Abstract: A method and apparatus for generating a simulated reverberation characteristic in an input audio signal having a plurality of channels, each channel transmitting time-synchronized data elements. The method and apparatus imparts a unique delay to each element of a set of time-synchronized data elements and thereafter delays the data element set through a series of delay elements. Each of a plurality of early reflection taps reads in and outputs a unique linear combination of the delayed data elements. The delayed elements are selectively input to at least one summing filter incorporating a plurality of comb filters. The output of the at least one summing filter is selectively combined based upon a summing scheme to provide a plurality of sum output elements. The sum output elements are combined with the early reflection tap output and a preselected combination of the time-synchronized data elements of the input signal to create a simulated reverberation signal.

Abstract: A tone to which resonance characteristics have been added to is repetitively delayed so as to reverberate. The delay of the tone is determined based on the resonance characteristics, and a modified tone in which the reverberation and resonance characteristics are related to one another is generated.

Abstract: In accordance with a kind of effect, an effect adding section (38) adds an original microphone signal with an effect signal and outputs the added signal, or outputs an effect signal in place of the original microphone signal. A mixing section (42) mixes the effect signal and the original microphone signal at a variable mixing ratio. The mixed signal is supplied to an echo signal generating circuit (20) to generate an echo signal. An output signal of the effect adding section (38) and the echo signal output from an echo adding section (40) are mixed by a mixing circuit (18). In the case in which both an effect and an echo are added such that the microphone signal and the effect signal are added and then output, the mixing section (42) reduces the mixing ratio of the effect signal, whereby the effect is prevented from being excessively increased.

Abstract: A music apparatus is constructed for introducing a music sound into an acoustic space, and is equipped with an acoustic feedback system for introducing a reflection sound into the acoustic space. In the music apparatus, an input device sequentially provides performance information. A sound source generates a music signal representative of the music sound in response to the performance information provided from the input device. A plurality of channels of the acoustic feedback system are provided in the music apparatus, and are spatially separate from each other.

Abstract: In each sampling period, sample data and a count value WA of a ring buffer counter 11 are supplied to a RAM 202 through a bus control circuit 14 as write data and a write address. Each cache block stored in a data buffer DB.sub.0 and so on is used for executing an FIR filtering operation. As an amount of unused sample data in the cache block is decreased, subsequent cache blocks used for a subsequent FIR filtering operation are supplemented to the data buffers DB.sub.0 and so on by the cache counter units CC.sub.0 and so on.

Abstract: Early reflection and reverberation processing using a decimating filter simulates the high frequency attenuation of an actual physical and acoustical environment and advantageously reduces the memory storage and computational burden of the early reflection and reverberation processing method. A method of generating a reverberation effect in a sound signal includes decimating the sound signal in the sound signal path and forming an early reflection sound signal from the decimated sound signal. The early reflection sound signal has a reduced sample rate an attenuated high frequency components in comparison to the sound signal.

Abstract: A digital wavetable audio synthesizer is described. The synthesizer can generate up to 32 high-quality audio digital signals or voices, including delay-based effects, at either a 44.1 KHz sample rate or at sample rates compatible with a prior art wavetable synthesizer. The synthesizer includes an address generator which has several modes of addressing wavetable data. The address generator's addressing rate controls the pitch of the synthesizer's output signal. The synthesizer performs a 10-bit interpolation, using the wavetable data addressed by the address generator, to interpolate additional data samples. When the address generator loops through a block of data, the signal path interpolates between the data at the end and start addresses of the block of data to prevent discontinuities in the generated signal. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data.

Abstract: On the basis of localization control data, a sound image localization controlling circuit reproduces input audio signals via a plurality of speakers after having applied predetermined delay-involving signal processing to the audio signals, to thereby perform sound image localization processing to localize sound images of direct sounds in a desired range including an area outside a space surrounded by the speakers. The audio signals are also supplied to a sound field controlling circuit after having been delayed by a predetermined time. The sound field controlling circuit performs operations to convolute the audio signals with reflected sound parameters so as to generate reflected sounds. The output signals of the sound image localization controlling circuit and sound field controlling circuit are fed to adders each adding together the signals of same channel. The resultant added signals are then sent to the speakers in a listening room for audible reproduction.

