Abstract: A nonperiodic waveform is forced to a periodic character to facilitate looping of the waveform without introducing audible, and thus objectionable, sound artifacts. Nonperiodic waveforms are typically nonperiodic due to the presence of nonharmonic high frequency spectral components. In time, the high frequency components decay faster than low frequency components and looping of the waveform is facilitated. A loop forcing process and loop forcing filter facilitate looping of a nonperiodic waveform by accelerating the removal of the nonperiodic high frequency components. A loop forcing filter accelerates the removal of nonperiodic high frequency components using a comb filter having a frequency selectivity that varies in time.

Abstract: A system and method for creating and composing musical works by selecting existing musical elements and applying modification modules to them. Existing musical selections reside in a database as metrics, or targets for the composition of new pieces. The present invention composes a new piece of music by, for example, specifying component musical features in the selected targets that should be modified or retained in creating the new work. Such musical features include, for example, rhythmic characteristics, harmonic characteristics, and the like. In this manner, a user who is, musically untutored is able to create satisfying, original works having desired characteristics as specified by the user.

Abstract: There are provided a tone generator system using computer software and a storage medium storing the computer software. A musical tone synthesis module generates a musical tone signal by generating an excitation signal in response to an request for generating the excitation signal and at least delaying and looping the generated excitation signal. A first monitoring module monitors whether or not the request for generating the excitation signal is issued. A second monitoring module monitors whether or not the musical tone signal is generated by the musical tone synthesis module. An interruption module interrupts execution of the musical tone synthesis module when the first monitoring module detects that the request for generating the excitation signal is not issued, and at the same time the second monitoring module detects that the musical tone signal is not generated.

Abstract: A signal processing apparatus is provided with: a music tone judging device for judging a music tone or genre of a music composition composed of an input audio signal: a selecting device for selecting one of a plurality of characteristics prepared in advance for each parameter used for a compression encoding method on the basis of a judgment result of the music tone judging device; and an encoding device for generating a spectrum signal out of the input audio signal and compression-encoding the generated spectrum signal on the basis of the selected characteristic.

Abstract: A sound synthesizer for a plucked string instrument simulates noises and transients produced before and after and during the transitions between notes. A particularly convincing synthesis of a plucked string instrument is produced by combining the transients into the final output. In particular, the transient tap tone and longitudinal vibration of the string resulting from the string pluck, and the damping tap tones resulting from the string damping, are all simulated. In one embodiment, the tap tone created upon attack and plucking of a string, is synthesized, and then used to stimulate a resonant circuit, thereby simulating both the pluck tap tone, and the transverse vibration of the string. In addition, the final damping transients which occur at the end of an articulated note are also simulated, using either an independent transient synthesis section, or by stimulating the resonant circuit with a tap tone, while simultaneously re-tuning that resonant circuit.

Abstract: A musical sound generating system which generates musical sounds having desired tone regardless of the variation of volume setting. In the musical sound generating system, the standard tone control touch curve TCstd, the maximum tone control touch curve Tcmax, and the minimum tone control touch curve TCmin when the amplification factor V is standard (a standard amplification factor Vstd), maximum, and minimum respectively are stored. If the amplification factor is higher than the standard amplification factor Vstd, an interpolation between the standard tone control touch curve TCstd and the maximum tone control touch curve TCmax is performed, and the tone control touch curve TC according to the amplification factor V is obtained. Based on the tone control touch curve, the cut-off frequency F corresponding to the velocity value Tv is determined, and then the tone of the musical sound is controlled by the low pass filter with the cut-off frequency F.

Abstract: An audio apparatus extracts information of a musical performance from an acoustic signal having a frequency and an amplitude, which time-vary during the musical performance. In the audio apparatus, a filtering device has a plurality of filters which are set with different cutoff frequencies so as to pass different frequency ranges of the acoustic signal. A pitch detecting device is connected to the filtering device for processing the acoustic signal to detect therefrom a pitch. A controlling device operates according to the detected pitch fed back from the pitch detecting device for selecting one of the filters set with one of the different cutoff frequencies adapted to the pitch of the acoustic signal so that the pitch detecting device can detect the pitch based on the acoustic signal filtered by the selected filter. Further, the envelope detecting device has a plurality of envelope followers corresponding to the plurality of the filters.

