ADDITIVE

Does what it says on the tin. By adding one or more basic and simple waveforms together and their harmonics you create a complex waveform. However, you need to add an enormous amount of harmonics to create the simplest of sounds and this type of synthesis can be complicated to create in the form of a synthesizer but the Kawai K5000 does exactly that. You can create extremely rich textures or wild and crazy sounds on this beast. Personally, I love additive synthesis but then again I am receiving very intense therapy. The process of additive synthesis is also referred to as summing the waveforms and harmonics. This method adopts Fourier analysis. Described as the representation of a sound’s frequency components as a sum of pure sinusoidal waves. An analysis of a sound’s frequency components is taken at a steady state to give an approximation of that sounds spectrum. As most natural sounds are spectrally dynamic, one single Fourier analysis could not possibly represent a sound in sine waves. By ‘windowing’, a Fast Fourier Transform (FFT) takes several of these approximations and strings them together to better predict a sound’s spectrum over time. Although this is daunting to take in it is crucial to know and I don’t expect you to understand Fourier analysis but just thought I would bring it in now as we will come back to this at the advanced stages of these tutorials.

SUBTRACTIVE

This process involves the generating of complex waveforms and then filtering the frequencies so that you are then left with the sound you want. You take away the frequencies. Obviously the filters are crucial in subtractive synthesis and the better the filters and the wider the choice of filters available, the better the end result will be. When I teach students I find Subtractive Synthesis to be the easiest and most logical introduction to sound design.

FREQUENCY MODULATION (FM)

The output of one oscillator (modulator) is used to modulate the frequency of another oscillator (carrier). These oscillators are called operators. FM synthesizers usually have 4 or 6 operators. Algorithms are predetermined combinations of routings of modulators and carriers. To really explain this I would have to go into harmonics, sidebands, non-coincident and coincident series and the relationships between modulators and carriers. So I won’t go there. What I will say is that FM synthesis can create lovely digital types of sounds, from brittle to lush. A little bit of info for DX owners, is that the oscillators on these synthesizers were all sine waves.

PHYSICAL MODELLING (PM or PHM)

This form of synthesis simulates the physical properties of natural instruments, or any sound, by using complex mathematical equations in real-time. This requires huge processing power. You are not actually creating the sound but, you are creating and controlling the process that produces that sound. Waveguides and algorithms come into this process heavily but, again, I won’t go into that as it would complicate issues and confuse you. What I do find interesting is that the Nord Lead uses PM synthesis to emulate an analog synthesizer. Weird huh?

LINEAR ARITHMETIC SYNTHESIS

This type of synthesis takes short attack sampled waveforms called PCM, or Pulse Code Modulation and combines them with synthesized sounds that form the body and tail of the new sound. By layering these and combining them with the synthesized portion of the sound you arrive at the new sound. The final sound is processed by using filters, envelope generators etc. This is one of the most common forms of synthesis used in the 90s and even today. Roland was the most famous for adopting this type of synthesis and the D50 was one of the most common of the synthesizers that used LA synthesis. By the way, a great synthesizer and still used today.

WAVETABLE SYNTHESIS

This form of synthesis incorporates the use of pre-recorded digitized audio waveforms of real or synthetic instruments. The waveforms are then stored in memory and played back at varying speeds for the corresponding notes played. These waveforms usually have a looped segment which allows for a sustained note to be played. Using envelopes and modulators, these waveforms can be processed and layered to form complex sounds that can often be lush and interesting. The processes are algorithmic and memory is crucial to house the waveforms. I could get into linear crossfading sequentially, quasi-periodic and sine functions etc. But I won’t. I care about your sanity.

GRANULAR SYNTHESIS

This is the method by which tiny events of sounds (grains or clouds) are manipulated to form new complex sounds. By using varying frequencies and amplitudes of the sonic components, and by processing varying sequences and durations of these grains, a new complex sound is formed. Simply put, this form of synthesis creates some crazy and interesting sounds.

