Abstract: A distributed system and a corresponding data processing method are disclosed, for decomposing audio signals including mixed audio sources. The system comprises at least one client terminal, a remote queuing module and at least one remote audio data processing module connected in a network. A client terminal stores source audio signal data, selects at least one signal decomposition type, uploads source audio signal data with data representative of the decomposition type selection to the queuing module, and downloads decomposed audio signal data. The queuing module queues uploaded source audio data and distributes same to data processing module(s). The queuing module also queues uploaded decomposed audio signal data and distributes same to client terminal(s). An audio data processing module processes distributed source audio data into decomposed audio signal data according to the type selection, and uploads decomposed audio signal data to the at least one remote queuing resource.

Abstract: Various embodiments may include systems and non-transitory computer-readable media for assigning three-dimensional spatial data to sounds and audio files. In one embodiment, a method can include receiving at least one audio signal, receiving sonic spatial data via a foot pedal, associating the sonic spatial data with the at least one audio signal, associating the at least one audio signal and sonic spatial data with a time code generated by a video camera, storing the sonic spatial data, the at least one audio signal, and time code in an encoded sound file, and directing playback of the encoded sound file.

Abstract: A method and device for obtaining a sound, wherein an information representative of a speed of a first object moving on a first surface is obtained. The obtained speed information is used with one or more reference sounds to obtain the sound. The one or more reference sounds are associated with a determined speed of displacement of a second object moving on a second surface, the first surface being different from the second surface.

Abstract: A method of performing audio synthesis is disclosed. An audio event is input to an audio algorithm along with associated parameters including source sample data. An interpolation function is provided and the source sample data are interpolated to generate one or more interpolated samples based on the source sample data. A filter function is provided and at least one of the interpolated samples is filtered to generate a filtered sample. A gain function is provided and the filtered sample is processed to generate a gained sample. At least one of the interpolation, filter, and gain functions include outputting an earlier-calculated value along with an estimated difference value in lieu of calculating a new value.

Abstract: A system and method is disclosed teach how to synthesizing audio. It allows specification of a musical sound to be generated. It synthesizes an audio source, such as noise, using parameters to specify the desired frequency slit spacing and the desired noise-to-frequency band ratio, then filtering the audio source through a sequence of filters to obtain the desired frequency slit spacing and noise to frequency band ratio. It allows modulation of the filters in the sequence. It outputs musical sound.

Abstract: A system and method is disclosed teach how to synthesizing audio. It allows specification of a musical sound to be generated. It synthesizes an audio source, such as noise, using parameters to specify the desired frequency slit spacing and the desired noise-to-frequency band ratio, then filtering the audio source through a sequence of filters to obtain the desired frequency slit spacing and noise to frequency band ratio. It allows modulation of the filters in the sequence. It outputs musical sound.

Abstract: A noise feedback coding (NFC) system and method that utilizes a simple and relatively inexpensive general structural configuration, but achieves improved flexibility with respect to controlling the shape of coding noise. The NFC system and method utilizes an all-zero noise feedback filter that is configured to approximate the response of a pole-zero noise feedback filter.

Abstract: A method for increasing ring tone volume is provided. The method includes steps of: reading an audio file which is set as a current ring tone; determining whether the ring tone is a MP3 audio file or a musical instrument digital interface (MIDI) audio file; adjusting frequencies by using an equalizer technique to increase volume of the ring tone if the ring tone is the MP3 audio file; adjusting a volume level of the ring tone to be the highest volume level, and adjusting timbre of the ring tone to increase the ring tone volume by simulating a musical score of the ring tone by using different instruments if the ring tone is the MIDI audio file. A related system is also provided.

Abstract: Resonators generate resonance sound data of given resonant frequencies based on sound signal data corresponding to a pitch of a key. Each resonator includes a delay circuit which delays an input based on delay time data, a second adder which adds an output from the delay circuit to input sound signal data, and a low-pass filter which performs filtering depending on a filter control signal on the output from the second adder. An output from the low-pass filter is input to the delay circuit, and a first adder adds outputs from the resonators. A storage device stores a coefficient table containing items of delay time data to be provided to the resonators, and the items of the delay time data do not match with frequencies corresponding to pitches of keys.

Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.

Abstract: 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.