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: 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 method is disclosed for synthesizing acoustic waveforms, especially musical instrument sounds. The acoustic waveforms are characterized by time-varying amplitudes, frequencies and phases of sinusoidal components. These time-varying parameters, at each analysis frame, are obtained in one embodiment by short term Fourier transforms (STFT). The spectrum envelope at each frame is parameterized with an autoregressive moving average model and applied to a waveform consisting of unit amplitude sinusoids via time-domain filtering. The resulting synthetic waveform preserves the time-varying frequency and phase information and has the same relative energy distribution among different sinusoidal components as that of the original signal. Finally, a general waveform shape for the type of acoustic signal being synthesized is applied.

Abstract: This invention provides a circuit for synthesizing musical sounds. In one embodiment, the circuit includes a first order FIR highpass filter that is placed between a modulation phase increment oscillator and a carrier phase increment oscillator. The invention may also include waveshaping circuits, adders, multipliers, time division multiplexing, and other types of filters. Another embodiment of the invention provides a music synthesis method where a modulation phase increment is multiplied by a modulation index to produce a modulation signal. That modulation signal is then filtered and added to a carrier phase increment. Finally, that sum is multiplied by an amplitude envelope to produce a signal representing a musical sound. The method may include using a first order FIR highpass filter, waveshaping of both sinusoidal and non-sinusoidal waveforms, additional multiplying and time division multiplexing. The invention also includes self-modulation and cascading.

Abstract: A tone waveform reproduction apparatus, for an electronic musical instrument employing a plurality of samples in consonance with tone ranges, that can reduce a change in a timbre so that it will not be noticeable, and that can provide a natural transition of musical tones. An address generator generates a read address for a waveform memory. In the waveform memory are stored waveform data that differ for each timbre and each specific pitch range. A digital filter has a low-pass property and is controlled by a cutoff controller to adjust a timber or a tone quality. The cutoff controller generates a cutoff control signal for compensating for a discontinuity, at a point where tone ranges are switched, of reproduced frequencies of tone signals extracted from the waveform memory, and outputs the cutoff control signal fcn.

Abstract: An audio signal analyzer and encoder is based on a model that considers audio signals to be composed of deterministic or sinusoidal components, transient components representing the onset of notes or other events in an audio signal, and stochastic components. Deterministic components are represented as a series of overlapping sinusoidal waveforms. To generate the deterministic components, the input signal is divided into a set of frequency bands by a multi-complementary filter bank. The frequency band signals are oversampled so as to suppress cross-band aliasing energy in each band. Each frequency band is analyzed and encoded as a set of spectral components using a windowing time frame whose length is inversely proportional to the frequency range in that band. Low frequency bands are encoded using longer time frames than higher frequency bands.

Abstract: The signal from a piezo-electric transducer in an electro-acoustic guitar is modified by a variable depth notch filter. The notch filter includes two paths between an input and a summation circuit. The signals in the two paths are 180.degree. out of phase and one of the paths includes a bandpass filter. The notch is located at approximately 5 khz. The depth of the notch depends upon either the broadband amplitude or the narrowband amplitude of the signal from the transducer and the depth of the notch is controlled using either feedback or feedforward control.

Abstract: The PC audio circuit described interfaces with and provides audio enhancement to a host personal computer of the type including a central processor, system memory and a system bus. The PC audio circuit includes a digital signal processor (DSP) for processing wavetable data and generating digital audio signals for a plurality of voices. The wavetable data is stored in the host computer's system memory and transferred in portions, as needed by the DSP, to a smaller, low-cost cache memory included with the PC audio circuit. The DSP processes several frames of data samples for an active voice before processing another voice. Processing in this manner alleviates concerns about the percentage use of system bus bandwidth and the maximum allowable system bus latency. These concerns are further alleviated by deriving frequency compensated wavetable data and storing it in system memory to be retrieved by the DSP for generating digital audio signals having high frequency ratios.

