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1. FREQUENCY

This is simply calculated at how many cycles (waves) occur every second. These

cycles are repeated so really we only need to look at how many cycles (waves)

occur in one second. The result is measured as cycles/second and this unit of

frequency is called a Hertz and the abbreviation is Hz. You cannot get simpler

than that...how many cycles hit you in one second. Heinrich Hertz was a dude who

worked with wavelengths and frequency, so we have to thank the man and it seemed

only right to name this little calculation after him. I always remember the rent-a-car

agency when I think of frequencies and Hertz and it makes me smile every time

so remembering that name is easy.

To give you an example of how easy this is check out the following:

If you had 50 cycles hit you in one sec then that would be a 50Hz wave. There,

simple and makes you look cool in the bar when you want to impress someone.or

maybe not.

So it also follows and makes complete sense that if you had 10,000 cycles per

second then that would be 10,000 Hz, but, because we don’t want to have to write

so many numbers every time a thousand appears we use the k letter to mean a thousand. So, 10,000 Hz is now written as 10 kHz. Now you look

even cooler. There is a reason we do this and it’s not because we want to look

deep and complicated individuals but simply because of all the work that has been

carried out on our hearing range in the past. And a range was formed, sure it

varies but generally speaking, our hearing range is anywhere from 20 Hz, deep,

to 20 kHz, high.

Now, let us think of that range and make life a lot easier by giving names you

recognise to the frequency range. So: bass, midrange and treble are easy to remember

and if you are old enough then that’s about all that used to exist on hi-fi systems

back in the days of armour and jousting. Now let us give those tags a frequency

range and then all becomes so much easier to understand.

Bass: 10 Hz to 200 Hz

Midrange or mid , a term you hear a lot of engineers use: 200 Hz to about 3 kHz
Treble: 3 kHz to whatever the highest value you can hear.

It is important to mention, at this stage, what frequencies most natural instruments

lie in. Not only will this help you later when it comes to equalisation and filtering

but it also gives you a nice tagging point for the instruments.

Kick drum....20-150Hz

Bass......20-250Hz

Piano.......80-4500Hz

Snare......100-200Hz

Cymbal... 300-600Hz

Bear in mind that these are only guides and nothing more as there are so many

different instruments with different characters and even the same instrument played

differently will generate a different frequency range.

So we now know that higher frequency sounds are higher in pitch as there are

more cycles per second and lower frequency sounds have fewer cycles per second.

Easy.

Right now I think it is important to show you a frequency chart for all the notes

on a keyboard or scale and the midi note numbers as well as this will come into

play at a later date when we deal with synthesis and programming with the use

of midi.

You do not need to learn this chart in parrot fashion but it is important to

understand some of the frequencies that are used as, later, you will need to know

these frequencies so that if you need to use equalisation or filters to shape

a sound or remove or add certain frequencies, then the chart can prove to be invaluable.
In most cases, you only need to recognise the main frequencies for certain notes.

For example: C4 at 261.63 Hz is a great reference point, because then you can

find, easily, C5 or C3 etc..

I cannot stress how important frequencies are for the understanding of sound

and synthesis. Engineers live by them as do producers and Sound Font developers.
If there is one piece of information that overrides any other in terms of importance

it is the understanding of frequencies. How often have you tried to mix your track

only to be mystified by the result? Terms like ’muddy’ or ’thin’ spring to mind

and these are all because the mixer or producer does not have an understanding

of frequencies and their effect on other frequencies in a mix.

Understand this basic concept and you will be armed with the most potent weapon.

The rest of synthesis is the understanding of how to shape these frequencies to

create new ones or to bring out the best in a spectrum of sound.

Waveforms and frequencies go hand in hand. Understand these two and the rest

is all about using the tools.

So, let’s get on with the CHART.

The Chart

Midi No Note Keyboard Freq

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As you can see from the funky chart that for every octave you go up you double

the frequency and it is the same in reverse for every octave that you go down,

in that case you halve the frequency. The other cool part of this chart, are the

midi numbers that go along with every note. These will become very useful later.

For now, you don’t need to worry about them too much.

Example: C4 is 261.63 Hz. To get to C5 we double the frequency so it is now 523.25

Hz. And if we wanted to go from C4 to C3, it would be 130.81 Hz. There, a few

secrets to throw about.

Now let us create the tag for this whole sound thing. I always imagine a wave

as a 3 dimensional entity and with that I attach colours and size. So, for a low

frequency wave I will think of it as a large and flowing wave with nice warm colours

like orange or deep red and the whole image is nice and slow. For higher frequencies

I use smaller and faster waves and in harder colours like bright yellow or striking

blue. This image is then enhanced further by having a person standing in front

of the waves, usually me, but my name is Hertz and I am listening to these waves

in a rent a car. Although this may now confirm the urgency for me to seek therapeutic

help, it is the best way for me to remember things. You can create whatever images

or story lines to the definitions in this tutorial. They are your images and must

work for you.

2. AMPLITUDE

Generally speaking this means the loudness or level of a sound or waveform . I prefer the word waveform for sound as it is the form or shape that the waves

take and the further we go into this tutorial the more that term will make sense

as waveforms vary in shape and character so, from now on, I want you to use the

word waveform for sound. It is better defined with a simple graph. In fact, now is as good

a time as any to introduce you to graphs. Enter fig2.

Fig2

As you can see, the waveform, it’s actually a sine wave but don’t worry about

that for now as that is the next subject we will cover, is 2 cycles and I have

arrowed in the second cycle, no difference which cycle I arrow as they are both

repeats, anyway I had to arrow the second cycle so as not to intrude on the amplitude

line in the first cycle. The height or peak of the waveform is the amplitude and

the length is measured as 2 cycles and this is done very simply. Imagine a sound

and how it starts. It starts from 0 then goes up, hangs about and then drops off.

