Dynamic Piano Filter Coefficients for Harmonic Tone Color Control
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Solution Overview
Problem
Existing electronic musical instruments struggle to replicate the dynamic tone color changes of an acoustic piano, particularly in controlling the amplitude ratio of harmonic tones, due to limitations in filter characteristics and the PCM method, which fixes tone colors and fails to attenuate higher order harmonics naturally over time.
Innovation Solution
A filter device with a filter coefficient outputting means that uses a parameter table to generate filter coefficients based on frequency and signal strength, allowing for dynamic filter processing that mimics the tone color changes of an acoustic piano by adjusting filter characteristics such as turnover frequency and gain, approximated using a low-order IIR filter.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If a conventional filter circuit with cutoff frequency fc and selectivity Q is used to change tone color, then the filter characteristics can be adjusted, but it is extremely difficult to realize tone color change equal to the tone color change of the acoustic piano
Solution Approach 1:
The patent changes the parameter basis from conventional (fc, Q) to piano-specific parameters (turnover frequency f0, maximum attenuation level Amax, maximum enhancement level Amax). This parameter transformation enables the filter to accurately replicate piano tone color changes by mapping acoustic piano characteristics to digital filter parameters, resolving the contradiction between adaptability and precision.
Solution Approach 2:
The patent copies the spectral envelope characteristics of acoustic piano by measuring and reproducing its frequency response. The filter characteristics are designed to match the piano's natural attenuation and enhancement patterns, allowing electronic instruments to replicate piano tone color changes with high fidelity.
2Quantity of substance
If the PCM method is used to store waveform data, then the waveform memory capacity is decreased, but the color tones become fixed and monotonized without natural attenuation of higher order harmonics over time
Solution Approach 1:
The patent introduces dynamic filter characteristics that change over time to replicate the natural decay of piano harmonics. The turnover frequency and attenuation levels are not fixed but evolve during sound playback, creating temporal variation in tone color that mimics acoustic piano behavior while using efficient PCM storage.
Solution Approach 2:
The patent pre-calculates and stores filter parameters (turnover frequency, attenuation levels) in lookup tables based on measured piano characteristics. During playback, these pre-computed parameters are applied to rapidly adjust filter behavior without real-time computation, maintaining both efficiency and temporal dynamics.
3Measurement precision
If a filter uses turnover frequency f0, maximum attenuation level Amax, and maximum enhancement level Amax as parameters, then the filter characteristics can be transformed to match acoustic piano, but the turnover frequency moves according to parameter change
Solution Approach 1:
The patent implements a feedback mechanism where the turnover frequency is automatically adjusted based on the fundamental frequency of the played note. The system continuously monitors the input signal and adapts the filter parameters accordingly, ensuring that the turnover frequency moves in sync with the musical context without requiring manual intervention.
Solution Approach 2:
The filter system serves itself by automatically adjusting its parameters based on the input signal characteristics. The turnover frequency and attenuation levels are self-modulated according to the fundamental frequency and velocity of the played notes, eliminating the need for external control and simplifying the user interface.
Data Source
AI summary
In a filter device, a filter coefficient calculation circuit has a parameter table. The parameter table stores a plurality of sets of filter coefficients associated with a first parameter based on a frequency and a second parameter based on respective plurality of levels representing a degree of attenuation or enhancement of a gain of a filter in filter characteristics. The filter coefficient calculation circuit extracts a set of filter coefficients from a parameter table with the use of the first parameter and the second parameter determined according to a frequency and a strength of a musical sound signal, and outputs the extracted set of filter coefficients to the filter. The filter circuit performs filter processing for the musical sound signal, based on the filter characteristics determined by the set of filter coefficients.