Abstract: An electronic musical instrument has a processing circuit for adding a reverberation tone after volume control using expression data generated upon operation of an expression pedal to generate tone data which does not vary the attenuation characteristics of the reverberation tone by operation of the expression pedal. The volume control and reverberation tone addition are implemented by a digital signal processor. Auditory sense correction of the volume control is also done by exponentially transforming expression data.

Abstract: A music sound correcting apparatus for correcting a music sound signal entered thereinto to selectively supply the corrected music sound signal to a speaker and a headphone, is provided with a jack into which a plug of the headphone is inserted, and a detector for detecting whether or not this plug of the headphone is inserted into the jack. When the detector detects that the plug is not inserted into the jack, a frequency of the entered music sound signal is changed so as to be properly reproduced by the speaker. On the other hand, when the detector detects that this plug is inserted into the jack, the entered music sound signal is corrected so as to be properly reproduced by the headphone in such a manner that a sound image produced based upon the entered music sound signal is localized to a predetermined position. As a result, preferable music sound can be reproduced even when the speaker is used and the headphone is employed irrespective of a small amount of hardware.

Abstract: A sound or music synthesizer includes a reverberation simulator having a substantially reduced volatile storage, random access memory, or buffer size in comparison to conventional reverberation simulators by decimating the sound signal prior to applying the sound signal to a reverberator and then interpolating the sound signal generated by the reverberator to restore the sample frequency. The substantial reduction in buffer size enables the usage of the reverberator in low-cost, reduced size and single-chip environments.

Abstract: A silent violin converts vibrations of strings to an electric signal, and the electric signal is supplied to two delay circuits and, thereafter, two multipliers so as to produce a left sound signal and a right sound signal different in delay time and magnitude; when a headphone produces stereophonic sounds from the left/right sound signals, the player feels the source of stereophonic sounds to be at a certain point on the silent violin same as an acoustic violin.

Abstract: In a karaoke vocal adaptor 100, voice signals to which a harmony effect is imparted by the harmony effect imparting circuit, and voice signals to which the harmony effect is not imparted are output from wet voice output terminals 121 and 122, and dry voice output terminals 111 and 112, respectively. In a karaoke amplifier 200, an echo is imparted only to voice signals input through dry voice input terminals 221 and 222, by an echo imparting unit 233, the voice signals are then mixed with voice signals input through wet voice input terminals 241 and 242, and musical signals input through music input terminals 251 and 252, and the mixed signals are output.

Abstract: In a reverberation effect imparting apparatus, sampled data of an input signal is written into a RAM 23 and the data is read out after an elapse of a predetermined time period, thereby imparting a reverberation effect. A storage area of the RAM 23 is divided into an area 1 for storing waveform data and an area 2 for imparting a reverberation effect. During evaluation, the waveform data is sequentially read out from the area 1 by using an absolute address, and sampled data of the waveform data is written into and read out from the area 2 by using a relative address, thereby imparting a reverberation effect.

Abstract: An audio signal processing system including an input circuit for inputting musical instrument digital interface (MIDI) commands in real time over a plurality of channels, a computer including a central processing unit (CPU) supplied with the MIDI commands for simultaneously synthesizing one or more voices for each of the channels in response to the MIDI commands, each of the voices being generated by one or more of a plurality of predefined audio synthesis algorithms executed in software, a random access memory (RAM) for storing digital voice data representative of each of the voices generated by the CPU, an output circuit for audibly reproducing the voices from the digital voice data stored in the RAM, and wherein the CPU, in generating the voices selects the one or more audio synthesis algorithms based on one or more of the following criteria: the external processing demands placed upon the CPU by other operations being performed by the personal computer, a best match, according to predetermined criteria, b