Abstract: This invention provides a signal processing and signal synthesis technique from a family of signal processing and signal synthesis techniques designed to readily interwork or be used individually in creating new forms of rich musical timbres. A plurality of audio signal delays, each with high resonance positive feedback, distortion characteristics, and selectable delay times corresponding to a desired resonant frequency, provide twang and resonance synthesis for moments of sparkle or vibrantly-responsive ongoing backdrops. The selectable delay times may match a musical scale or other resonant frequency distribution. Delay, feedback, and signal processing characteristics and parameters may be recalled from stored program control or modulated in real-time by arbitrary control signals, including those derived from the original input signal. The invention may be used individually or in conjunction with other signal processing and signal synthesis techniques in creating new forms of rich musical timbres.

Abstract: An acoustic drum set amplifier specifically calibrated for each instrument within a drum set, specifically a snare drum, a bass drum, a bongo drum, tom-tom drum, a hi-hat, ride cymbal and crash cymbals, for optimizing the sound frequency of the instrument. In operation, an artist using the device might be able to filter, blend, and manipulate an electro-acoustic signal associated with an acoustic drum instrument, either captured by a microphone or generated by an electronic drum module to the artist's preferences while monitoring the sound directed towards an audience at a live performance.

Abstract: A system for control signal generation using detected dynamic characteristics of components of an incoming electronic signal. Fixed or adjustable bandpass filters are coupled to signal parameter measurement elements. Each filter isolates a particular overtone component from the incoming electronic signal for isolated signal parameter measurement. Pitch and amplitude of a plurality of overtones may be individually measured. A control signal processor generates one or more outgoing control signals based upon one or more isolated overtone parameter signals. The control signal processor may use these to generate individual associated output signals, subject to optional mathematical operations and warpings, or combine measured parameters of several overtones via mathematical operations to other output signals.

Abstract: Electromagnetic (EM) musical instrument systems and methods for configuring and using the instrument are provided. The EM musical instrument could include an EM resonant body to receive an excitation signal. The excitation signal varies the associated sound produced by the EM musical instrument. The EM musical instrument could also include a first circuit to convert an audio signal into an associated electrical signal, a second circuit to convert the electrical signal to an associated EM signal, and a third circuit to apply the EM signal to a body of an EM musical instrument to produce an associated sound. The EM musical instrument could include a body having a top plate of EM resonant material. The instrument could include a center medium defining a cavity coupled to a sound hole of the top plate and a bottom plate of EM resonant material coupled to the center medium. The instrument could further include a port coupled to the body and configured to apply the received excitation signal to the body.

Abstract: An electromagnetic (EM) musical instrument system includes an excitation signal generator to generate an excitation signal. The system also includes an excitation signal applier coupled to the excitation signal generator. The excitation signal applier receives the excitation signal and transmits a corresponding excitation signal to an EM musical instrument. The excitation signal applier is further configured to modulate the corresponding excitation signal to produce audio effects. An EM musical instrument, in response to receiving the corresponding excitation signal, generates an EM output signal associated with a produced sound. The system also includes an EM output receiver coupled to the EM musical instrument to receive the EM output signal.

Abstract: A method for generating electromagnetic (EM) music through excitation of an EM Musical Instrument is provided. An input queue, comprised of three sub-queues, receives three input audio queues. An EM excitation signal is generated based on the received input audio queues. The EM excitation signal is applied to the EM Musical Instrument. A measured response is generated, based on a response of the EM Musical Instrument to the EM excitation signal. Various time marks are marked in the measured response. Selected portions of the measured response are discarded based on particular time marks, generating a newly measured sample set. The newly measured samples are joined to a previously measured sample set of an output queue based on alignment of time marks. Various alternate embodiments and supporting systems are also provided.

Abstract: An electronic music instrument includes a musical-tone control that generates operation information of keys and a damper pedal to serve as musical-tone control information; a musical-tone generator simultaneously generating a plurality of musical tones according to the musical-tone control information; a resonance-tone generator that includes resonant circuits equal in number to harmonic signals of musical-tone signals that can be generated, for generating a resonance tone with the resonance circuits using a musical tone generated by the musical-tone generator as an input signal to each resonance circuit; and a resonance-tone mixer that multiplies the resonance tone generated by the resonance-tone generator by a predetermined degree according to the musical-tone control information, for adding the product to a musical tone input from the musical-tone generator, and outputting the sum.