In the diagram you can see the waveform starts at zero, goes up, drops to zero

then goes to the negative area and then climbs to zero again. This is using the

wave theory we defined earlier and all waveforms are represented like this, as

a graph, and how each cycle behaves or how a number of cycles behave in relation

to each other. For now you do not need to worry about complex waveforms and any

other factors regarding waveforms as we will deal with them as we go along, at

your pace, that way you do not feel as if there is too much information to learn.

This is meant to be fun so let’s keep it that way. Later we will look at wavelengths,

decibels, phase etc.. so for now we need to look at the most basic waveforms that

are found on synthesizers , what they are, how to draw them graphically and what

type of sound each waveform produces. This leads onto the next subject: WAVEFORMS

WAVEFORMS

The final most important component of sound is TIMBRE. This is what defines the tonal quality of a sound. A C4 note played on a piano

and at the same level as a C4 note played on a saxophone does not produce the

same sound or timbre. They are both the same level and both played at C4 but both

have distinctly different sounds or timbres. Timbres are made up of waveforms

and it is these waveforms that go to make up the tonal quality of a sound. This

is called timbre.

Although there are countless waveforms and some are very complicated in nature,

there are certain standard waveforms that are always seen on analogue synthesizers

or any modern day synthesizers, be it hardware or software, that have synthesis

capabilities.

They are as follows and are shown graphically and you can even listen (audition)

to them so you can familiarize yourself to the way they sound.

Saw at C4 (Right Click ’Save Target’)

Sine at C4 (Right Click ’Save Target’)

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Square at C4 (Right Click ’Save Target’)

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Triangle at C4 (Right Click ’Save Target’)

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Noise (Right Click ’Save Target’)

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These waveforms have their own sonic (sound) qualities and if you have a basic

understanding of what they sound like and what they are usually used for then

you are half way there to understanding how to manipulate them.

Sine waveforms are great for creating deep warm basses or smooth lead lines. They

can be used to create whistles, layered with kick drums to give that deep subby

effect. In fact the sine wave is a pure waveform and the harmonic content is fundamental.

That means that almost all other waveforms are created from sine waves. But don’t

worry about this for now as the theory will come later. All I want from you is

to understand and tag the above waveforms. I use the actual waveform shapes as

an indication to the type of sounds they can create.

The sine is a nice smooth flowing waveform.

Saw waveform, or sawtooths as they are more commonly known, have a rich and bright,

edgy sonic quality about them and are great for creating strings, brass, huge

Trance pads, searing leads and electro basses. Of course there is, as with all

the other waveforms, far more to it than that, but, as I said, I just want you

to get a general idea of what these waveforms are used for and how they sound.

The real fun starts when we start to layer them or trigger one with the other,

but that will come later when we get into synthesis.

For tagging, these waveforms have that jagged like shape so they are easy to

remember.

Triangle waveforms are great for bell type sounds or wind type sounds like flutes etc.and

I regularly use them for the FM type of sounds that you hear on Yamaha DX7s or

FM7s, great and very useful.

These waveforms look like triangles so that makes life easier.

Square waveforms are great for brass and deeper wind type of instruments and are usually

used along with other waveforms as they are quite strong and hard on their own.

But they are invaluable as are the rest listed above.

As you can see the square waveforms look like a bunch of squares with their tops

and bottoms missing and alternatively.

Noise waveforms are used more for effect than anything else but I find that they are

fantastic for creating pads along with other waveforms like saws and triangles.

You can also create great sea shore wave type of sounds or huge thunder or even

some great Hoover type sounds when used with saws. Endless what you can do with

these waveforms and for that reason alone you see that most synthesizers, software

or hardware, have these waveforms as the main sound source. The rest is all about

the actual synthesis or programming of these waveforms.

These are the easiest ones to remember, a mesh of what seems like radio static

or just rubbish.

SO, there you have it, your first week’s tutorial. A simple and basic explanation

of sound and waveforms, what their characteristics are, how we perceive them,

what they are used for and best of all ’ the tagging method ’.

Next week we will look at the different types of synthesis, a slightly more in

depth look at waveforms, what is ADSR and how we use it to shape a sound, what

are the components of synthesis and definitions for all the terminology and components

in synthesis along with a bunch of some very cool looking graphs. Remember that

the most important aspect of these tutorials is to tag every morsel of information so that you will have an easy way of remembering

and understanding all the data. Enjoy these tutorials and have some fun whilst

you learn. Oh, and don’t forget the German rent a car dude, he comes up all the

time in these tutorials.
dB: deciBel . The level of a sound or volume is measured using the dB scale. Deci being a

tenth and Bel being the unit. A deep and moving question at this point is: ‘Why

do we measure in tenths and not in single units?’ And the equally moving answer

is: Our ears can hear a vast number of audio levels and it would be a mathematical

nightmare to try to use the ‘actual’ numerical representations of audio levels,

so we use tenths to make it easy to understand and calculate. I am sure you have

seen this on your mixer, on software editors, on DAT players, just about everywhere

in the audio industry. In the next tutorial it will become clear why we need to

know this.

But for now………

ADSR

A : attack

D :decay

S :sustain

R :release

These are the components of the envelope of a waveform.

Envelope: generally this is the term used for the rise and fall in volume of a single note

over time. Think of it in terms of the shape of a sound and you will have a better

and clearer concept of what an envelope is. This shape determines what the note

or waveform will sound like and the above ADSR are the components that make up

that shape. Of course I am being very simplistic here but I want you use the tagging

and visualising techniques we used in Part1. I would also like to point out at

this point that envelopes are also used for many other areas of synthesis but

for this example we are looking at the amplitude envelope, volume over time.

The best way, as always, is to show you graphically how the ADSR of an instrument

looks and behaves. Fig 1 shows a standard ADSR envelope.

