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Sound Synthesis Tutorial

:: Introduction to analogue synthesis - Envelope generator

A way to control signals within a synthesizer is by using a module called envelope generator (EG). When an envelope generator receives an "on" gate signal, it sends out a new signal that can be used to control another module on the synthesizer. Analogue synthesizers usually have two envelope generators; one that controls the voltage controlled filter (VCF) and another that controls the voltage controlled amplifier (VCA). In the first instance, the envelope modifies the timbre of a sound over time by controlling the cut-off frequency of the VCF; in the second instance, the envelope modifies the amplitude (volume) of the sound over time to create the natural dynamic movement of a sound. In the second chapter of this tutorial I covered the concept of ADSR envelope; in fact, most of the envelope generators work with the principle of ADSR (Attack - Decay - Sustain - Release).

In the two following pictures we can see the ADSR controls present in the control panels of the MiniMogue VA (left) and the FreeMoog (right) virtual synthesizers. In the first instance, the ADSR control for the filter is labelled as Filter Contour while the ADSR control for the amplifier is labelled as Loudness Contour. In the second instance the controls layout is quite similar, but it is interesting to note how Release is implemented only as an on/off feature.

MiniMogue Luxus Filters FreeMoog Filters

Now it is time to divagate about triggers and gates, the components that control how the signal is sent from the keyboard to the envelope generators. When one press a note in the keyboard, a trigger sends a signal to the envelope generators, so these can start to shape the signal that the oscillators have started to generate. A gate is opened, that will keep the envelope acting as long as the note is pressed (sustained) in the keyboard, and when the note is released, a trigger closes the gate. So it is easy to understand that gate opening is related to Attack while gate closure is related to Release.

As a practical example of all of this, let's suppose that we have programmed some kind of pad sound in our synthesizer and have adjusted in a certain way the envelopes for amplitudes and frequencies. When we press a note in the keyboard, that pad would start to grow in amplitude, which would be responsibility of the ADSR of the VCA, while at the same time the sound would start to get brighter or any other way that would have been programmed, and this would be then responsibility of the ADSR of the VCF. While we keep pressing the note on the keyboard, the sound would remain sustained, and when we release it, the sound would either stop or fade away, depending on the Release time. So when a key is pressed, triggers will activate OSCs, VCF and VCA all together to generate the corresponding sound, that will remain alive while the key is pressed.

In a more technical approach, the process can be explained as this: when a note is pressed on the keyboard, a signal that informs about the corresponding pitch arrives to the oscillators bank; the combination of several oscillators create the corresponding waveform, which includes the desired pitch in its fundamental frequency, and also a spectrum of harmonics (the harmonics series) needed to have the possibility to extract many timbres from the waveform; then this raw waveform is directed into the filters bank, where part of the frequencies (harmonics) are removed or attenuated (subtracted) to approximate a certain timbre, being the filter ADSR envelope responsible of modifying the cut-off frequency (timbre) over time; in the last stage the waveform is sent to the amplificator, whose ADSR envelope will modify the amplitude of the sound over time. So in this way, a signal continuously flows from the oscillators to the amplifier of the synthesizer, during a certain time on which it will be modified according to the programmed parameters, generating so a complex waveform or sound over time. And that is called a musical note.

So until here I have explained enough to get a basic idea on how an analogue synthesizer works and therefore how can it be used to program possible sounds on it. To finish, it is imperative to understand what really are the basic components of an analogue synthesizer, in a very fundamental approach, to avoid wrong concepts that could have been acquired. So always remember that: oscillators are generators, they receive a signal (input), generate a new signal (waveform) based on the signal received, and then send the new signal (output); filters are modifiers, they receive a signal (input) and return a modified signal (output); and envelope generators are controllers, they do not receive signals, but they just modify signals by sending the corresponding parameters (output) to devices that create, or more usually, modify signals (filters and amplificators). In the next chapters I will divagate about additional knowledge on the physics that take place in sound synthesis.

:: Theory on sound synthesis - Filters

To begin, let's see two additional types of filters that are not directly used for subtractive synthesis, but can add additonal features to music generation equipment.

