This section covers the general principles of electronic sound generation and processing in more detail, including references to Bass Station II’s facilities where relevant. It is recommended that this chapter is read carefully if analogue sound synthesis is an unfamiliar subject. Users familiar with this subject can skip this section and move on to the next.
To gain an understanding of how a synthesiser generates sound it is helpful to have an appreciation of the components that make up a sound, both musical and non-musical.
The only way a sound may be detected is by air vibrating the eardrum in a regular, periodic manner. The brain interprets these vibrations (very accurately) into one of an infinite number of different types of sound.
Remarkably, any sound may be described in terms of three properties, and all sounds always have them. They are:
What makes one sound different from another is the relative magnitudes of the three properties as initially present in the sound, and how the properties change over the duration of the sound.
With a musical synthesiser, we deliberately set out to have precise control over these three properties and, in particular, how they can be changed during the “lifetime” of the sound. The properties are often given different names, e.g., Volume may be referred to as Amplitude, Loudness or Level, Pitch as Frequency and sometimes Timbre as Tone.
As stated, sound is perceived by air vibrating the eardrum. The pitch of the sound is determined by how fast the vibrations are. For an adult human, the slowest vibration perceived as sound is about twenty times a second, which the brain interprets as a low bass sound; the fastest is many thousands of times a second, which the brain interprets as a high-pitched sound.
If the number of peaks in the two waveforms (vibrations) are counted, there are exactly twice as many peaks in Wave B as in Wave A. (Wave B is actually an octave higher in pitch than Wave A.) The number of vibrations in a given period determines the pitch of a sound. This is the reason pitch is sometimes referred to as frequency. It is the number of waveform peaks counted during a given period of time which defines the pitch, or frequency.
Musical sounds consist of several different, related pitches occurring simultaneously. The lowest is referred to as the ‘fundamental’ pitch and corresponds to the perceived note of the sound. Other pitches making up the sound which are related to the fundamental in simple mathematical ratios are called harmonics. The relative loudness of each harmonic as compared to the loudness of the fundamental determines the overall tone or ‘timbre’ of the sound.
Consider two instruments such as a harpsichord and a piano playing the same note on the keyboard and at equal volume. Despite having the same volume and pitch, the instruments still sound distinctly different. This is because the different note-making mechanisms of the two instruments generate different sets of harmonics; the harmonics present in a piano sound are different to those found in a harpsichord sound.
Volume, which is often referred to as the amplitude or loudness of the sound, is determined by how large the vibrations are. Very simply, listening to a piano from a metre away would sound louder than if it were fifty metres away.
Having shown just three elements may define any sound, these elements now have to be realised in a musical synthesiser. It is logical that different sections of the synthesiser ‘synthesize’ (or create) each of these different elements.
One section of the synthesiser, the Oscillators, generate raw waveform signals which define the pitch of the sound along with its raw harmonic content (tone). These signals are then mixed together in a section called the Mixer, and the resulting mixture is then fed into a section called the Filter. This makes further alterations to the tone of the sound, by removing (filtering) or enhancing certain harmonics. Lastly, the filtered signal is fed into the Amplifier, which determines the final volume of the sound.
Additional synthesiser sections - LFOs and Envelopes - provide further ways of altering the pitch, tone and volume of a sound by interacting with the Oscillators, Filter and Amplifier, providing changes in the character of the sound which can evolve over time. Because LFOs’ and Envelopes’ only purpose is to control (modulate) the other synthesiser sections, they are commonly known as ‘modulators’.
These various synthesiser sections will now be covered in more detail.
The Oscillator section is the heart of the synthesiser. It generates an electronic wave (which creates the vibrations when eventually fed to a loudspeaker). This waveform is produced at a controllable musical pitch, initially determined by the note played on the keyboard or contained in a received MIDI note message. The distinctive tone or timbre of the waveform is actually determined by the waveform’s shape.
Many years ago, pioneers of musical synthesis discovered just a few distinctive waveforms contained many of the most useful harmonics for making musical sounds. The names of these waves reflect their actual shape when viewed on an instrument called an oscilloscope, and they are: Sine waves, Square waves, Sawtooth waves, Triangle waves and Noise. Each of Bass Station II’s Oscillator sections can generate all these waveforms, and can generate non-traditional synth waveforms as well. (Note that Noise is actually generated independently and mixed in with the other waveforms in the Mixer section.)
Each waveform (except Noise) has a specific set of musically-related harmonics which can be manipulated by further sections of the synthesiser.
The diagrams below show how these waveforms look on an oscilloscope, and illustrate the relative levels of their harmonics. Remember, it is the relative levels of the various harmonics present in a waveform which determine the tonal character of the final sound.
These possess just one harmonic. A sine waveform produces the “purest” sound because it only has this single pitch (frequency).
These contain only odd harmonics. The volume of each decreases as the square of its position in the harmonic series. For example, the 5th harmonic has a volume 1/25th of the volume of the fundamental.
These are rich in harmonics, and contain both even and odd harmonics of the fundamental frequency. The volume of each is inversely proportional to its position in the harmonic series.
Square/Pulse waves contain only odd harmonics, which are at the same volume as the odd harmonics in a sawtooth wave.
The square waveform spends an equal amount of time in its ‘high’ state as in its ‘low’ state. This ratio is known as the ‘duty cycle’. A square wave always has a duty cycle of 50% which means it is ‘high’ for half the cycle and ‘low’ for the other half. Bass Station II lets you adjust the duty cycle of the basic square waveform (via the Shape controls) to produce a waveform which is more ‘rectangular’ in shape. These are often known as Pulse waveforms. As the waveform becomes more and more rectangular, more even harmonics are introduced and the waveform changes its character, becoming more ‘nasal’ sounding.