Abstract: An electronic keyboard musical instrument is responsive to a step-on of a soft pedal so as to impart effects similar to those of a soft pedal of an acoustic piano to an electronic sound; a tone generator incorporated in the electronic keyboard musical instrument causes a series of waveform data codes to be read out at a higher speed so as to slightly increase the pitch of an electronic sound, an envelop generator to give a gentle envelop with a small attack and a long sustain to the electronic sound and a filter circuit to emphasize a certain high frequency component, thereby making the electronic sound to be quite close to the piano sound produced under the step-on of the soft pedal.

Abstract: Digital waveform data stored in a waveform memory is read out in response to the ON/OFF operations of key switches such as a keyboard of an electronic musical instrument. The waveform data is passed through a low-pass filter, then subjected to amplitude envelope processing such as attack, decay, release, and the like. The processed waveform data is D/A-converted to output it as a tone signal. A look-up table that stores resonant frequency data and resonance sharpness data (quality factor) of a filter in correspondence with the touch (operation strength or key-ON velocity) of the switch operation at, e.g., the keyboard is used. The resonant frequency and resonance sharpness data are read out from the table in correspondence with the touch data of the keyboard operation to control filter characteristics such as a cutoff frequency, roll-off or slope, and the like, by coefficient data to the filter.

Abstract: A method for resampling includes convolving a given set of samples with the impulse response function of a low-pass filter. In this method, values of the impulse response required for the convolution calculation are computed at the time of resampling from a segmented polynomial approximating the impulse response. In one embodiment, the method is applied to provide musical tones of various pitches from a stored waveform.

Abstract: A musical instrument amplifier system (FIGS. 1 and 2) with a frequency selective crossover circuit (10) using passive band pass filters (36, 44) to separate the amplified signals from the power amplifier (14) of an amplifier system (12) into low frequency signals on an output connector (42) for a specially adapted bass audio speaker (38) while the higher frequency signals are applied to another speaker (46). The crossover circuit (10) is selectively connected with the normal speaker output (24) in lieu of the speaker (38) to add frequency crossover capability to any amplifier system. In another musical instrument amplifying system (88), the crossover circuit (FIG.

Abstract: An efficient digital waveguide synthesizer is disclosed for simulating the tones produced by a non-linearly excited vibrational element coupled to a resonator, such as in a piano. In a preferred embodiment, the synthesizer creates an excitation pulse from a table containing the impulse response of a piano soundboard and enclosure. Alternatively, this excitation pulse can be synthesized by filtering white noise. The excitation pulse is fed into a filter that simulates the collision of the piano hammer and string. Because the hammer-string interaction is nonlinear, the characteristics of this filter vary with the amplitude of the tone produced. The filtered excitation pulse is then fed into a filtered delay line loop which models the vibration of a piano string. Because the excitation pulse already contains the effects of the resonator, the tone produced by the delay line loop does not require additional filtering in order to model the resonator.

Abstract: An electronic keyboard instrument for reproducing reflected sound generated by an acoustic piano and providing the feeling that sound is reflected and shifted. Left and right system sound signals are generated corresponding to the position of a depressed key on a keyboard. The sound signals are processed through a digital signal processor, a digital-to-analog converter and amplifiers. The processed sound signals are transmitted to left and right loudspeakers. The digital sound processor is composed of filters for extracting predetermined frequency components from each of the sound signals, delay elements for transmitting outputs from the filters with a delay of predetermined time, and adders for adding outputs from the delay elements to the original left and right system sound signals.

Abstract: A parametric signal modeling musical tone synthesizer utilizes a multidimensional filter coefficient space consisting of many sets of filter coefficients to model an instrument. These sets are smoothly interpolated over pitch, intensity, and time. The filter excitation for a particular note is derived from a collection of single period excitations, which form a multidimensional excitation space, which is also smoothly interpolated over pitch, intensity and time. The synthesizer includes effective modeling of attacks of tones, and the noise component of a tone is modelled separately from the pitched component. The input control signals may include initial pitch and intensity, or the intensity may be time-varying. A variety of instruments may be specified.