Fig1

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So, as you can see from my outstanding diagram that the note starts at zero,

has a quick attack and rises to it’s highest value, then has a small fast decay

( quick drop) down to about half way of the volume of the note, then stays there

(sustain) for some time then has a short smooth release.

As you can imagine, a pad sound would have a very long attack followed by a slow

decay then a long sustain and a very long release. This will give you that smooth,

slow build and nice tailing off.

An acoustic bass will have a quick attack for the plucked effect, with a short

decay, where the plucked sound drops to the sustained section which will be constant

as it is the body of the sound, and then a short release as acoustic basses are

‘fingered’ and to get that ‘let go’ effect the release needs to be relatively

quick. But on some acoustic basses the release can be longer to give that ‘let

go’ effect that stays around and decreases in volume slowly.

For synthesizer type of basses the attack is either quick, if snappy, or slightly

slower when dealing with the more throbbing type of bass sound that builds over

time, the decay will be short but slow and the sustain will be lengthy and close

to the maximum peak of the attack, the release will be either fast or slow depending

on the type of sound chosen and whether the user wants to be able to ‘roll’ from

one note into another without cutting out.

A drum sound, in fact most percussive sounds, will have a quick attack followed

by a short decay, very small sustain and an immediate release, easily recognisable.
Other ways of manipulating sounds by adjusting the ADSR is to take, for example,

a string sound that has a slow attack, slow decay with a long sustain and long

release. If you were to change the ADSR to have a quick or fast attack, a short

decay, a sustain that maximises to half the volume of the envelope and a very

quick release, then you have now created a stab type of effect, short notes that

can sound like the string is being plucked. All you have done is adjust the ADSR.

As you can see the ADSR is critical in how a sound behaves when a note is engaged

and these characteristics determine if the sound is plucked, pulled, slapped,

pushed etc…These four components shape the sound and determine it’s behaviour

over time.

A good way to experiment with the ADSR of any sound is to create different waveforms

using any software editor or hardware sampler with a graphic or waveform display.

That will familiarise you with the different types of envelopes for different

sounds. I use Sound Forge and in fact most of the examples on these tutorials

and screen captures (images) are all taken from Sound Forge. But use whatever

means you have at your disposal for creating different waveforms and spend some

time changing the attack times or lengthening the sustain points, basically alter

any of the ADSR components and you can then hear the differences these changes

make. Sound Forge has a nice little option whereby you can create waveforms using

the FM synthesis tool. By creating a waveform you can then see the nodes for each

ADSR component and you can move these around and have some fun, both visually

(eyes) and aurally (ears). The screen capture below illustrates this beautifully.

Those little squares you see are the ADSR nodes and by clicking and dragging those

nodes you can create new envelopes. You can see how similar it is to Fig1. Now

is that funky or is that funk

Okay, so now we have covered ADSR and glossed over a little about what an envelope

is. As this tutorial progresses, you will see how often we use the word envelope

and ADSR can also be applied to just about anything that can be manipulated. You

will often see the terms filter envelope or velocity envelope etc…..You will also

come across ADSR envelopes for filters on analogue synthesizers etc…

Below is a screen capture of Fruity Loops using a plugin vsti called Helga, manufactured

by a company called Kiesel.

If you look at the bottom of the Helga main page you will see the section called

AMP, there you will see the ADSR knobs. Take your eyes half way up and to the

right where it says FILTER and you will see the ADSR for the filter.

By the way, that is a very cool vsti. Just for reference, ‘VST’ stands for Virtual

Studio Technology and the ‘I’ stands for Instrument. However, don’t worry about

that for now as we will cover these two in detail at a later date.

As promised in the last tutorial, here is a list of the general terms and abbreviations

used in synthesis. For now, I just want you to familiarise yourself to the terminology.

I do not want you to learn it parrot fashion as that will just serve to confuse

you and, remember, this is a tutorial and not a test.

For this month I want you to create as many waveforms as possible and alter the

ADSR of each waveform and save the results. Study the waveforms of commonly used

instruments like pianos, basses, drums etc…whilst listening, and soon you will

get a good idea of the ‘shapes’ of their envelopes and how they sound. Apply any

‘tagging’ method that works for you. I want you to get to the point whereby you

will look at a waveform and deduce what instrument it is.

Kybd: keyboard

KybdTrk: keyboard tracking also known as key follow/scaling

Osc: oscillator

Amp: amplifier

Env: envelope

Vel: velocity

Vol: volume

Freq: frequency

dB: decibel

Hz: Hertz

kHz: Kilohertz

EQ: equalisation

EG: envelope generator

Atk: attack

Dcy: decay

Sus: sustain (in synthesis , in music it means a suspended chord/note)

Rls: release

Mod: modulation

Lfo: low frequency oscillator

Vca: voltage controlled amplifier

Vcf: voltage controlled filter

Vco: voltage controlled oscillator

Dco: digitally controlled oscillator

Dcf: digitally controlled filter

Dca: digitally controlled amplifier

RlsVe:l release velocity

Midi: musical instrument digital interface

Aux: auxiliary

VolEnv: volume envelope

FilEnv: filter envelope

AuxEnv: auxiliary envelope

Xfd: crossfade

RTXfade: real time crossfade

KeySust: key sustain

Chrs: chorus

ChrsAmt: chorus amount

Amt: amount

Pan: pan

AmpPan: amplifier pan

Fil: filter

FilFreq: filter frequency

FilRes: filter resonance (note-on)