ADVANCED VECTOR SYNTHESIS ( AVS )

This method of synthesis incorporates the combining and processing of digital waveforms. Using PCM samples, effects and filtering this method of synthesis can create stunning sounds, from lush and evolving pads to strange stepped sequences. Korg made the famous Wavestation range of synthesizers and these were based around the Sequential Circuits Prophet VS. Working off a 2-dimensional envelope using an X and Y axis (joystick) and 4 voices, this synthesizer also had wave sequencing, playing a loopable sequence of PCM samples in a rhythmic and/or crossfaded fashion. The idea was to be able to crossfade 2 or more waveforms using the joystick. Freaky but incredible.

Synthesizers used to be categorized into 2 categories, analog and digital, but in the modern world, we have the hybrid digital-analog synthesizer. Confused? Don’t be. It actually makes good sense. Whether the processing is digital or analog, the sounds are produced in much the same way, albeit with different terminology and some different routings. A good example of the terminology being adapted is: VCO, voltage controlled oscillator becomes a DCO, digitally controlled oscillator.

Of course, with today’s influx of synthesizers, be they analog, digital or a hybrid system, we are offered many varying forms of synthesis, from granular to transwave and so on.

Ultimately, synthesis is about sound design and manipulation and we now have fully fledged systems whereby effects are incorporated into the signal matrix, dynamics at the output stage and modulation matrices that make modular synthesizers look primitive. But would I swap a modular synth for a hybrid plugin? Hell no!

A basic guide into how Reason’s Subtractor works.

Let us start with the sound generation engine of Subtractor.

These are the oscillators.

Subtractor has 2 oscillators.
Each one has 32 waveforms but you only need to concern yourself with the basic 4 waveforms as these are your raw waveforms and the balance of the waveforms are simply varying harmonic versions of the main four.

The first waveform is:

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.

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.

Square waveforms look like a bunch of squares with their tops and bottoms missing and alternatively.

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.

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.

The sine is a nice smooth flowing waveform.

The rest of the waveforms, from 5-32, are variances on harmonic content and shape structures. This basically means that certain waveforms are created with certain characteristics in mind. For example waveform 11 has been created for designing voice-like sounds, waveform 8 has been created with clav sounds in mind. Each waveform has it’s own attributes and therefore a great starting point for your sound design needs.

OSCs generate waveforms and pitch and these are nicely displayed next to the OSCs in their section. Oct is simply a shift in pitch by octaves either down or up.
Semi deals with semitone shifts and this is the standard 12 up or down which makes up one octave.
Cents are 100th fractions of the semitones.

Phase

Each oscillator has it’s own Phase knob and a selector button. The Phase knob is used to set the amount of phase offset, and the selector switches between three modes:

Waveform multiplication (x)

Waveform subtraction (-)

No phase offset modulation (o).

When phase offset modulation is activated, the oscillator creates a second waveform of the same shape and offsets it by the amount set with the Phase knob. Depending on the selected mode, Subtractor then either subtracts or multiplies the two waveforms with each other. The resulting waveforms can be used to create new timbres.

Let’s take a simple saw wave example. Offset this slightly and use the subtract mode and you have now created a pulse wave. Pulse waveforms are great for many types of sounds.
Again, I suggest you read my synthesis tutorials to listen to and understand how certain waveforms sound and what they can be used for.

But my best advice in this tutorial is to simply play around and attune your ears to the different types of sounds that can be created by simply using the phase functions.
We will get deep into this in later tutorials when we come to creating pulse width modulated sounds etc.]

Keyboard Tracking

We now come to osc keyboard tracking.
This is an extremely useful little function.

If you deselect keyboard tracking or switch it off, then you will have a constant pitched oscillator. In other words, no matter which note you press, the same pitch is outputted. This is handy when it comes to creating non-pitched sounds like drum and percussive sounds, or special effects, where you do not want the pitch to be evident when you play up or down the keyboard.

