in reverse to a standard resonance control.

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If you chose to install a capacitor here, you will need to cut the track below the component.
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We can see extra overtones are added to more familiar harmonics f, 2f, 3f, 4f, etc. Having just one modulating oscillator does not give spectacular results. Usually an FM synth will have 4 to 6 oscillators (‘operators’) that can be routed in a variety of ways (‘algorithms’) the carrier always being the last in the modulation chain. The technique is notoriously hard to implement, most musicians opting for the subtle tweaking of presets rather than full on programming. Though the technique does well at impersonating, electric pianos, bells and xylophones; acoustic pianos and guitars don't sound so convincing.


The original unit uses LM1458 op-amps.

Look for any shorts between tracks, or open circuits due to over etching.
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Transient and steady state responses of the instrument are emulated by a time dependant voltage envelope usually consisting of attack, decay, sustain and release sections whose respective durations can be controlled by the user.


see textParts listThis is a guide only.

The idea is that you can create an out-of-phase wave with the same frequency that will interfere with your basic waveform and "warp" it, removing the "perfect", or flat tone from it.

The array contains frequency modulation functions, you can use them however you'd wish, but always apply at the end - it contains your carrier wave, and acts as a way to normalize all of the interference you create by combining and modulating other waveforms.

Our final waveform is a modulated combination of our initial waveform interfering with two "quieter" versions of itself, one at 75% amplitude, and the other at 10% amplitude, both slightly out-of-phase with their parent.

Finally, we multiply , our dampened amplitude, by our final waveform to finish creating our note.

Step 4.

Parts needed will vary with individual constructor's needs.

However, each additional time you loop through the "noise" buffer, you average each point in the buffer with the one immediately preceding it, effectively filtering it.

See the to see if I've already answered the question.

Sound is the perceived vibration (oscillation) of air resulting from the vibration of a sound source (e.g. guitar sound board, speaker cone, hair dryer, etc). We can describe such regular (periodic) vibration in terms of the sum of simpler vibrations (harmonics). In other words any periodic oscillation and hence resulting waveform can be described in terms of the sum of its harmonics. Each harmonic being a simple sine wave (often called a pure tone) with it’s own respective frequency and amplitude. The graphs below shows how a simple pure tone varies with respect to time.

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We're first going to add noise to our signal, and then we're going to modulate it back to the carrier signal to normalize it back to within the range we'd like.