Abstract: The present synthesizer generates an underlying spectrum, pitch and loudness for a sound to be synthesized, and then combines the underlying spectrum, pitch and loudness with stored Spectral, Pitch, and Loudness Fluctuations and noise elements. The input to the synthesizer is typically a MIDI stream. A MIDI preprocess block processes the MIDI input and generates the signals needed by the synthesizer to generate output sound phrases. The synthesizer comprises a harmonic synthesizer block (which generates an output representing the tonal audio portion of the output sound), an Underlying Spectrum, Pitch, and Loudness (which takes pitch and loudness and uses stored algorithms to generate the slowly varying portion of the output sound) and a Spectral, Pitch, and Loudness Fluctuation portion (which generates the quickly varying portion of the output sound by selecting and combining Spectral, Pitch, and Loudness Fluctuation segments stored in a database).

Abstract: 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.

Abstract: A music sound generator imitates released key string vibration sounds (RKSV) and cabinet resonances of an acoustic piano. When a key is pressed, waveform is readout from a normal music sound waveform memory 15. The normal sound waveform is inputted into a filter 21, and inputted into a filter 22 through a band-pass filter 33. Output waveform of a cabinet resonance waveform memory 17 is inputted into a filter 23 when the key is pressed. Outputs of the filters are synthesized by an adder 27 through multipliers 24 through 26. When the key is pressed, a cut-off frequency of the filter 22 is sufficiently low and RKSV is not generated. If the damper is not on when key-releasing, the cut-off frequency is returned to normal and RKSV is generated. A level controller 32 attenuates the RKSV and the cabinet resonance for a longer time than the normal music sound.

Abstract: 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.

Abstract: In a voice synthesizer, an envelope acquisition portion obtains a spectral envelope of a reference frequency spectrum of a given voice. A spectrum acquisition portion obtains a collective frequency spectrum of a plurality of voices which are generated in parallel to one another. An envelope adjustment portion adjusts a spectral envelope of the collective frequency spectrum obtained by the spectrum acquisition portion so as to approximately match with the spectral envelope of the reference frequency spectrum obtained by the envelope acquisition portion. A voice generation portion generates an output voice signal from the collective frequency spectrum having the spectral envelope adjusted by the envelope adjustment portion.

Abstract: A method for digital simulation of a non-linear interaction between an excitation source and a wave in a resonator, and is particularly applicable, to real-time synthesis of digital signals representing an oscillating phenomenon such as the sound produced by a musical instrument. The invention is characterized in that it consists in calculating the digital signals from equations whereof the solution corresponds to the physical representation of the phenomenon to be simulated which is expressed, each time and in each point of the resonator, by a relationship of impedance or of admittance between two variables representing the effect and the cause of the phenomenon and in directly transcribing the equation of the impedance or of the admittance in the form of a linear filter including delays, so as to produce a non-linear interaction between the two variables of the impedance or admittance relationship.

Abstract: Resonators generate resonance sound data of given resonant frequencies based on sound signal data corresponding to a pitch of a key. Each resonator includes a delay circuit which delays an input based on delay time data, a second adder which adds an output from the delay circuit to input sound signal data, and a low-pass filter which performs filtering depending on a filter control signal on the output from the second adder. An output from the low-pass filter is input to the delay circuit, and a first adder adds outputs from the resonators. A storage device stores a coefficient table containing items of delay time data to be provided to the resonators, and the items of the delay time data do not match with frequencies corresponding to pitches of keys.

Abstract: A signal processing apparatus and method is disclosed by which signal components of different musical intervals of an audio signal can be obtained with a small amount of arithmetic operation. A re-sampling section re-samples an audio signal inputted thereto with a sampling frequency equal to a power-of two times a frequency at a boundary of an octave. An octave dividing block divides the audio signals outputted from the re-sampling section into audio signals of octaves and outputs the resulting audio signals to respective band-pass filter banks. Each of the band-pass filter banks includes 12 band-pass filters and extracts and outputs audio signals of 12 musical intervals from the audio signals of one octave. The invention can be applied to a signal processing apparatus which analyzes, for example, signal components of musical intervals of an audio signal.