VEnvRts: velocity envelope rates

VEnvAtk: velocity envelope attack

VEnvDcy: velocity envelope decay

VEnvSus: velocity envelope sustain

VEnvRls: velocity envelope release

FEnvRts: filter envelope rates

FEnvAtk: filter envelope attack

FEnvDcy: filter envelope decay

FEnvSus: filter envelope sustain

FEnvRls: filter envelope release

FEnvTrig: filter envelope trigger

AEnvRts: auxiliary envelope rates

AEnvAtk: auxiliary envelope attack

AEnvDcy: auxiliary envelope decay

AEnvSus: auxiliary envelope sustain

AEnvRls: auxiliary envelope release

AEnvTrig: auxiliary envelope trigger

Arp: arpeggiator

LPF: low pass filter

HPF: high pass filter

PWM: pulse width modulation

Mono: monophonic

Poly: polyphonic

AM: amplitude modulation

FM: frequency modulation

RM: ring modulation

AS: additive synthesis

WS: wavetable synthesis

PhM: physical modelling

SS: subtractive synthesis

Oct: octave

Lin: linear

Sync: synchronise

Dig: digital

A/D: analogue to digital

D/A: digital to analogue

DAC: digital to analogue converter

DI: direct injection

DAW: digital audio workstation

DSP: digital signal processing

EFX: effects

HD: hard disc

Mic: microphone

MD: mini disc

S/N: signal to noise ratio

SPL: sound pressure level

VU meter: volume unit meter

SYSEX: systems exclusive

These are the most common ones you will come across. I have also inserted some

common terms for the technical side of our industry just to help you understand

the terminology. Each month I will add to the terminology and abbreviations. As

we move along in this tutorial, the definitions will become more detailed and

the use of the above will become clearer.
FILTERS
Welcome to the wonderful world of FILTERS

Briefly explained:

A filter allows you to remove unwanted frequencies and also allows you to boost

certain frequencies. Which frequencies are removed and which frequencies are left

depends on the type of filter you use.

Before we can list the different types of filters and what they do, there are

a few terms and definitions we need to cover. These are crucial and are used all

the time so it is important that you know what these terms are and what they mean.

Last month we covered harmonics and their fundamentals so I hope you are up to

date with this subject as we will be using it as we progress in this tutorial.

It is also a crucial part of this part of the tutorial.

Cut-off frequency This is the point (frequency) at which the filter begins to filter (block or

cut out). The filter will lower the volume of the frequencies above or below the

cut-off frequency depending on the type of filter used. This ‘lowering of the

volume of the frequencies,’ is called Attenuation. In the case of a low pass filter, the frequencies above the cut off are attenuated. In the case of a high pass filter, the frequencies below the cut off are attenuated. Put simply: in the

case of a low pass filter, we are trying to block the (higher) frequencies above

a certain point and allow the lower frequencies through. In the case of a high

pass filter, the opposite is true. We try to cut out or block frequencies below

a certain point and allow the higher frequencies through. On analogue synthesizers

this cut-off was called the slope or gradient. The actual terminology was more accurately described as the RC (resistor/capacitor). Don’t worry about this for now. What you do need to know

is that on analogue synthesizers, the filter behaves differently to modern synthesizers

that use algorithms.

Analogues use circuitry and for that reason alone, it takes time for the filter

to attenuate frequencies, in proportion to the distance from the cut-off point.

Today’s technology allows for instant cut-off as the filter attenuation is determined

by algorithms as opposed to circuits. That is why the filters off an Arp or Oscar

etc, are so much more expressive and warm as they rely completely on the resistors

and capacitors to, first warm up, then to work but in a gradual mode(gradual meaning

sloped or curved as opposed to instant). Depending on how well a filter attenuates

or the way it attenuates gives us an idea of the type of sound we will achieve

with an analogue filter. You often hear someone say ‘That Roland is a warm man’

or ‘Man, is that Arp punchy’. These are statements that explain how a Roland’s

filters sound or how potent the Arp’s filters are. So, the speed at which the

filter attenuates is called the slope or gradient. Another point to raise now

is that you will often see values on the filter knobs on analogue synthesizers

that have 12dB or 24dB per octave. That basically means that each time the frequency

doubles, the filter attenuates by 12dB or 24dB everything at that frequency. These

are also known as 2 pole or 4 pole filters (fig 2), each pole represents 6dB of attenuation. This is how analogue circuits were

built, the number of circuits being used by the filter to perform the task at

hand.

Although this is a general belief, it is one that is not entirely accurate and

we will come to that later, as this is tied into the topic of rolloff. I do not expect you to understand the concept of 2 or 4 pole at this stage.

I am just throwing it in for tagging purposes. Since the term has been mentioned,

it will stay in your mind and when we come to tackle this area, you will say ‘Yeah,

I remember the dude saying something about that’. The best way, and one that I

always sign off with, is to experiment. Sweep that filter. Sweep is another one

of these funky words we programmers or engineers use and all it really means is

grab the filter knob and twist it. Yes, I know, we like to be a bit flash, but

it sounds cool….

So far, it’s all been a bit easy for you. The texts in bold is my method of tagging, so that at any point, you can come back to any parts

of these tutorials and have a reference point. There is more to come on the subject

of filters but first, I would like to list and describe all the different types

of filters, or rather, the most common types. If you delve into the filters that

Emu provide on their synthesis engines, then it could go into pages, if I had

to list them all. Right now I have kept it all simple as we are still in the beginners

stages and when we come to more advanced synthesis techniques, then I have to

get a touch more technical. Topics I will hurt you with, with regards to filters,

will be phase, harmonics and overtones, Laplace Transform, passive and active filters, resonance, isolation, self-oscillation, rolloff, and a few more mind bending surprises. He he, evil I be.

But for now, I shall be a gentle soul and just list the different types of filters

and explain what they do.