The width of the pulse waveform (the ‘Pulse Width’) can be altered dynamically by a modulator, which results in the harmonic content of the waveform constantly changing. This can give the waveform a ‘fat’ quality when the pulse width is altered at a moderate rate.
A pulse waveform sounds the same whether the duty cycle is – for example - 40% or 60%, since the waveform is just “inverted” and the harmonic content is exactly the same.
Noise is a random signal, and does not have a fundamental frequency (and therefore has no pitch property). Noise contains all frequencies, and all are at the same volume. Because it possesses no pitch, noise is often useful for creating sound effects and percussion type sounds.
A Ring Modulator is a sound generator that takes signals from two oscillators and effectively “multiplies” them together. Bass Station II’s Ring Modulator uses Oscillator 1 and Oscillator 2 as inputs. The resulting output depends on the various frequencies and harmonic content present in each of the two oscillator signals, and will consist of a series of sum and difference frequencies as well as the frequencies present in the original signals.
To extend the range of sounds that may be produced, typical analogue synthesisers have more than one Oscillator. By using multiple Oscillators to create a sound, it is possible to achieve very interesting harmonic mixes. It is also possible to slightly detune individual Oscillators against each other, which creates a very warm, ‘fat’ sound. Bass Station II’s Mixer allows you create a sound consisting of the waveforms of Oscillators 1 and 2, the separate sub-octave oscillator, a Noise source, the Ring Modulator output and an external signal, all mixed together as required.
Bass Station II is a subtractive music synthesiser. Subtractive implies that part of the sound is subtracted somewhere in the synthesis process.
The Oscillators provide the raw waveforms with plenty of harmonic content and the Filter section subtracts some of the harmonics in a controlled manner.
7 types of Filter are available on Bass Station II; they are all variations of the three basic filter types: Low Pass, Band Pass and High Pass. The type of Filter most commonly used on synthesisers is Low Pass. On a Low Pass Filter, a “cut-off frequency” is chosen and any frequencies below this are passed, while frequencies above are filtered out, or removed. The setting of the Filter Frequency parameter dictates the point above which frequencies are removed. This process of removing harmonics from the waveforms has the effect of changing the sound’s character or timbre. When the Frequency parameter is at maximum, the filter is completely “open” and no frequencies are removed from the raw Oscillator waveforms.
In practice, there is a gradual (rather than a sudden) reduction in the volume of the harmonics above the cut-off point of a Low Pass Filter. How rapidly these harmonics reduce in volume as frequency increases above the cut-off point is determined by the Filter’s slope. The slope is measured in ‘volume units per octave’. Since volume is measured in decibels, this slope is usually quoted as so many decibels per octave (dB/oct). The higher the number, the greater the rejection of harmonics above the cut-off point, and the more pronounced the filtering effect. Bass Station II’s filter section provides two slopes, 12 dB/oct and 24 dB/oct.
A further important parameter of the Filter is its Resonance. Frequencies at the cut-off point may be increased in volume by the Filter Resonance control. This is useful for emphasising certain harmonics of the sound.
As Resonance is increased, a whistling-like quality will be introduced to the sound passing through the filter. When set to very high levels, Resonance actually causes the filter to self-oscillate whenever a signal is being passed through it. The resulting whistling tone being produced is actually a pure sine wave, the pitch of which depends on the setting of the Frequency knob (the filter’s cut-off point). This resonance-produced sine wave can actually be used for some sounds as an additional sound source if wished.
The diagram below shows the response of a typical low pass filter. Frequencies above the cut-off point are reduced in volume.
When resonance is added, the frequencies around the cut off point are boosted in volume.
In addition to the traditional Low Pass Filter type, there are also High Pass and Band Pass types. On Bass Station II, the Filter type is selected with the Shape switch .
A High Pass Filter is similar to a Low Pass Filter, but works in the “opposite sense”, so that frequencies below the cut-off point are removed. Frequencies above the cut-off point are passed. When the Filter Frequency parameter is set to zero, the filter is completely open and no frequencies are removed from the raw Oscillator waveforms.
When a Band Pass Filter is used, only a narrowband of frequencies centered around the cut-off point are passed. Frequencies above and below the band are removed. It is not possible to fully open this type of Filter, and allow all frequencies to pass.
In earlier paragraphs, the synthesis of the pitch and the timbre of a sound was described. The next part of the Synthesis Tutorial describes how the volume of the sound is controlled. The volume of a note created by a musical instrument often varies greatly over the duration of the note, according to the type of instrument.
For example, a note played on an Organ quickly attains full volume when a key is pressed. It stays at full volume until the key is released, at which point the volume level falls instantly to zero.
A Piano note quickly attains full volume after a key is pressed, and gradually falls in volume to zero after several seconds, even if the key is held.
A String Section emulation only attains full volume gradually when a key is pressed. It remains at full volume while the key is held down, but once the key is released, the volume falls to zero fairly slowly.
In an analogue synthesiser, changes to a sound’s character which occur over the duration of a note are controlled by a section called an Envelope Generator. Bass Station II has two Envelope Generators; one (Amp Env) is always related to the Amplifier, which controls the note’s amplitude – i.e., the volume of the sound - when the note is played. Each envelope generator has four main controls, which are used to adjust the shape of the envelope (often referred to as the ADSR parameters).
Adjusts the time it takes after a key is pressed for the volume to climb from zero to full volume. It can be used to create a sound with a slow fade-in.