Abstract: The present invention is concerned with an audio equipment capable of processing video signals and a method of displaying operation of the acoustic equipment. An input selector switch 1 is supplied with reproduced audio signals from devices connected to a tuner unit 2 and input terminals 3, and supplies a signal to a sound field processing DSP unit 4. A signal processed by the DSP unit 4 is applied through a muting switch 5 to an output terminal 6. An input selector switch 7 is supplied with reproduced video signals from devices connected to input terminals 8, and supplies a signal to a graphic controller (GDC) 9 which generates a video signal to display an image. One of a signal generated by the GDC 9 and an original video signal are selected by a switch 10, and applied to an output terminal 11. The GDC 9 also generates an image simulating a control panel. When a key switch 13 is operated, the display corresponding to the operated key in the simulating image is varied.

Abstract: A reverberation generating system stores parameter data representative of a series of timings and a series of sound intensities for an initial part of reverberation, and calculates a series of parameter data for the later part of reverberation by using a return map of the proportional constant for time intervals of the initial part inversely proportional to the square of lapse of time so that the reverberation generating system stores the parameter data for the initial part only.

Abstract: In order to change an electronic sound between a piano sound generated without manipulation of a damper pedal and a piano sound generated under manipulation of the damper pedal, an electronic keyboard musical instrument assigns two tone generating channels to a depressed key for generating a fundamental tone signal representative of the piano sound generated by striking the set of strings and a resonating tone signal representative of a resonating sound generated through the resonance, and the resonating sound signal is mixed with the fundamental tone signal under manipulation of a pedal corresponding to the damper pedal, thereby making the electronic sound close to the piano sound.

Abstract: An electric guitar amplifier which utilizes a digital signal processor to produce vacuum-tube-like distortion without certain unwanted audio artifacts created by previous digital realizations of nonlinear, high-gain functions. By virtue of a microprocessor-controlled digital signal processor embodiment, the invention gives the user programmable control over parameters normally associated with state of the art guitar amplifiers (e.g. tone controls, reverb controls, tremolo controls, etc.), as well as other musically useful parameters which are not normally included among the controls of a guitar amplifier (e.g. selection of preamp type, autovolume, reverberation type, autowah, etc.).

Abstract: A reverberation network for processing an audio signal including a plurality of wave-ladder filters, each of which simulates a corresponding reflected audio signal from wave propagation in a selected direction, wherein each of the plurality of wave-ladder filters receives as input a signal derived from the audio signal; and an adder which combines the corresponding reflected signals of the plurality of wave-ladder filters to produce a reverberation signal.

Abstract: An object of the present invention is to obtain musical tones being equivalent to those of an acoustic guitar with a hollow body with only an electric guitar with a solid body. For the sake of attaining this object, the effecting system according to the present invention is composed of an absolute value detecting means for detecting absolute values of musical tone signals in response to oscillations of the strings, a delay time setting means for setting a delay time based on the absolute values detected by the above described absolute value detecting means, and a delay means for delaying the above described musical tone signals by the delay time which was set by the above described delay time setting means.

Abstract: In a karaoke apparatus, a pickup device picks up a live singing voice of a music piece to convert the same into a vocal signal. A music source device provides a music signal representative of a karaoke accompaniment of the music piece to accompany the live singing voice, and also provides effect control information which is predetermined in matching with a mood and a tempo of the music piece and which is effective to create and control a vocal effect. A vocal effecter device processes the vocal signal according to the provided effect control information to apply the vocal effect to the live singing voice and to control the applied vocal effect. A mixer device mixes the provided music signal and the processed vocal signal with each other to concurrently reproduce the karaoke accompaniment and the live singing voice which is modified by the applied vocal effect so as to enrich karaoke performance of the music piece.

Abstract: The present invention relates to a pedal resonance effect simulation device for digital pianos consisting of a resonance effect circuit for the simulation of the resonance effects in the strings of a traditional mechanical piano, coupled with a reference model which varies the resonance contributions of the various strings, which are equivalent to those of a mechanical piano, by using delay lines with a method which considers the position of the resonance pedal pressed by the performer upon reproduction.