Osc mix is the mix for both OSCs. Set it to the left and you only hear OSC 1, set it all the way to the right and you hear OSC 2, set it midway and both OSCs are heard.
The OSC 2 output mix is also important because both the ring modulator and noise generator are output through here.

Frequency Modulation

or FM is when the frequency of one oscillator (called the “carrier”) is modulated by the frequency of another oscillator (called the “modulator”).
In Subtractor, Osc 1 is the carrier and Osc 2 the modulator.

Using FM can produce a wide range of harmonic and non-harmonic sounds.
The best advice I can give, short of a deep and emotional explanation, is to try this function out for yourself.
In fact, let’s try a simple example of FM in Subtractor.

To use OSC 2 you need to ‘switch’ it on by clicking on the little box above it.
Now select a waveform for OSC 1.
Let’s use a saw for OSC 1 and a sine wave for OSC 2.
Now set the FM dial to midway and the mix dial to halfway and play away.
Sounds nice and metallic huh?
You can vary the amounts, change the waveforms etc and later when we come to the mod matrix of Subtractor where we can route sources and destinations, you will find that we can create an endless array of sounds by simply changing what modulates what.

Ring Modulators

basically multiply two audio signals together. The ring modulated output contains added frequencies generated by the sum of, and the difference between, the frequencies of the two signals.
In the Subtractor Ring Modulator, Osc 1 is multiplied with Osc 2 to produce sum and difference frequencies.

Ring modulation can be used to create complex and enharmonic sounds.
Although this sounds emotional, it’s actually very simple to use.

Let’s try a little ring mod example.
Switch on OSC 2 and leave it at the default saw waveform.
Now turn the OSC mix dial all the way to the right as the ring mod is outputted on OSC 2.
Now switch on the ring mod icon and go to either OSCs semitone tuning section and move the values down a few semitones. You can immediately hear the ring mod sound being outputted.

Cool huh?

We now come to the final part of the OSC section, the noise generator.
This can actually be viewed as a third OSC but it does not behave in the same way as the other two as noise is non-pitched.

Noise waveforms

are used more for effect than anything else but I find that they are fantastic for creating pads when used with other waveforms like saws and triangles.

You can also create great seashore wave types of sounds or huge thunder or even some great Hoover type sounds when used with saw waveforms. Hell, you can even create drum and percussive sounds with noise.

Let’s try a little example;

First off, in Subtractor the noise generator is internally routed to OSC 2, so if you switch OSC 2 on then the noise is mixed with the OSC 2 waveform.
By switching off OSC 2 and using only osc1 with the OSC mix set all the way to the right, you will only hear the noise generator and not OSC 1 or 2.
This sounds complex but is actually quite simple when you practice a little and we will do that right now.

So, use an initialised patch.
Move the OSC amt dial all the way to the right.
This will bypass OSC 1 ‘s waveform, and because OSC 2 is not switched on, the noise will be heard on its own.
Now switch on the noise generator by clicking on the little box above it till it lights up red.
You now have noise.

The three parameters are easy to understand and use.
Decay is simply how long it takes for the sound to die when you play a note. The less decay the shorter the sound. The more decay you use the longer the sound. It’s actually that simple.

Colour

is a little more involved.
If you move the dial all the way to the right, you will get pure or white noise. This comes across as a bright noise sound. Move it back towards the left and the brightness will fade. Move it all the way and you get a low-frequency rumble (great for waves and earthquakes etc).
Level is self-explanatory.

Now try the dials.
Move the colour all the way to the right. Move the level all the way to the right and move the decay to 1/4 of the way. You will now hear an electro hi-hat.
Move the decay to the right on the dial, move the colour to midway and level to midway and you will hear a type of thunder.
This is all for now but when we come to modulating this generator you will see that we can create a huge array of different textures.

Filters

A filter is the most important tool for shaping the overall timbre of the sound.

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.

