include a Filter of some sort because it is a
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simple and easy method of controlling timbre. However, this article has not been about
filters only. When encountering a new synthesis engine, the principles of Waveforms,
Envelopes and Modulation will still apply. With new synths, the challenge will be
learning the routing and modulation processes available. But this is true of the older
analogue synths too. Learning to control a synth has much to do with familiarisation of
the synth layout and how it works. There are no short-cuts. In the end, it's still about
knowledge, experimentation and imagination.
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