Abstract: A karaoke apparatus is provided with a pickup device that collects a singing voice sound to convert the same into a corresponding vocal signal. Further, a music generator device generates a music signal representative of a karaoke accompaniment sound which is selected to accompany the singing voice sound. A harmony generator device generates a harmony signal representative of a harmony chorus sound which is made consonant with the singing voice sound. A first processor device processes the vocal signal to impart an effect to the collected singing voice sound. A second processor device processes the harmony signal separately from the vocal signal to impart another effect to the harmony chorus sound independently from the singing voice sound. A sound device amplifies the processed vocal signal, the processed harmony signal and the generated music signal so as to sound the singing voice sound, the harmony chorus sound and the karaoke accompaniment sound concurrently with each other.

Abstract: A sound reproduction control apparatus processes an audio signal in reproduction of a musical sound to control a tone of the musical sound while imparting an effect to the musical sound. A preceding tone controller has a variable frequency response for amplifying the audio signal by the variable frequency response so as to variably control the tone of the musical sound. A signal processor is connected to the preceding tone controller for processing the audio signal to impart a desired effect to the musical sound. A succeeding tone controller is connected to the processor means and has an adjustable frequency response adjusted separately from the variable frequency response of the preceding tone controller for amplifying the audio signal by the adjusted frequency response so as to adjustively control the tone of the musical sound.

Abstract: A multi-channel reverberation system has a substantially constant multi-channel power gain at audio frequencies and comprises multiple signal inputs, one for each input channel, a number of comb filter networks connected to each signal input, each comb filter network including a feed forward stage, a cross-coupling network cross-coupling the comb filters to increase the reverberation echo density, and multiple signal outputs, one for each output channel. Preferably the feed forward stage of each comb filter provides a substantially constant multi-channel power gain at audio frequencies and the cross-coupling matrix is an orthogonal cross-coupling matrix cross-coupling a number of single channel allpass comb filters, positioned immediately before or after the delay lines, to create a multi-channel allpass comb filter with a unitary transfer function matrix at all frequencies.

Abstract: In a tone generating method or device, where an arithmetic unit such as a CPU or DSP is used to execute a specific tone generating processing program to generate tone signals, control is conducted to execute the tone generating processing with its content controlled to be most suitable for the computing capability of the arithmetic unit. Namely, performance or computing capability of the arithmetic unit is detected prior to execution of the tone generating processing, and depending on the detected performance, control is conducted to modify the content of the tone generating processing so as to best fit the performance.

Abstract: There are provided a tone source for generating tone signals in plural channels independently and effect impartment sections, provided in corresponding relations to the channels, for imparting individual effects to the tone signals of the channels generated by the tone source. Each of the effect impartment sections, in accordance with tone control information unique to the tone signal of the corresponding channel, controls a parameter of the effect to be imparted. In a case where any of the effect impartment sections uses a signal delaying memory to impart an effect, when a damping (or truncating process) is performed in any of the channels, the storage area to be used for the channel is switched to another unused storage area so as to prevent any preceding tone's delayed signal from being undesirably mixed into the signal of a new tone.

Abstract: An external memory control circuit for a sound field processing digital signal processor capable of obtaining continuous sound listening by shortening the time that sound is cut off upon a change of the sound field algorithm depending on the listening place (i.e., concert hall, meeting hall, church or the like). The external memory control circuit is constructed to utilize a mute function only for the time required to transfer the algorithm itself where a sound field processing circuit is realized using a DSP, thereby reducing the time that sound is cut off and thereby obtaining continuous sound listening even when the sound field algorithm is changed while listening.

Abstract: Early reflection sounds and subsequent reverberation sounds are processed in different modes by separate processing devices in order to obtain reverberation effects close to those of natural sound as well as to freely produce reverberation effects that do not exist in natural sound. When the early reflection sound overlaps the subsequent reverberation sound in time, the early reflection sound and the subsequent reverberation sound are produced without noise. There is no need to omit either one of them. Besides, complex envelope control is executed for the early reflection sound, and the envelope levels are weighted and fluctuated. Therefore, it is possible to establish a highly sophisticated sound field.