The filter section in Subtractor contains two filters, the first being a multimode filter with five filter types, and the second being a low-pass filter.

Filter Keyboard Track (Kbd)

If Filter Keyboard Track is activated, the filter frequency will increase the further up on the keyboard you play. If a lowpass filter frequency is constant (a Kbd setting of “0”) this can introduce a certain loss of “sparkle” in a sound the higher up the keyboard you play, because the harmonics in the sound are progressively being cut. By using a degree of Filter Keyboard Tracking, this can be compensated for.

Filter 2
A very useful and unusual feature of the Subtractor Synthesizer is the presence of an additional 12dB/Oct lowpass filter. Using two filters together can produce many interesting filter characteristics, that would be impossible to create using a single filter, for example, formant effects.

The parameters are identical to Filter 1, except in that the filter type is fixed, and it does not have filter keyboard tracking.

To activate Filter 2, click the button at the top of the Filter 2 section.

Filter 1 and Filter 2 are connected in series. This means that the output of Filter 1 is routed to Filter 2, but both filters function independently. For example, if Filter 1 was filtering out most of the frequencies, this would leave Filter 2 very little to “work with”. Similarly, if Filter 2 had a filter frequency setting of “0”, all frequencies would be filtered out regardless of the settings of Filter 1.

Filter Link
When Link (and Filter 2) is activated, the Filter 1 frequency controls the frequency offset of Filter 2. That is, if you have set different filter frequency values for Filter 1 and 2, changing the Filter 1 frequency will also change the frequency for Filter 2, but keeping the relative offset.

Filter Envelope
The Filter Envelope affects the Filter 1 Frequency parameter. By setting up a filter envelope you control how the filter frequency should change over time with the four Filter Envelope parameters, Attack, Decay, Sustain and Release.

Filter Envelope Amount
This parameter determines to what degree the filter will be affected by the Filter Envelope. Raising this knob’s value creates more drastic results. The Envelope Amount parameter and the set Filter Frequency are related. If the Filter Freq slider is set to around the middle, this means that the moment you press a key the filter is already halfway open. The set Filter Envelope will then open the filter further from this point. The Filter Envelope Amount setting affects how much further the filter will open.

Filter Envelope Invert
If this button is activated, the envelope will be inverted. For example, normally the Decay parameter lowers the filter frequency, but after activating Invert it will instead raise it, by the same amount.

Relevant content:

Synthesis and Sound Design – Various Types

Preparing the Listening Environment

I do not want to get into serious sound reinforcement or acoustic treatment here, for the very simple reason that it is a vast subject and one that is so subjective, that even pros debate it all day, with differing views.

I also believe that every room has it’s own unique problems and must be treated as such, instead of offering a carte blanche solution that would probably make things worse. However, to fully understand what needs to be done to a room to make it more accurate for listening purposes, requires that we understand how sound works in a given space, and how we perceive it within that space.

I think a good place to start, without getting technical, is to think of a room that is completely flat in terms of a flat amplitude response.

This would mean the room has almost no reflective qualities and would invariably be too dead for our purposes. The other side of the coin is a room that is too reflective, and that would be worse than a completely dead room.
We need to concentrate on a happy compromise and a realistic scenario.

What we are trying to achieve is to utilize the room’s natural reflective qualities, and find ways to best expose audio, whilst beating the reflective battle.

Whoa, deep statement….

To put it more simply: we are trying to limit the interference of the room with speaker placement and the listening position.

The way we determine the location of sound in a given space is by measuring, with our brains, the delay of the sound between our ears. If the sound reaches the left ear first, then our brain determines that the sound is coming from the left. If there is no delay and the sound arrives at both ears at the same time, then we know that the sound is directly in front of us.

This piece of information is crucial in locating sounds and understanding the space they occupy.

 Now, imagine a room that has loads of reflections and reflections that come from different angles, and at different time intervals.  You can see why this would provide both confusing and inaccurate data for our brains to analyze.

