there are three audio driver standards that you are likely to come across: asio (audio stream

input/output) wdm (windows driver

model) and core audio (which is for apple computers only)

aliasing is most prominent when a sample is played above the pitch at which it was sampled; this can be significant in many
synthesizers where the sound source is a sampled waveform

to understand aliasing, you need to understand the nyquist point; the nyquist point is one-half of the sample rate; so, if the sampling
rate is 22,050 hz (which is quite a low rate), then the nyquist point will be at just over 11,000 hz, which is well within the audio
spectrum; any sound (or element of a sound) that has its frequency (pitch) above the nyquist point will lead to digital distortion or
aliasing; with the sample rate set at 44,100 hz (in other words, the usual rate for cds), the nyquist point is 22,050 hz, which is above
the threshold for human hearing; therefore, you can filter out any sounds that may lead to aliasing without affecting the sound that is
heard; well, theoretically you can, because in practice, sometimes sounds above the threshold of human hearing can have an effect
on sounds below the threshold; however, you will usually need highly sensitive ears to notice this effect

each synthesizer has an audio path, a control path and a modulation path

reverse sawtooth is also called ramp

white noise is brighter, the pink noise is darker

if you hold a note and drag the cutoff frequency slider the sound will get duller (or less bright); not only does the tone change, but
the characteristic of the sound changes, too; the sound is far less aggressive and could perhaps be regarded as being warmer; the
sound gets progressively quieter as more of the waveform is filtered out; toward the bottom of the range, you may have difficulty
hearing any sound

resonance boosts sound at cutoff frequency

self-oscillation happens when a filter starts creating sound on its own rather than simply filtering the sound

a source modulates a destination

filter env
attack governs the time it takes for the cutoff frequency to open fully after a note is triggered
decay controls how quickly the filter cutoff drops (to the level set by the sustain level) after it has reached its maximum
frequency
sustain cutoff frequency while a key is held; this level stays constant until the key is released
release time it takes the filter to close after a key is released

filter sweep happens when cutoff is swept (or changed) as the note is held

acoustic instruments exhibit a behavior where, as higher notes are played, the sound becomes brighter

as you increase the filter key tracking , the filter’s cutoff frequency will automatically open as you play higher notes; the key tracking
function will only have effect if the filter cutoff frequency is set at less than its maximum amount

the output level in any particular patch depends on many factors, including: the number of notes being played; the selected
waveform; the amount of filtering

audio path
sound sources>filters>amplifiers

control elements allow you to access and directly control certain parameters (er. cut-off)

modulation elements affect the audio path modules but that don’t create any sound in their own right

midi has 127 levels of loudness; this range is called velocity

if you look at an acoustic piano, the loudness of a note affects volume, tone and sustain; pitch affects tone and sustain

over time: the sound will get duller (however bright it may have been initially) and the volume will decay until it reaches zero

velocity is often used to control loudness and tone (by modulating the filter cut-off)

key tracking is the process by which changes are made according to the pitch of a note

a pitch bend wheel can generate positive and negative output; a modulation wheel only generates positive output

if your keyboard has aftertouch, the pressure on the keyboard while notes are sustained can be used to control your patch

sine wave is the purest form of tone; it consists of the fundamental note and has no overtones
if you run a sine wave through a filter, there are no overtones to filter out and therefore, the only effect that a filter would have would
be to reduce the volume; if you put any sound through a low-pass filter, as you take out the harmonics it will tend to sound like a sine
wave

sine wave is often used to thicken up patches (eg. to give a subsonic, foundation-shaking quality to bass) or for fm synthesis

sawtooth wave has a bright sound that is often used as the basis for brass and string sounds, as well as general fat synthesizer
sounds (such as stabs and basses); the wave also tends to dominate a broad proportion of the sound spectrum

pwm is generally used to change the tone of a waveform, or to fatten a sound (eg. when two waves are modulated in different ways
and combined)