- Comb filter: this filter has a number of band reject (notch) filters at certain distances (delays), and it is not intended to attenuate any part of the signal, but instead for adding a delayed version of the input signal to the output, basically a very short delay that can be controlled in lenght and feedback. The delays are so short that only the effect can be heared, rather than the delays themselves. The lenght of the delays is determined by the cut-off frequency while the feedback is controlled by the resonance - a concept which I will explain later -. What is now practical to know is that this kind of filter is used to create effects such as chorus or flange, effects that analogue synthesizers usually include in their output section.

- Parametric filter: this filter, also known as parametric equalizer, controls three parameters on the signal, which are frequency, bandwidth and gain. It allows to select a range of frequencies (bandwidth) to be attenuated or amplificated (gain), and the desired amount of attenuation or gain. The frequencies out of the selected bandwidth are not altered. Parametric filters are built to cover all the range of frequencies usually used in music production (0 Hz to 20 kHz) and their precision is determined by the number of bands that they have; each band can be considered as an individual filter. So this kind of filter is a grouping of several filters that allow for more complex filtering applications, and they could be used to create dynamic effects over time if an ADSR envelope could be attached to control the bandwidth parameters.

Now it is time to deal with the concept of resonance, which is also referred as emphasis or Q. If you look again at the pictures above, you will notice the knobs labelled as Emphasis that are present in the filter controls of the MiniMogue VA and FreeMoog synthesizers; and if you look at the pictures in the previous chapter, you could see a slider labelled as Resonance in the VCF of the Arppe2600 VA synthesizer. In that previous chapter about filters I explained about the slope which is caused by the relative slowness that analogue filters have when cutting off frecuencies. The range of frequencies where the slope climbs is called the transition band, and the boosting of that range of frequencies is what is called resonance, as the graphic below shows.


Resonance occurs when sound in the pass band near the cut-off frequency is sent back into the filter as it comes out, creating feedback. The amount of feedback affects the volume of these frequencies, as well as the timbre of the sound. Which practical use would one make of this hump on the slope? Altering the resonance of a filter along with the cut-off frequency can create incredible sounds, for example the effect of self-oscillation, that would generate an audible - and screaming - sine wave. Another undulating sound effects that are related to sine waves as well, such as shrieks, sirens or throbs, are possible, specially if a low frequency oscillator is assigned to the filter resonance. Another trick consists in using a high resonance to highlight the higher harmonics of a low frequency or bass sound, which adds presence to the sound, or the opposite, using a low resonance to highlight the lower harmonics of a high frequency or treble sound.

If you look again at the pictures above, you will see the knobs labelled as Contour Amount (MiniMogue VA) and Amount Of Contour (FreeMoog). Filter amount (contour amount) determines how much sensitive the filter cut-off will be to the control by the envelope generator, so the higher the amount, the more open the filter (the larger the range of frequencies the filter allows to pass through it). It is just the amount of modulation that the filter envelope generator does on the cut-off point.

To end this chapter, I will explain the distinction between passive filters an active filters. This knowledge is not really essential for the practical utilization of a synthesizer, but it would help to better understanding how a filter works.

- Passive filters derive their power from the input signal, so they have no power on their own until a signal passes through them. In this case, the amplitude response and phase response of a filter are crucial in determining the relationship of what enters into a filter and what leaves out of it, which is called the transfer function. The basic transfer functions were explained in the previous chapter about filters, and the concept of phase will be covered in the next chapter.

- Active filters use active components such as amplifiers, placed between the transfer functions implemented on the filter, and designed to improve the performance and predictability of the filter, while avoiding the need for inductors, which are typically expensive compared to other components. An amplifier also prevents the load impedance of the following stage from affecting the characteristics of the filter, and can have complex poles and zeros without using a bulky or expensive inductor. The shape of the response, the quality factor and the tuned frequency can often be set with inexpensive variable resistors. In some active filter circuits, one parameter can be adjusted without affecting the others. Using active elements has some limitations: available active devices have limited bandwidth, so they are often impractical at high frequencies, while amplifiers consume power and inject noise into a system.

Most analogue synthesizers have active filters, and the diverse designs of amplificators are a reason whereby analogue synthesizers from different manufacturers have a distinctive sound.

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