Abstract: This reverberant characteristic signal generation apparatus comprises a measuring room where a speaker and one channel of microphone is provided to supply one channel of a pulse train indicative of the reverberant characteristic of the room, a direction data generation portion for generating direction data indicative of an imaginary incoming direction of a indirectly transmitted impulse sound, an operation portion for operating the time differences due to the inclination of the incoming direction of the indirectly transmitted impulse sound to an imaginary dummy head, having right and left ears having a distance therebetween, at the position of the microphone, and an output portion for outputting an amplitude of each pulse and the delay time of the indirectly transmitted impulse sound to the microphone and the time differences to provide the reverberant characteristic signal which may be recorded by a recorder.

Abstract: A sound field control device includes an acoustic feedback signal path including a chamber and one or more loudspeaker or loudspeakers and one or more microphone or microphones provided in the chamber, the loudspeakers being disposed at a predetermined distance from the microphones, and at least either the loudspeakers or the microphones being provided in the plural, an electrical feedback signal path for feeding back a collected sound signal collected by the microphone or microphones to the loudspeaker or loudspeakers, and a collected sound signal supply control circuit provided in the electrical feedback signal path for changing, with lapse of time, a signal level of each collected sound signal fed back from the microphones to the loudspeakers. The transmission route of the collected sound signal is completely switched or the ratio of distribution of the collected sound signal is changed. Coloration in hearing is thereby reduced and a margin of howling is expanded without causing unnaturalness in hearing.

Abstract: A system for mixing five channel sound which surrounds an audio plane. The position of a sound source is displayed on a VDU (16) relative to the position of a notional listener (31). The sound source is moved within the audio plane by operation of a stylus (23) upon a touch tablet (24). Thus, an operator is only required to specify positions of a sound source over time, whereafter a processing unit (17) calculates actual gain values for the five channels at sample rate. Gain values are calculated for the sound track for each of the loudspeaker channels and for each of these specified points. Gain values are then produced at sample rate by interpolating calculated gain values for each channel at sample rate.

Abstract: A reverberation effect imparting system includes plural comb filters, each of which has a signal delay line and a feedback loop for filtering a delayed output signal from the delay line and feeding the filtered signal back to the input side with a variable loop gain. The device further includes a junction section which receives the respective output signals from the delay lines of the plural comb filters. The junction section controls the received signals with respective variable scattering coefficients and sums up the thus-controlled signals, so as to introduce the summed signal into the feedback loop of each of the comb filters. In this manner, reverberation control according to the waveguide theory is performed by the junction section, while the conventional-type reverberation control is performed by the comb filters. Thus, reverberation can be controlled through a combination of the two controls.

Abstract: A reverberation imparting apparatus comprises at least a data shift register, a coefficient shift register and an arithmetic convolution unit. The data shift register stores a string of sampling data each representative of an instantaneous value of an audio signal at each sampling period, while the coefficient shift register stores a string of coefficients which are created on the basis of waveforms of reflected sounds in response to an impulse sound artificially produced in a certain sound-field space such as a church. Normally, the data shift register successively stores new sampling data which are newly inputted thereto so that the contents of the data shift register is successively renewed. The arithmetic convolution unit performs an arithmetic convolution, using the string of coefficients on the string of sampling data.

Abstract: A signal processor executes a plurality of microprograms to perform delaying processing and various arithmetic computation processings of digital signals input thereto. The signal processor has a storage area divided into a plurality of divided areas corresponding, respectively, to the microprograms. The write and read of the divided areas are controlled. When an instruction for changing at least one of the microprograms is given, the write and read of the digital signals are controlled such that at least one of the digital signals stored in at least one of the divided areas corresponds to the at least one microprogram is cleared without clearing the others of the digital signals stored in the others of the divided areas.

Abstract: A plurality of reverberation signal generating circuits are provided for generating a plurality of reverberation signals from input audio signals of two channels. Each of the reverberation signals has a predetermined function of correlation. A pair of filters are provided for controlling the function of correlation of the reverberation signal and for applying controlled reverberation signals to reverberation speakers, thereby providing a sound field.