triangle wave is often used as a low-frequency oscillator (lfo) waveform

when we talk about the phase of an oscillator, we generally mean the position in a wave’s cycle; with one oscillator the phase
matters little; with two oscillators, the effect of phase can be dramatic

with hard sync, one oscillator is slaved to another oscillator; the slave oscillator will restart its phase each time the master oscillator
restarts its phase

with doubled oscs, you combine two (or more) oscillators with identical waveshapes; with combined oscillators, you combine two (or
more) oscillators with different waveshapes

by combining waves, and controlling their relative phase, polarity and levels, you can create completely new sounds before you get
to the filter or other audio processing tools

many synthesizers use sampled single-cycle waves as oscillators

the loop of a sample is called a sustain loop, since it sounds for the sustain portion of the sample

in subtractive synthesis you take a sound source that is rich in harmonic content and reduce those harmonics with a filter

in additive synthesis a group of sine waves with different frequencies are combined together; the sine waves (called partials or
harmonics) are multiples of the fundamental frequency; the advantage of this approach is that each partial can be individually
controlled

a square wave is made of odd-numbered harmonics with decreasing amplitudes in the ratio 1 /n (where n is the number of that
harmonic); the first harmonic (the fundamental) has its full amplitude, the third harmonic has an amplitude of one-third of the
maximum, the fifth harmonic, has an amplitude of one-fifth, and so on

a sawtooth wave has both odd-numbered and even-numbered harmonics with amplitudes decreasing in the ratio 1/n;

a triangle wave has odd harmonics with decreasing amplitudes in the ratio 1/n2

frequency modulation synthesis, phase modulation synthesis and phase distortion synthesis are all based on frequency modulation

instead of having oscillators, fm synthesizers has operators; to create an fm sound, you need two operators, a modulator and a
carrier: the modulator modulates the frequency of the carrier, the carrier is connected to the output and is therefore heard

modulators and carriers can be doubled to thicken up a sound in the same way that oscillators are doubled in a subtractive
synthesizer

parallel carriers one modulator drives two or more carriers

parallel modulators two or more modulators drive one carrier
cascade one modulator drives another modulator, which itself then drives the carrier

in fm synthesis, sounds can be shaped without the need for a filter

two key aspects affect sound: the frequency of the modulator relative to the carrier (the higher the frequency, the brighter and more
metallic the sound) and the amount of modulation (controlled by the output level of the modulator)

to achieve the constant shifts and design the desired sound, the pitch relationship between each operator is fixed

the level of each operator is dynamic and will vary according to a range of factors, including: initial algorithm,
time (as the note decays, the effect of the modulator decays) and velocity

if the carrier and the modulator have related frequencies (generally, if their frequencies are integer multiples) the resulting sound will
tend to give a far more useful and cohesive tone
1:2 ratio modulator is 12 semitones higher than the carrier
1:3 ratio modulator is 19 semitones higher than the carrier (an octave and a fifth)
1:4 ratio modulator is 24 semitones higher than the carrier

ring modulation is the effect of combining the sum and difference between two waveforms frequencies

some filters do cut the signal even when fully open

some sound designers also use high-pass filters with resonance to boost the fundamental tone of a patch; you can do this by tuning
the resonance to the fundamental frequency of a note and engaging key tracking so that the resonance boost then follows the
played note

band-pass filters tend to take a lot of energy out of the signal, so you may have to boost the level after the filter

notch (band-reject)

if you’re trying to mix two sounds and they don’t sit well together, it may be that they’re both trying to operate in the same frequency
range; in this case, you can notch out one of the sounds to allow the other to sit properly; you can do this in your patch design or, as
many mixing engineers do, with eq

a peaking (or peak) filter boosts sound

formant filters are usually used to emulate vowel sounds

comb filters work by adding a slightly delayed version of a signal to itself; this causes phase cancellations and can give a slightly
chorused or metallic type of sound; the spectrum produced by these filters looks like a comb, hence the name

some filters offer a drive option that can be used to overload the filter and get a warmer and richer sound