Low Pass-LPF (fig1) As mentioned earlier, this filter attenuates the frequencies above the cut-off

point and lets the frequencies below the cut-off point through. In other words,

it allows the lower frequencies through and blocks the higher frequencies, below

and above the cut-off (the frequency at which the filter begins to kick in). The

low pass filter is one mutha of a filter. If you use it on a bass sound, it can

give it more bottom and deep tones. If used on a pad sound, you can have the filter

open and close or just sweep it and it gives that nice closing and opening effect.

You can also use this filter cleverly by removing higher frequency sounds or noise

that you don’t want in your sound or mix. Because it blocks out higher frequencies

at the cut off you set, then it’s a great tool if you want to remove hiss from

a noisy sample or, if you use it gently, you can remove tape or cassette hiss.

Fig1 Fig2

High Pass-HPF (fig3) This is the opposite of the low pass filter. This filter removes the frequencies

below the cut-off and allows the frequencies above the cut-off through. Great

for pad sounds, gives them some top end and generally brightens the sound. It’s

also really good on vocals as it can give the vocals more brightness and you can

also use it on any recordings that have a low frequency hum or sound that is dirtying

the sound, although, in this instance it would be a limited tool, as you could

also cut out the lower frequencies in the sound itself, but still a tool that

has many uses.

Fig3

Band Pass-BPF (fig4) This is a great filter. It attenuates frequencies below and above the cut-off

and leaves the frequencies at the cut-off. It is, in effect, a low pass and a

high pass together. The cool thing about this filter is that you can eliminate

the lower and higher frequencies and be left with a band of frequencies that you

can then use as either an effect, as in having that real mid range type of old

radio sound, or use it for isolating a narrow band of frequencies in recordings

that have too much low and high end. Sure, it’s now really made for that but the

whole point of synthesis is to use tools because that’s what they are, tools.

Breaking rules is what real synthesis is all about. Try this filter on synthesizer

sounds and you will come up with some wacky sounds. It really is a useful filter

and if you can run more than one at a time, and select different cut-offs for

each one, then you will get even more interesting results. Interestingly enough,

band pass filtering is used on formant filters that you find on so many softsynths,

plugins, synthesizers and samplers. Emu are known for some of their format filters

and the technology is based around band pass filters. It is also good for thinning

out sounds and can be used on percussive sounds as well as creating effects type

of sounds. I often get emails from programmers wanting to know how they can get

that old radio effect or telephone line chat effect or even NASA space dialogue

from space to Houston. Well, this is one of the tools. Use it and experiment.

You will enjoy this one.

Fig4

Band Reject Filter-BRF-also known as Notch (fig5)

This is the exact opposite of the band pass filter. It allows frequencies below

and above the cut-off and attenuates the frequencies around the cut-off point.

Why is this good? Well, it eliminates a narrow band of frequencies, the frequencies

around the cut-off, so, that in itself is a great tool. You can use this on all

sounds and can have a distinct effect on a sound, not only in terms of eliminating

the frequencies that you want eliminated, but also in terms of creating a new

flavour to a sound. But its real potency is in eliminating frequencies you don’t

want. Because you select the cut-off point, in essence, you are selecting the

frequencies around that cut-off point and eliminating them. An invaluable tool

when you want to hone in on a band of frequencies located, for example, right

in the middle of a sound or recording. I sometimes use a notch filter on drum

sounds that have a muddy or heavy mid section, or on sounds that have a little

noise or frequency clash in the mid section of a sound.

Fig5

Comb (fig6) The comb filter is quite a special filter. It derives its name from the fact

that it has a number of notches at certain distances (delays), so it looks like

a comb. The Comb filter differs from the other filter types, because it doesn’t

actually attenuate any part of the signal, but instead adds a delayed version

of the input signal to the output, basically a very short delay that can be controlled

in length and feedback. These delays are so short that you only hear the effect

rather than the delays themselves. The delay length is determined by the cut-off.

The feedback depth is controlled by the resonance. Don’t worry about resonance for now as we will cover this extensively in the

next installment of these ongoing tutorials. This filter is used to create a number

of different types of effects, chorus and flange being two of the regulars. But

the comb filter is more than that. It can be used to create some incredible dynamic

textures to an existing sound. When we talk of combs, we have to mention the Waldorf

synthesizers. They have some of the best comb filters and the sounds they affect

are so distinct, great for that funky metallic effect or sizzling bright textures.

Fig6

Parametric This is also called the swept eq. This filter controls three parameters, frequency,

bandwidth and gain. You select the range of frequencies you want to boost or cut,

you select the width of that range and use the gain to boost or cut the frequencies,

within the selected bandwidth, by a selected amount. The frequencies not in the

bandwidth are not altered. If you widen the bandwidth to the limit of the upper

and lower frequencies ranges then this is called shelving. Most parametric filters have shelving parameters. Parametric filters are great

for more complex filtering jobs and can be used to create real dynamic effects

because they can attenuate or boost any range of frequencies.
Resonance

Most synthesizer manufacturers, and in the case of most analogue synthesizers,

the term resonance is used most commonly. Other manufacturers of synthesizers,

or software synthesizers, might call it emphasis or Q.

Last month we talked about filter cut-off and slope, and what they meant. Boosting

the narrow band of frequencies at the cut-off point is called resonance (fig1)

Fig1

If you were to boost the resonance to the maximum, then the filter will begin

to actually self oscillate. This means that it will generate an audible sine wave,

more like whistling, even when receiving no input signal. A very cool way of understanding

what resonance sounds like is to perform what we call a sweep. Yes, another flash and funky term we programmers use to explain something really

simple. Sweeping the filter means manually turning the resonance knob, clockwise

and anti-clockwise. Select a waveform, set the cut-off point and turn the resonance

knob and listen to the results. As you are sweeping, the resonance goes through

all the different frequency harmonics, of the waveform, and boosts them, at the

cut-off point. By the way, I expect you to know what harmonics, waveforms, cut-off

are, as I have covered them in the earlier parts of this tutorial. So, if you

are confused about these terms, then go back and read the earlier parts. I want

you to easily understand and enjoy these explanations.