Abstract: An audio signal processing system including an input circuit for inputting musical instrument digital interface (MIDI) commands in real time over a plurality of channels, a computer including a central processing unit (CPU) supplied with the MIDI commands for simultaneously synthesizing one or more voices for each of the channels in response to the MIDI commands, each of the voices being generated by one or more of a plurality of predefined audio synthesis algorithms executed in software, a random access memory (RAM) for storing digital voice data representative of each of the voices generated by the CPU, an output circuit for audibly reproducing the voices from the digital voice data stored in the RAM, and wherein the CPU, in generating the voices selects the one or more audio synthesis algorithms based on one or more of the following criteria: the external processing demands placed upon the CPU by other operations being performed by the personal computer, a best match, according to predetermined criteria, b

Abstract: A sound synthesis system, having a pitch adjusting function, comprises a pitch adjusting portion and a loop including a tone-color forming portion and a delay portion. A signal circulating through the loop is delayed using an amount of delay by the delay portion; and then, the tone-color forming portion imparts a tone-color characteristic to the signal delayed so as to produce a sound signal. In the pitch adjusting portion, a pitch of the sound signal is detected; and an initial constant is generated based on a designated pitch and/or a designated tone color. The initial constant represents an initial delay effected by the delay portion. In addition, an amount of correction is created based on a difference between the pitch detected and the designated pitch. Thus, the amount of delay of the delay portion is corrected by using the amount of correction in such a way that the pitch of the sound signal is automatically adjusted to be equal to the designated pitch in real time.

Abstract: According the present invention, a sound field controller for reproducing a sound field with presence comprising; input terminals for inputting an audio signal having a first and a second channel signals, a signal extracting circuit for receiving and processing the audio signal, and producing an extracted signal of the audio signal, an operation circuit for receiving the extracted signal from the signal extracting circuit, performing the convolution on the extracted signal, and generating a convolution sum signal, a delay circuit for delaying the convolution sum signal by a predetermined time, and producing a delayed signal, an adding circuit for receiving the audio signal and the delayed signal, and adding the audio signal and the delayed signal with a predetermined summation ratio to produce a summed signal, and output terminals for reproducing the summed signal to localize a sound image in a desirable direction.

Abstract: In a closed wave guide network having a bidirectional signal transmitting channel section and a signal junction section, signal delay time is variably controlled by a first parameter group so as to control the resonance frequency characteristics of the wave guide network. A signal excitor is connected to the wave guide network so that an excited signal is supplied to the network. The excitation frequency of the excitor is controlled in accordance with a second parameter group. There are also provided a combination determination section which, in correspondence to the pitch of a tone to be generated, determines a combination of the first parameter group to be used in the wave guide network and the second parameter group to be used in the excitor, and a parameter generator which, in accordance with the combination determined by the combination determination section, generates and supplies individual parameters of the first and second parameter groups.

Abstract: An audio signal processing system produces an output 24 having an illusory distance effect for a sound source signal S by feeding it via a direct signal path 25 and an indirect signal path 22, 23 passing through early reflection simulation apparatus 1 which feed an output mixing mechanism 9. A control system adjusts the relative delays 3, 4 and relative gains 5, 6 in the direct 25 and indirect 22, 23 signal paths to modify the illusory distance effect so as to substantially maintain the mathematical relationship between the gains and time delays of simulated reflections relative to first sound arrivals at the output 24 encountered for sounds at that source distance in actual rooms. Signal paths 22, 23, 24, 25 may be stereophonic or multichannel using matrix gain coefficients in the early reflection simulator 1, and may produce different simulated distances for different sound positions. A plurality of sound sources S may have different simulated distances while feeding a common early reflection simulator 1.

Abstract: When a plurality of effects are simultaneously applied to an input audio signal, a plurality of effect-algorithms each for applying solely an effect to the input signal are stored in a memory, and a plurality of combination-algorithms or combination-data which represent combinations of the effects and orders in which effects are applied to the input audio signal are stored in the memory. When one of the combination-algorithms or the combination-data is selected, effect-algorithms included in the selected combination-algorithm or combination-data are selectively read out from the memory. A program for applying effects in a predetermined combination and order is written by CPU based on the read out effect-algorithms and combination-algorithm or combination-data. Receiving the program, DSP successively applies effects to the input audio signal in accordance with the program.