a 6-db/octave filter (sometimes called a 1-pole filter) will reduce the level of the sound by 6 db at one octave above the cutoff point;
a 12-db/octave filter (sometimes called a 2-pole filter) will reduce the level of the sound by 12 db at one octave above the cutoff
point

a 24-db filter could be a better starting point when designing bass sounds, whereas a 12-db filter produces better results with pads
or sample-based sounds

two 12-db filters in series are equivalent to a 24-db filter

using two filters allow for a broader range of sounds

dahdsr envelope adds delay time (period before the envelope begins) and hold time (after attack , the envelope remains at its
maximum level for the hold time); if you set the delay and hold controls to zero, the envelope will work as a conventional adsr
envelope

multi-stage envelopes have as many stages as needed; this increases flexibility, and allows to create rhythmic patterns

unipolar env only output positive values (or negative values, if inverted); bipolar env output positive and negative values during the
same cycle

when you load up samples, the samples will have their own volume envelope

an envelope follower module extracts the volume envelope shape from an incoming audio signal so that it can be used as a
modulation source

an lfo is generally used to modulate pitch (vibrato), to modulate volume (tremolo), to modulate filter cutoff (wah-wah) or for
gating/stuttering effects

a regular envelope is a single-shot modulator (it begins, it modulates, it sustains at the sustain level and then it ends), an lfo is
generally a single-cycle wave that repeats; a regular envelope is unipolar, an lfo is bipolar

sine and triangle waves are great for vibrato, tremolo, and wah-wah

hz = cycles per second

most lfos have a range of around 0.1hz to 20hz

amount setting will often control the size of the lfo wave, changing the depth of modulation

many lfos allow you to synchronize the phase to a key press and to then control the phase
free-running lfos do not have the consistent effect on a sound that key-synchronized lfos have

with a monophonic lfo, notes share the same lfo; with a polyphonic lfo, notes have their own lfo, which will start its cycle at a
different time from any other lfo

polyphonic lfo is great for pads, but less useful for rhythmic sounds

with a step generator, you can specify

steps number
steps time base (eg. 1 step = 1 sixteenth note)
steps lvl
steps length

like lfos, step-generator patterns can be looped; unlike (most) lfos, the played pattern can be precisely determined

in legato mode, the portamento/glide will occur only if the first note is still held when the second note is struck; the clear advantage
of legato mode is that it allows you to control the portamento/glide effect by your playing

often allied with legato mode is the non-retriggering of envelopes

glide time
fixed the glide will take the same time irrespective of the range of the glide
variable the time of the glide varies by the length of the range

a cmp can make a sound fatter and/or smoother, enhance the perceived loudness (without increasing its volume), add more punch

with parametric eq, you can set the frequency to be boosted or cut, the amount of boost or cut and the bandwidth of the boost or cut

a mono delay adds a simple delay to the input signal

a stereo delay has two separate delay lines, which may (or may not) have different delay times; conventionally the delay lines are
hardwired left and right

a cross delay has two channels; however, the output from the first delay is fed into the second and the output from the second is fed
into the first; this can give the effect of spreading the delays across the stereo spectrum or ping-ponging the delays between the two
channels

the modulation control allows you to change the pitch of the delayed signal

in modulation delay effects the dry signal is mixed with a delayed copy of it; the delay is so short that cannot be perceived as a
separate event and its pitch is modulated by a lfo

delay although the delay may be imperceptible, it has a significant effect on the sound, hence this control
feedback sets the number of times the delayed signal is repeated; this has an effect on the perception of depth
rate controls the speed of the low-frequency oscillator, which modulates the delayed signal
depth controls the extent to which the low-frequency oscillator modulates the pitch of the delayed signal
tone controls the tone of the delayed signal, taking away any sharpness to the sound and any metallic ringing
mix controls the balance between the dry and delayed signals

phasers, flangers and choruses are modulation delay effects

some reverb units act like delay units with short delay time; some are more complex and split the reverb into two parts, which you
can separately control: the initial early reflections the longer reverberation tails