Altering the resonance of a filter can create incredible sounds. Using the resonance

to it’s maximum will give the shrieking effect, as the frequency band is so small

when the resonance is maxed out. This leads to the filter self oscillating and

the resultant sound is a sine wave that simply screams at you. There are other

ways of using resonance to spice up your sounds. Assigning an LFO to the resonance

can give a nice undulating effect, if used subtly. In this instance, if the resonance

is set higher, then you will have a more dramatic effect. Using a sine wave and

assigning the LFO to modulate the resonance at a high value will give you the

siren type of effect, or if you assign a lower resonance, you will get a deeper

throbbing effect.

These are all dependant on where you select the cut-off point to be. Using resonance

in more subtle ways (higher resonance values on a low frequency bass sound) can

bring out the higher harmonics of low frequencies. This is a great way to depict,

‘presence’ and’ perception’. Although you are boosting the higher frequencies

of a low frequency sound, your ears perceive the sound as more pronounced. The

same can be applied in reverse. Using a low resonance on a high frequency sound

will give the perception of more bass or low end. Another great trick with resonance

is to attain that Moog type of squelchy bass sound. I usually draw a graph or,

input values, that displays a negative filter (ie, the filter start at the negative

and rises to the positive) and then assign a lower frequency and higher resonance.

I then assign a velocity curve to the filter attack and this, when a note is hit,

gives that open/close filter effect on the bass sound. The squelch. Remember the

envelope topic in the earlier part of the tutorial? Well, this is what I am shaping,

the filter envelope. You probably understood nothing of what I just said. I just

thought I’d drop it in there so when we come to programming later, you will remember

this little tip and understand what it does.

If you recall, at the start of these tutorials, I gave a huge list of useful

source and destination routings, terminology, components etc. In there, you would

have seen Filter Env and Filter Amt.

Filter Env is the filter envelope, much like the ADSR envelope we discussed in

and earlier part of this tutorial. This should make sense to you now. The filter

has an envelope as well and it can be shaped much in the same way as the Osc envelope.

Filter Amt refers to the amount of movement there is at the filter cut-off. The

filter env and filter amt go hand in hand. The higher the amount, the more open

the filter, the lower the amount, the less open the filter. This defines the filter

envelope amount at the cut-off.

How many Trance tracks have you heard where a sound starts muffled and then gets

brighter and brighter? This is the filter opening up. It starts closed and opens

up over time. You can easily create this effect by having your mod wheel assigned

to the filter frequency (FilFreq). As you move the mod wheel up it opens the filter by raising the frequencies.

You can then close back down again. There are other nice little tricks you can

use. Try assigning the mod wheel to resonance. This is great for lead sounds.
You could also use the technique we touched on earlier, by choosing white noise

as the osc and having the LFO modulate the resonance of the osc, and using small

amounts of resonance and slow LFO rate, you will get an ocean wave effect. You

could even have a sine as the osc, use an LFO with a mid rate, assign it to the

resonance (low values) and get a bubbling type of effect, like in sci-fi films.

You could even create the effect of static by using the white noise osc and set

high resonance values and have very high LFO rate at a higher pitch, assigning

the mod wheel to the resonance means that by using the mod wheel fervently, you

can create a disjointed, or modulated, static type of effect. You could even throw

in a square wave osc into the equation with a very high rate and have that running

with the white noise osc at the same time. Madness I tell you, madness.

There is so much you can do with just these simple parameters, imagine what you

can do once you have completed this entire tutorial?

I think now is a good time to explain the distinction between passive filters and active filters.

For me to clearly define the differences, I would have to explain some basic

electronics, but I do not feel that you are quite ready, at this point, to enter

that world of madness. So, I will try to explain in very layman’s terms and very

briefly the differences between the two. This is not crucial to know, at this

stage anyway, as you will not really need to know or adopt the distinction unless

you were to build your own filter. However, it does help in understanding the

basic concept of how a filter works in the circuit world.

Passive filters are filters that derive their power from the input signal. Another way

to look at this would be to say that passive filters have no power of their own

until a signal is passed through them, then they wake up and get to the job. That

is very simplistic but, to a certain degree, true. The amplitude response and

phase response of a filter are crucial in determining the result of what is put

into a filter and what comes out. The relationship of what goes into a filter

and what comes out is called the Transfer Function. Phase is a subject that is very important and one that I will deal with in

depth, at a later date, when you can fully understand what I am talking about.

Active filters. To explain this you would need to look at a very basic test case scenario.

Imagine that you had to build a circuit board and had to have a band pass filter

in there. Now, we know that having a low pass filter and a high pass filter, together,

creates a band pass filter, because we attenuate the frequencies below and above

the cut-off, and we are left with a band of frequencies. So, you think by putting

a low pass filter, followed by a high pass filter, on the circuit board, you would

get a band pass filter? I wish. It doesn’t actually work that way and is much

more complicated than that. The reasons are simple.

We know that the filter elements of the two filters interact, that their cut-off

frequencies would be different for each filter, and that there are phase shifting

effects for each of the filter stages. I don’t want to go into explaining each

of these elements as I will come to them at a later date. So, trust me on this.

Putting a low pass filter and a high pass filter together on a circuit board requires

a little more than just a bit of component soldering. What you do need between

these filters, are called operationalamplifiers, (op-amps). These components separate the filter elements from each other. This is what

makes a filter active, the fact that they have op-amps placed between the filters.

Most analogue synthesizers have active filters. There are exceptions but the

majority have active filters. The reason that you can distinguish a filter from

a Moog to an Arp is the fact that they have design elements in their circuitry,

mainly the op-amps, that separates them from their counterpart and gives the make

and model it’s own unique character or sound. Otherwise a low pass filter is just

a low pass filter and should sound no different if it were in a Moog or Arp or

any other make. This is a useful bit of information, as I am so often asked what

makes a certain make and model sound different to it’s counterpart. You now know.