Sound

Let us have a very brief look at how sound travels, and how we measure its effectiveness.

Sound travels at approximately1130 feet per second.

Now let us take a frequency travel scenario and try to explain it’s movement in a room. For argument’s sake, let’s look at a bass frequency of 60 Hz.

When emitting sound, the speakers will vibrate at a rate of 60 times per second. Each cycle (Hz) means that the speaker cones will extend forward when transmitting the sound, and refract back (rarefaction) when recoiling for the next cycle.

These vibrations create peaks on the forward drive and troughs on the refraction. Each peak and trough equates to one cycle.
Imagine 60 of these every second. We can now calculate the wave cycles of this 60 Hz wave.

We know that sound travels at approximately 1130 feet per second, so we can calculate how many wave cycles that is for the 60 Hz wave.

The Calculations

We divide 1130 by 60, and the result is around 19 feet (18.83 if you want to be anal about it). We can now deduce that each wave cycle is 19 feet apart. To calculate each half-cycle, i.e. the distance between the peak and trough, drive and rarefaction, we simply divide by two. We now have a figure of 91/2 feet. However, this is assuming you have no boundaries of any sort in the room, i.e. no walls or ceiling. As we know that to be utter rubbish, we then need to factor in the boundaries. Are you still with me here?

These boundaries will reflect back the sound from the speakers and get mixed with the original source sound. This is not all that happens. The reflected sounds can come from different angles and because of their ‘bouncing’ nature; they could come at a different time to other waves.

And because the reflected sound gets mixed with the source sound, the actual volume of the mixed wave is louder. In certain parts of the room, the reflected sound will amplify because a peak might meet another peak (constructive interference), and in other parts of the room where a peak meets a trough (rarefaction), frequencies are canceled out (destructive interference).

Calculating what happens where is a nightmare.

This is why it is crucial for our ears to hear the sound from the speakers arrive before the reflective sounds. For argument’s sake, I will call this sound ‘primary’ or ‘leading’, and the reflective sound ‘secondary’ or ‘following’.
Our brains have the uncanny ability, due to an effect called the Haas effect, of both prioritizing and localizing the primary sound, but only if the secondary sounds are low in amplitude.
So, by eliminating as many of the secondary (reflective) sounds as possible, we leave the brain with the primary sound to deal with. This will allow for a more accurate location of the sound, and a better representation of the frequency content.

But is this what we really want?

I ask this because the secondary sound is also important in a ‘real’ space and goes to form the tonality of the sound being heard. Words like rich, tight, full etc. all come from secondary sounds (reflected).
So, we don’t want to completely remove them, as this would then give us a clinically dead space.  We want to keep certain secondary sounds and only diminish the ones that really interfere with the sound.

Our brains also have the ability to filter or ignore unwanted frequencies.

In the event that the brain is bombarded with too many reflections, it will have a problem localizing the sounds, so it decides to ignore, or suppress, them.
The best example of this is when there is a lot of noise about you, like in a room or a bar, and you are trying to have a conversation with someone. The brain can ignore the rest of the noise and focus on ‘hearing’ the conversation you are trying to have.

I am sure you have experienced this in public places, parties, clubs, football matches etc.
To carry that over to our real world situation of a home studio, we need to understand that reflective surfaces will create major problems, and the most common of these reflective culprits are walls. However, there is a way of overcoming this, assuming the room is not excessively reflective and is the standard bedroom/living room type of space with carpet and curtains.

We overcome this with clever speaker placement and listening position, and before you go thinking that this is just an idea and not based on any scientific foundation, think again.

The idea is to have the primary sound arrive at our ears before the secondary sound. Walls are the worst culprits, but because we know that sound travels at a given speed, we can make sure that the primary sound will reach our ears before the secondary sound does. By doing this, and with the Haas effect, our brains will prioritize the primary sound and suppress (if at low amplitude) the secondary sound, which will have the desired result, albeit not perfectly.