Abstract: Disclosed herein is a signal processing apparatus, including: removal means for removing, from a sound signal in the form of a stereo signal, a center component which is a component of sound positioned at the center between the left and the right; extraction means for extracting, from the sound signal from which the center component is removed, first feature quantity representative of characteristics of sounds of different tones of the 12-tone equal temperament within a predetermined range; and decision means for deciding a chord within the predetermined range using the first feature quantity.

Abstract: A system for the extraction of vocal content from a musical track. A hardware platform and a multimedia software application are provided that are suitable for use in playback of the musical track. A filter software unit then dynamically intercepts the musical track in the hardware platform prior to its use there in playback. The filter software unit suppresses the vocal content from the musical track to create an instrumental track, and it passes the instrumental track on for use in playback immediately or after some period of storage.

Abstract: Disclosed herein is a signal processing apparatus, including: extraction means for extracting feature quantity indicative of characteristics of sounds of different tones of the 12-tone equal temperament within a predetermined range of a sound signal, the sounds being arranged in an order of a musical scale with reference to a reference sound which is a sound of a predetermined tone; and root decision means produced in advance by learning regarding the feature quantity for deciding whether the reference sound is a root from the feature quantity.

Abstract: In an electronic musical instrument, when a musical tone delay time T1 read from a delay time table TBL depending on a velocity generated by a key being pressed has elapsed, a hammer string-striking sound is generated from musical tone waveform data selected in correspondence with the velocity. At the same time, when an impact delay time T2 read from the delay time table depending on the velocity has elapsed, a shelf board impulsive sound is generated from impact waveform data. The generated hammer string-striking sound and shelf board impulsive sound are added and outputted. As a result, a relationship between the shelf board impulsive sound and the hammer string-striking sound is simulated, and tone variations of an acoustic piano are reproduced.

Abstract: The invention relates to an apparatus, and an associated method, in a communication system that is tolerant to interference. The received signal is divided into a real and an imaginary part, and the noise and interference is modeled as a filtered process. The invention solves the interference problem by utilizing the inherent signal structure.

Abstract: An apparatus and method simulates more accurately the natural statistics of a physical reverberation process. A new filter design is provided having a comb shaped group delay. Gain minimums at a plurality of frequencies are combined with a delay line to create a constant reverberation time independent of frequency while allowing for temporal spreading. In addition, the connection topology between the plurality of energy transmission networks is temporally randomized to facilitate energy distribution within the reverberation apparatus. Both the temporal and spectral responses are actively changed on each iteration of the energy recirculation. By making the response have a high echo density and a lack of spectral coloration in the decay, the illusion of a natural process is enhanced.

Abstract: A synthesiser to generate sounds often in a musical environment but also in a speech environment. Such synthesis of sounds is often accompanied by at least one mode of filtering or many modes. Hitherto, the application of such filtering has involved a control for each filter. This may be by way of a lever, toggle or rotatable knob. Thus, a user when applying the filters is required to adjust each control independently of the other. This enhances the complexity and a large amount of skill is involved in accurately selecting the required amount and types of filtering.

Abstract: A method is provided for designing an acoustic correction filter applicable to a stringed instrument, which is composed of a string member operable to undergo a vibration, a support member for supporting the string member, a body member responsive to the vibration transmitted through the support member for generating a natural sound and a mute attachment for muting the natural sound. The acoustic correction filter is operable when the natural sound is muted by the mute attachment for filtering a signal derived from the vibration so as to create an artificial sound instead of the muted natural sound.

Abstract: This invention provides a signal processing and signal synthesis technique from a family of signal processing and signal synthesis techniques designed to readily interwork or be used individually in creating new forms of rich musical timbres. A real-time controllable mix of a plurality of multiple output frequency-reduced signals and the cross-products of these signals provides rich animated harmonically partitioned spectra. Mixing parameters may be recalled from stored program control or modulated in real-time by arbitrary control signals, including envelope generators, low-frequency oscillators, and those derived from an original input signal, and specifically including the amplitude of an original input signal. The real-time controllable mix may have multiple outputs and be subjected to further signal processing and multiple-channel amplification.