Abstract: A sound effect-creating device frequency-modulates a monaural original sound signal obtained from stereophonic sound signals, delays the resulting modulated signal by delay amounts different from each other, selects and adds up a combination of a plurality of delayed modulated signals to form signals to be added to the stereophonic sound signals, and outputs the resulting stereophonic sound signals.

Abstract: A system for interpreting the requests and performance of an instrumental soloist, stated in the parlance of the musician and within the context of a specific published edition of music the soloist is using, to control the performance of a digitized musical accompaniment. Sound events and their associated attributes are extracted from the soloist performance and are numerically encoded. The pitch, duration and event type of the encoded sound events are then compared to a desired sequence of the performance score to determine if a match exists between the soloist performance and the performance score. If a match exists between the soloist performance and the performance score, the system instructs a music synthesizer module to provide an audible accompaniment for the soloist. The system can provide an accompaniment for a selectable amount of time even if the soloist intentionally or unintentionally departs from the score.

Abstract: An apparatus for synthesizing a musical tone comprises a loop circuit for circulating an input wave form signal through a delay circuit, a wave form memory, and a random modulation circuit for modulating the wave form signal stored in the wave form memory. The random modulation circuit generates a random signal, such as a noise signal. The random signal modulates address data to the wave form memory, so that the read wave form data read out from the wave form memory is delicately varied with time.

Abstract: An electronic musical instrument simulating a wind instrument is fundamentally configured by a loop circuit at least containing a linear portion and a non-linear portion. The linear portion simulates a pipe portion of the wind instrument, while the non-linear portion simulates a breath-blowing portion of the wind instrument so as to produce an excitation signal, which is supplied to the linear portion. By an interaction between the linear portion and the non-linear portion which is carried out while a signal circulates through the loop circuit, a sounding characteristic of the wind instrument is simulated. In order to accurately simulate a jet reed instrument (e.g., organ pipe), there are further provided a delay circuit and an edge tone generator. The delay circuit imparts a delay time to the output signal of the linear portion, and then, a delayed signal is supplied to the non-linear portion.

Abstract: A designating mechanism generates filtering designation data, which designate the contents of filtering, reverberating designation data, which designate the contents of reverberation, and characteristics data, which express the combined contents of filtering and reverberation. A parameters generating mechanism generates filtering parameters expressing filtering characteristics, and generates reverberating parameters expressing reverberation characteristics. A readout device reads out a first operating algorithm, which designates filtering designation data, and a second operating algorithm, which designates reverberating designation data, from a memory mechanism wherein a plurality of operating algorithms are stored.

Abstract: A method and a system for artificial spatialization of audio-digital signals x(k) making it possible to effect on elementary signals xi(k), replicas of the audio-digital signal, different delays creating delayed elementary signals (seri) summed after weighting with the signal x(k) in order to create the spatialized audio-digital signal y(k). A plurality of linear combinations of the signals (seri) as combined delayed elementary signals (serci) is summed with the elementary signals xi(k). So as to simulate a late reverberation, the linear combinations are effected by a unit loopback, and an attenuation hi(.omega.), a decaying monotonic function of the reverberation time Tr(.omega.) to be simulated and proportional to the delay, is effected with each delay. A spectral correction before weighted summation satisfying the relation: ##EQU1## is effected, .tau.i designating the value of each delay, increased by the phase delay due to the attenuation.

Abstract: A moment at which each of a plurality of pieces of sound data is generated in response to one sounding operation is controlled, and the plurality of pieces of sound data are output to different sounding units. Consequently, differences occur among moments at which sounds are radiated by the sounding units, resulting in an occurrence of differences among lengths of time taken by the sounds to reach the ears of a listener, whereby a stereophonic sound can be produced and the listener can localize a sound image.