There is , obviously, more to this than just having op-amps on the pathway. The

design of analogue filters varies as well. The filter itself has it’s own character

as the component parts are ‘rated’ and they are never 100% rated (technical term).

This is in the hardware design, measured and tested. If one component is rated

slightly differently to the next component, then it will have a slight variance

in it’s function and performance. This also adds to the characteristic of the

filter.
Phase and Harmonics
Phase

Fig1

Let us take our simplest example, two sine waves, and combine them together.

As you might have guessed from Fig1 (top), adding together two identical waves

produces the same sound, but louder. But what happens if you start the lower wave

halfway through the cycle of the upper one, as in Fig1 (bottom)? They cancel each

other out and you end up with silence. This may seem hard to bend your head around, but it makes perfect sense. It’s

also a cool example to impress the tech babes in cyber bars. As we saw in earlier

parts of this tutorial, a single sine wave can be measured by specifying just

its frequency and amplitude. But by combining two or more waves you must consider

their relative offset. This offset is usually called the ’phase’ of one wave with respect to the other. We can then measure these in degrees

or time. Don’t worry about this for now. What you do need to be clear about is

that the simple offset of one wave against another has very serious implications

in sound wave theory. Now let me complicate this further, simply because you are

having too easy a time of this…he..he.

Combining ‘complex out of phase’ signals does not necessarily lead to complete

cancellation. Let us take an example that deftly brings us to the topic of harmonics and fundamentals. You should know, from earlier parts of this tutorial, what fundamentals and

harmonics are. So if you do not know or are confused at this juncture, then go

back and reread the relevant part. Let us take a saw wave as our example. The

saw wave has every harmonic present. If the first harmonic (fundamental) lies

at 100Hz, the second harmonic will be at 200Hz etc. Adding two of these saw waves,

with the fundamentals offset half a cycle means that the fundamentals are cancelled

out. But the second harmonics, lying at 200Hz, will be added. The third harmonic

will be cancelled out, the fourth harmonic will be reinforced, the fifth harmonic

will be cancelled and the sixth reinforced and so on. The result is a waveform

with harmonics at 200Hz, 400Hz etc. What we are left with is a saw wave with the

amplitude of the original but twice the frequency.

Fourier analysis, the dude from the earlier parts of this tutorial, states that

any two complex signals can be described as an infinite number of sine waves that

represent all the frequencies present in the signal. So, it follows that any given

offset between two identical signals, each frequency will be phase-shifted by

a different amount. Make sense? No? Well, read it again.

You are probably wondering what all this means and what it has to do with synthesis.

Well, it is essential that you understand the concept of phase, and what I have

avoided is to go into deep and emotional graphs and explanations. The subtle move

into the topic of phase is an ongoing diatribe that follows on from the subject

of filters in the last part of this tutorial. The final conclusion is to state

that filtering leads to changes in phase. The very fact that we are effecting

frequencies means that, using the Fourier dude’s thinking, phase is not only created

as a by product of audio manipulation, as outlined above, but filtering can also

lead to phase changes.

Ok, we have touched on phase and, as stated earlier, I want to now enter the

world of harmonics.

I cannot begin to stress how important it is for you to try to bend your head

around this subject. You have often heard the word harmonics used, and in almost

all cases, it is referred to in just about every area of sound synthesis and design.

It is also used in production, mastering, playing/performing, recording etc.

Since we have already touched on this topic, not only in earlier parts of this

tutorial, but also in this month’s instalment, you should have a relatively good

idea as to what harmonics are. We have also covered the fundamental, so that does not need explaining here.

Harmonic Series - Also know as Overtones

To explain what the harmonics series is, it is best to take the sine wave as

our example. The reason for this is quite simple. A Sine wave is the most basic

waveform there is. It is the foundation for the harmonic series. We already know

that a sine wave has no harmonics. It only has the tone of the fundamental frequency. The fundamental frequency, denoted with the symbol F, is the base or root frequency which we identify as pitch. If you refer back

to the frequency chart in the first part of this tutorial, you will find that

all notes have a frequency. As an example, A4 equals 440 Hz. If we were to take

that as the fundamental, then the 1 st harmonic would be 880 Hz, 2 nd harmonic

would be 1320 Hz etc.

Since a sine wave is a pure waveform, i.e. it does not contain any harmonics,

we can create other waveforms simply by adding together any number of sine waves,

all at different frequencies and amplitudes. Any sound can be created using sine

waves at different frequencies and amplitudes. The reverse is also true. Any sound

can be broken down into discrete and distinct sine waves at different frequencies

and amplitudes. A waveform, that does not change it’s timbre over time, is made

up of sine waves which are multiples of the fundamental frequency. This is called

the Natural Harmonic Series, self explanatory really. We can now define the harmonics as F, F2, F3, etc.

In this case, we have the fundamental (F), the second harmonic (F2) and so on.

As we have already covered the topic of phase and what it does in terms of harmonics,

it is then easy to define odd and even harmonics. With the example earlier in this part, we saw that by offsetting

by half a cycle, we were left with the 2nd harmonic, 4 th harmonic, etc. This

is called even harmonics. It then makes sense to assume that we also have odd

harmonics. Both odd and even exist for different types of waveforms. For complex

waveforms, we need to break down the waveforms and ascertain their harmonics.

This is not crucial for you to understand at this juncture. What is crucial is

that you understand the concept of odd and even harmonics.

Saw and Pulse, generally have all the harmonics. Square and Triangle, generally have odd numbered harmonics. However, if you narrow the pulsewidth of the square waveform, even harmonics will appear.