Abstract: A synthesiser to generate sounds often in a musical environment but also in a speech environment. Such synthesis of sounds is often accompanied by at least one mode of filtering or many modes. Hitherto, the application of such filtering has involved a control for each filter. This may be by way of a lever, toggle or rotatable knob. Thus, a user when applying the filters is required to adjust each control independently of the other. This enhances the complexity and a large amount of skill is involved in accurately selecting the required amount and types of filtering.

Abstract: Multipliers respectively multiply a tone signal supplied from a sound source and a tone signal corrected by a filter according to hearing correction characteristics by balance control coefficients generated by a control coefficient generator based on tone volume control data, and an adder adds the products to adjust the tone volume balance. A multiplier multiplies the tone signal output from that adder by a tone volume control coefficient generated by the control coefficient generator based on the tone volume control data to adjust the tone volume. Frequency characteristics for correcting hearing characteristics are given to the tone signal in accordance with the tone volume of tones to be generated, thus maintaining tone volume balance upon hearing among tone colors of tones to be generated by an electronic musical instrument.

Abstract: An average is calculated of every predetermined number of sample amplitude values of a sound signal from an external sound source, and the respective averages are output as a time-series of average level information. On the basis of the average level information, each available section of the sound signal is detected where there appears to be a musical sound. On the basis of degrees of inclination in the average level information within the available section, stable sections are detected for detection of same-waveform sections. On the basis of the signals within the stable sections, a steady section is detected which corresponds to a note. A time-varying band-pass filtering operation is then performed on the sound signal, and detection is made of a plurality of periodic reference points of the sound signal.

Abstract: A system and a method of measuring, decomposing, processing and uniquely recombining forces and vibrations acting on stringed musical instruments (SMI). The system utilizes a digital signal processor and reproduces the musical sound characteristics of an acoustic instrument into high fidelity electrical signals for amplification, processing and/or filtering and reproduction of musical sounds by uniquely exploiting, through measurements and subsequent signal processing, the vector nature of string excitation forces (SEF) and body vibrations of stringed musical instruments. A signal processing system of the current invention also utilizes a plurality of sensors, each responsive to at least one of force, displacement, velocity or acceleration indicative of the vibrational energy of the strings, to produce a sensor signal vector, which is then processed and transformed by a plurality of re-creation filters into a transformed signal vector, and then resynthesized into an output signal.

Abstract: A musical sound signal synthesizer with a simple construction for synthesizing complex musical sound waveform signals including many harmonics in a peculiar form. A phase information generator 11 supplies phase information x with a sawtooth wave changing according to a phase of the musical sound to be generated via a calculator 12 to a sine wave table 13, from which output waveform information y is read out. The waveform information y is fed back to the calculator 12 via an absolute value transformer 14, a low pass filter 15 and a gain controller 16. The output waveform information y outputted from the sine table 13 is transformed nonlinearly, which is fed back to the phase information x, which makes a complex change. As a result thereof, the waveform, which is represented by waveform information outputted from the sine wave table 13, becomes peculiar for a musical sound signal synthesized by phase or frequency modulation technique.

Abstract: Multipliers respectively multiply a tone signal supplied from a sound source and a tone signal corrected by a filter according to hearing correction characteristics by balance control coefficients generated by a control coefficient generator based on tone volume control data, and an adder adds the products to adjust the tone volume balance. A multiplier multiplies the tone signal output from that adder by a tone volume control coefficient generated by the control coefficient generator based on the tone volume control data to adjust the tone volume. Frequency characteristics for correcting hearing characteristics are given to the tone signal in accordance with the tone volume of tones to be generated, thus maintaining tone volume balance upon hearing among tone colors of tones to be generated by an electronic musical instrument.

Abstract: An electric instrument amplifier emulates an audio characteristics of a traditional vacuum-tube type amplifier. The electric instrument amplifier is formed in a single housing for amplifying an audio signal from an electric instrument. The electric instrument amplifier includes an A/D (analog-to-digital) converter for converting a first analog signal from the electric instrument to a digital signal, a digital signal processing circuit for processing the digital signal to add an intended effect to the digital signal, a D/A (digital-to-analog) converter for converting the digital signal processed by the digital signal processing circuit to a second analog signal, a tube amplifier having at least one vacuum-tube for amplifying the second analog signal, and a virtual power circuit formed with semiconductor devices for amplifying or attenuating an audio signal produced by the tube amplifier.

