Electronic data processing apparatus and method for sound...

Data processing: speech signal processing – linguistics – language – Speech signal processing – Synthesis

Reexamination Certificate

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C704S265000, C704S258000

Reexamination Certificate

active

06208969

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to an electronic data processing system and method for sound synthesis using sound samples, and particularly to such a system or method using transfer functions.
2. Discussion of the Related Art
Most conventional electronic musical instruments use so-called wavesamples of actual musical instruments as building blocks for synthesizing simulations of the instruments that sound realistic. The electronic instruments must switch or fade between multiple time-domain sample waves, which must be sufficiently numerous to encompass an entire keyboard and to provide adaptability for various rates of sound change. The resulting stored sample sets have sizes in the megabyte range.
Alternatives for avoiding the large amount of data in sampled sets include physical modeling or additive synthesis. Additive synthesis can, for example, interpolate very simply between loud and soft sounds for a sound in between. Nevertheless, such additive synthesis becomes prohibitively expensive in its use of logic because of the addition of many sinusoids (up to 64 per voice) and the complexity of controlling the amplitudes of the constituent sinusoids.
SUMMARY OF THE INVENTION
The invention is based on the recognition that the best of both worlds of sampling and synthesis can be obtained.
According to one aspect of the invention, a method of additive sound synthesis includes the computer-based steps of reading stored data that include transfer functions representing harmonic data derived from recorded sounds, and combining the read transfer functions to interpolate between them. These steps produce a resultant transfer function that corresponds to a sound spectrally interpolated between the harmonic data. The computer converts the resultant transfer functions to time domain signals, and peripheral apparatus generates sound from the time domain signals.
According to a preferred implementation of the method of the invention, transfer functions to be combined are read in respective first and second processes. Preferably, the stored transfer functions include Chebyshev polynomial-based transfer functions. Advantageously, when the transfer functions in the first and second processes represent harmonic data having different timbre, the method yields timbre morphing.
Further, according to a related feature of the invention, anharmonic spectra are generated. To a plurality of parallel processes using the method of the invention is added the step of driving the reconversion of the transfer functions by sinusoids having frequencies that are not harmonically related.
According to another feature of the invention, the method operates very efficiently in real time because the transfer functions are prepared from the sound samples in advance of the real time application.
According to another feature of the method of the invention, useful in producing speech sibilants or noise envelopes of instruments, for example, selected noise spectra are supplied in the conversion step for modulating the base frequency of the driving sinusoid. Alternatively or in addition, according to this feature, a band-limited frequency modulation signal modulates the sinusoid that drives the conversion step.
According to a second aspect of the invention, an electronic data processing system for sound synthesis includes an electronic memory storing a plurality of frames of data that include sequences or collections of transfer functions representing harmonic data derived from recorded sounds. A transfer function reader reads from the memory the transfer functions and supplies them to apparatus for combining pairs of transfer functions for interpolation between them. Each of the pairs of transfer functions represent adjacent data points with respect to some parameter of the recorded sound samples. Therefore, the interpolated transfer function represents an interpolation with respect to that parameter of the recorded sound samples. Excitation apparatus converts the resultant transfer functions to time domain signals representative of the sound to be synthesized. A speaker or other transducer generates sound from the time domain signal.
According to a preferred implementation of the system of the invention, the transfer functions include Chebyshev polynomial-based transfer functions. Optionally, compression of the stored data may be obtained by storing those transfer functions as the pertinent polynomial coefficients only and regenerating the full transfer functions from the stored coefficients as needed by the interpolation process.
According to a feature of the system of the invention, related sequences of transfer functions or coefficients are read into parallel synthesis paths for interpolation between different sound qualities.
According to other features of the invention, the excitation apparatus supplies a plurality of driving sinusoids of selected frequency relationships, or band-limited noise modulation of a driving sinusoid that is also involved in the reading steps of the method. In one implementation, an external instrument or sound source for which the waveform has been filtered to a band close to its fundamental frequency could take the place of the excitation oscillator. Thereby, the external instrument or sound source could supply an excitation source for synthesizing the sound of another instrument.


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