Other areas to note are the amplitudes

• Triangle. The amplitudes of the harmonic series decreases exponentially.
• Square. The amplitudes of harmonic series decreases steadily.
• Saw. The amplitudes of harmonic series decreases steadily.
• Pulse. The amplitudes of harmonic series decreases but the odd and even numbered harmonics

have different amplitudes depending on the Pulse Width.

As with most analogue synthesizers, two or more oscillators can be mixed together

to give a resultant waveform. The resultant waveform is simply the sum of the waveforms (the oscillators). The resultant harmonic series would also

be the sum of the harmonics of the oscillators.

What I have not mentioned, so far, is noise.

In essence, noise is a randomly changing, chaotic signal, containing an endless

number of sine waves of all possible frequencies with different amplitudes. However

randomness will always have specific statistical properties. These will give the

noise its specific character or timbre. If the sine waves’ amplitude is uniform,

which means every frequency has the same volume, the noise sounds very bright.

This type of noise is called white noise. If the amplitude of the sine waves decreases with a curve of about -6 dB per

octave when their frequencies rise, the noise sounds much warmer. This is called pink noise. If it decreases with a curve of about -12 dB per octave we call it brown noise. Bet you didn’t know that one, huh, brown noise?

So we have all these funky names for noise, even though you need to understand

their characteristics, but what are they used for? White noise is used in the

synthesizing of hi-hats, crashes, cymbals etc, and is even used to test certain

generators. Pink noise is great for synthesizing ocean waves and the warmer type

of ethereal pads. Brown noise is cool for synthesizing thunderous sounds and deep

and bursting claps. Of course, they can all be used in varying ways for attaining

different textures and results, but the idea is simply for you to get an idea

of what they ‘sound’ like.
Sample and Hold S/H or S & H

This is the periodic `capture’ of a frequency along a dynamic waveform which

is held until the next `sample’ is taken. The S/H module is used to capture a

signal’s voltage level and hold it until the next voltage is input.

What does this mean in simpler terms? It means that the S/H function is used

in creating voltage sequences, either in repeating patterns (with lower values)

or random patterns (with higher values). And what does what I just wrote mean?

It means turn the damn knob and listen to what happens. Haha. Seriously though,

the S/H is one great feature and most commonly used as a modulator, even though

it itself can have more than one source triggering it. It is normally governed

by the clock generator that sends out pulses, but this is in most modular systems

whereby the S/H module sits alone. You can actually assign anything into the S/H

input, and in most cases oscillators are used and patched into the S/H module.

However, the most common source for the S/H is the noise waveform. On some synthesizers,

these two are hard patched together and you can apply varying amounts of noise

to the input signal. However, oscillators are the most commonly used sources.

Be funky and assign an oscillator to the S/H, which is then patched to the filter

cut-off, or resonance, and move the knob and you will hear the undulating effect

that the S/H creates. Used on very high rates, the S/H starts to randomise the

destination. This makes for some really deep and serious pitch mangling, great

for mad effects or interesting textures. Actually, think of a car siren in Dance

music. This is a great example of S/H being used with a sine wave and pitch.

Other little funky trick is to use saw, S/H on a high value, and noise. You get

burbling and spluttering effects.

I love using S/H on a very small value and have it modulate the pitch of an Lfo.

This makes for a nice moving, undulating texture. Lovely for spacey sounds and

film score type of music.

Sync

You know what the word sync means in general terms. So, let us apply it in terms

of this subject matter.

When you sync one oscillator to another and use the first oscillator as the synchroniser,

or master, and the receiving oscillator as the slave, or receiver, what takes

place is quite simple really. As we know, oscillators behave in cyclical fashion,

ie, they have cycles, the waveforms repeat themselves (cycles). When the master

completes one cycle it then resets the slave to start it’s cycle again. Let me

explain that in simpler terms with an example. If you were to take a sine wave

as the master and a saw wave as a slave, then every time the sine wave completed

a cycle, the saw wave would start it’s cycle. Since we know that cycles determine

frequencies, then it’s obvious that the slave would be running at the frequency

of the master. The new waveform that is formed will be a combination of the contour

of both waveforms (oscillators) but at the frequency of the master waveform.

There, that was simple enough huh?

A good test for ‘hearing’ sync is to detune 2 oscillators and listen to them.

They will sound wide and fat. Now hit the sync button and the sound suddenly behaves

itself and sounds straight. It’s hard to explain, but once you have tried it you

will understand. Of course, sync does have it’s benefits and can be used to create

layered sounds or simply to create a new waveform from 2 existing ones.

Cross Modulation

Is where the pitch of one oscillator is controlled (modulated) by another. However,

what is crucial here is that the modulating oscillator’s pitch is determined by

pressing a note on a keyboard. In effect, the voltage of the key pressed determines

the pitch of the modulating oscillator. This, in turn, modulates another oscillator.

Confused? Hohum….Press a note and the oscillator fires up and the pitch of the

fired up oscillator then modulates the other oscillator. Pah! I am not going to

explain that again. This is sometimes referred to, incorrectly, as Pitch Modulation.

Whereas it is a form of Pitch Modulation, it is not actually Pitch Modulation

itself. However, this form of modulation is better known as FM (Frequency Modulation).

This is why FM synthesis is so cool for creating those bell-like sounds, weird

and wondrous pads and metallic sounds. Yamaha made their name with FM synthesis,

thus the DX7 was

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[9367778> milo milo mi kaska jestem
[Witus> yhy
[9367778> a ty?
[Witus> zmien najpierw w danych publicznych, pa
[9367778> co mam zmienic??
[Witus> w danych publicznych masz wpisane andzelika;p
[9367778> bo to moje pierwsze imie
[9367778> ale pzedstawiam sie drugim
[Witus> witasz sie tez dupa?