Abstract: A pitch shifter capable of shifting an acoustic signal in pitch to an arbitrary level with a high degree of accuracy without any change in reproduction time, and also sufficiently reducing high-frequency distortion without being increased in size or speeded-up is provided. Stored in a filter coefficient string storage 6, four filter coefficient strings corresponding to four sub-filters produced through polyphase decomposition of a low-pass filter for 4-fold oversampling. Filter coefficient string selectors 5a and 5b select, based on the first and second bits of the decimal part of each of read addresses generated by the read address generators 4a and 4b, respectively, any one of the four filter coefficient strings stored in the filter coefficient string storage 6. Filter operation units 2a and 2b receive paired sound data strings, and carry out a filter operation by using the filter coefficient strings selected by the filter coefficient string selector 5a and 5b, respectively.

Abstract: A tone synthesis system employs digital filtering methods to enable the practical usage of unstable filter elements in a real-time synthesis model. Truncated infinite impulse response (TIIR) filters are used to approximate portions of the reflection impulse response of an acoustic horn such as a trumpet bell. Methods for resetting filter state enable the use of internal unstable filter poles. Similar state resetting methods enable the use of unstable one-pole filters in the scattering junction formed between two conical acoustic bores. High quality tone synthesis can be achieved without the necessity of a complicated filter representing large sections of the bore of a woodwind instrument.

Abstract: This application describes a method for musical signal analysis and synthesis using global waveform fitting (GWF). GWF uses a sinusoidal model with quadratic phase. GWF tries to fit the entire data record (signal waveform) to the assumed signal model. The model parameters obtained by GWF have clear physical interpretations or meanings.

Abstract: A music apparatus has a set of controls manually operable to input a performance event, a waveform memory storing an original waveform sample composed of a series of digital values sequentially readable from a default start address, and a processor for executing a tone generating process in response to the performance event.

Abstract: Physical model tone generator, which includes a loop section with a signal delay element, generates a driving signal by modifying a loop output signal from the loop in accordance with a performance parameter such as a bowing velocity, and introduces the generated driving signal to the loop. This way, the tone generator generates a tone signal with a characteristic controlled by the performance parameter, in a pitch period corresponding to a time delay of the loop. To generate the driving signal, a nonlinear conversion section performs a nonlinear conversion on an input signal based on the loop output signal and performance parameter. The conversion section switches an input-output characteristic, to be used for converting the input signal, into one of at least first and second input-output characteristics in accordance with intensity of a signal based on the loop output signal or the input signal, so that a desired nonlinear conversion characteristic is achieved.

Abstract: The present invention describes methods and means for estimating the time-varying spectrum of an audio signal based on a conditional probability density function (PDF) of spectral coding vectors conditioned on pitch and loudness values. Using this PDF a time-varying output spectrum is generated as a function of time-varying pitch and loudness sequences arriving from an electronic music instrument controller. The time-varying output spectrum is converted to a synthesized output audio signal. The pitch and loudness sequences may also be derived from analysis of an input audio signal. Methods and means for synthesizing an output audio signal in response to an input audio signal are also described in which the time-varying spectrum of an input audio signal is estimated based on a conditional probability density function (PDF) of input spectral coding vectors conditioned on input pitch and loudness values.

Abstract: An electronic musical instrument, which is configured to simulate one or more instruments, provides a sound source, a display unit and a control portion. Herein, under control of the control portion, the display unit displays the predetermined graphic pattern corresponding to the tone-generation mechanism, i.e., tone-generation algorithm of an instrument to be simulated. The sound source contains a drive portion, a tone-generation portion and a resonance-radiation portion, each of which further contains a digital signal processor (DSP). All of these portions function to create a digitized musical tone signal corresponding to a simulated sound of the instrument by combining operation data outputted from the DSPs. By controlling the display unit, it is possible to arbitrarily vary the contents of the displayed tone-generation algorithm, so that the performer can easily and freely perform a music on this electronic musical instrument.

Abstract: A filtering apparatus for an electronic musical instrument includes at least a digital filter. The digital filter has a frequency characteristic determined by a filtering parameter and performs a filtering operation on a musical tone signal inputted thereto. In addition, the filtering parameter is changed in accordance with a keycode signal representing a tone pitch of the musical tone signal. Thus, the frequency characteristic is varied in accordance with a keycode signal.