A room affects the sound of a speaker by the reflections it causes. We have covered this and now we need to delve a little more into what causes these reflections. Some frequencies will be reinforced, others suppressed, thus altering the character of the sound. We know that solid surfaces will reflect and porous surfaces will absorb, but this is all highly reliant on the materials being used. Curtains and carpets will absorb certain frequencies, but not all, so it can sometimes be more damaging than productive. For this, we need to understand the surfaces that exist in the room. In our home studio scenario, we are assuming that a carpet and curtains, plus the odd sofa etc, are all that are in the room. We are not dealing with a steel factory floor studio.

In any listening environment, what we hear is a result of a mixture of both the primary and secondary (reflected) sounds. We know this to be true and our sound field will be a combination of both. In general, the primary sound, from the speakers, is responsible for the image, while the secondary sounds contribute to the tonality of the received sound.

The trick is to place the speaker in a location that will take advantage of the desirable reflections while diminishing the unwanted reflections.

Distance to side wall and back wall.

Most speakers need to be a minimum of a foot or two away from the side and back walls to reduce early reflections. Differences among speakers can also influence positioning, so you must always read the manufacturer’s specifics before starting to position the speakers. A figure-of-eight pattern may be less critical of a nearby side wall, but very critical of the distance to the back wall. The reverse is true for dynamic speakers that exhibit cardioid patterns. In general, the further away from reflective surfaces, the better.

It is also crucial to keep the distances from the back wall and side walls mismatched.

If your speakers are set 3 feet from the back wall, do NOT place them 3 feet from the side walls, place them at a different distance.

Another crucial aspect of the listening position and speaker placement is that the distance from your listening position to each speaker be absolutely identical. It has been calculated that an error of less than ½” can affect the speaker sound imaging, so get this absolutely correct.

Distance to speakers from listening position.

 Once you have established the above, you now need to sort out the distance from the listener to the speakers. I work off an equilateral triangle with the seating position being at the apex of this triangle. The distances must all be equal.

The other factor to consider is the distance between the speakers. Too close and you will get a narrow soundstage with the focus being very central. Widening the distance between the speakers will afford you a wider stereo width, but too far and you will lose the integrity of the soundstage.

Toe-in.

This is the angle of the speakers facing the listener. There are a number of factors that influence the angle of the speakers.

The room, the speakers themselves, and your preferable listening angle. I always start at an excessive toe-in and work outwards until I can hear the soundstage perfectly.

Tilt.

 Tilt is also crucial. Depending on the make of the speakers, most speakers are meant to be level set, but some might require tilting and in most cases, the tilt is rear high. If you have to have the speakers tilted then start off level and work from there.

Personally I prefer a level speaker setup.

Listening height.

 You will find that the optimum listening height is that of the speaker’s center being at exactly ear height.

However, certain speakers have their own specific height recommendations. You will find that with 3-way systems that incorporate top, mid and subwoofers, the listening height is more customized to account for the woofer placements in the speaker cabin or housing.

Seating location.

 I find that keeping the seating position 1-3 feet from the boundary wall gives me the best bass response, and because the distance is too short for the brain to measure the time delay and thus locate the source of the reflection.

Please look at the figure below (Fig 1)

 

 

The listening position is at the rear of the room with the speakers facing and forming the equilateral triangle setup, and the listening position forming the apex of the triangle.

The elliptical shape denotes the soundstage and as you can plainly see, the side and rear walls do not interfere with the soundstage.

As you can see, I have created this soundstage using the longer walls as the back and front walls, instead of creating the soundstage with the listening position on the shorter walls. This allows me to position the speakers as wide as is sonically possible and thus affording me a wider stereo field.

Place the listening chair near the rear wall, because the distance (1 to 3 feet) is too short for the brain to measure the time delay and locate the source of the reflection. Also, it places you at the room boundary where the perception of bass is greatest.

 Please make sure to take care in optimizing your listening environment.

Once this has been achieved, you can mix far more accurately and truthfully.