Abstract: A music composition system, comprising receiving a first harmony including a first melody, analyzing the first harmony to derive in real-time a rule relating the first melody to the first harmony, receiving a second melody, and applying the rule in real-time to the second melody to produce a second harmony relating to the second melody.

Abstract: The device is designed to receive at its input the analog AC output signal of an electric guitar either directly from the guitar, its amplifier or a speaker. The incoming signal is fed to three series connected circuits which successively boost the bass and treble frequencies, and then reduce the signal level between the bass and treble peaks before applying the processed signal to an output circuit. A high frequency adder circuit connected in parallel with the three series connected circuits permits frequencies above a certain level (10 KHz) to be added back to the processed signal at the output circuit. One switch is operable in the bass boost circuit to introduce different resistance selectively to raise or lower the frequency center and gain of the circuit, and another switch at the input of the device is operable to add resistance when the input signal is from a speaker.

Abstract: A musical tone signal forming apparatus includes a pickup which detects vibrations of a string of an electric guitar and produces a string vibration signal. The string vibration signal and the output of a first delay imparting a first delay time are combined in a first combiner. Second and third delays impart second and third delay times to the string vibration signal, respectively. Delayed signals from the second and third delays are combined in a second combiner. Outputs of the first and second combiners are combined in a third combiner. The first delay time is determined by the distance between a bridge and a first imaginary pickup supposed to detect vibrations of the string. The second delay time is determined by the distance between the first imaginary pickup and a second imaginary pickup supposed to be disposed nearer to the bridge. The third delay time is determined by the distance between the bridge and the mid-point of the first and second imaginary pickups.

Abstract: A musical tone signal producing apparatus having a loop circuit portion including a filter circuit and a delay circuit for circulating therethrough a vibratory waveform signal to be produced as a musical tone signal and a control signal generator for generating a first control signal for effecting excitation of the waveform signal in the loop circuit and applying the first control signal to the filter circuit and for generating a second control signal for applying time variation to the waveform signal. The musical tone signal producing apparatus further includes a filter control circuit for modifying the second control signal based on the vibratory waveform signal and for applying a control parameter defined by the modified second control signal to the filter circuit for control of a signal transmission characteristic of the filter circuit.

Abstract: A music synthesizer includes main resonator, such as a digital waveguide network, that is coupled to a digital passive nonlinear filter. The passive nonlinear filter receives traveling wave signals from the resonator and generates modified traveling wave signals having a different frequency spectrum than the received traveling wave signals without changing the received traveling wave signals' energy content. The passive nonlinear filter then transmits the modified traveling wave signals back into the resonator. The passive nonlinear filter includes a first memory element for retaining an internal energy state and a dual-mode signal generator that generates the modified traveling wave signal from the received signals and the internal energy state using a first signal processing method when the internal energy state has a negative value and using a second distinct signal processing method when the internal energy state has a positive value.

Abstract: An electronic music system which imitates acoustic instruments addresses the problem wherein the audio spectrum of a a recorded note is entirely shifted in pitch by transposition. The consequence of this is that unnatural formant shifts occur, resulting in the phenomenon known in the industry as "munchkinization." The present invention eliminates munchkinization, thus allowing a substantially wider transposition range for a single recording. Also, the present invention allows even shorter recordings to be used for still further memory improvements. An analysis stage separates and stores the formant and excitation components of sounds from an instrument. On playback, either the formant component or the excitation component may be manipulated.

Abstract: According to a first invention, provided is an electronic musical instrument, which decodes and reads waveforms that are compressed by the DPCM method or the ADPCM method, that stores a prediction filter coefficient that is consonant with each waveform and reproduces musical tones by using the prediction filter coefficient. In the first invention, a waveform that is stored in the electronic musical instrument is stored together with a prediction filter coefficient that is used when the waveform was prepared, and the optimal prediction filter coefficient is employed for each waveform to reproduce a waveform.According to a second invention, provided is an electronic musical instrument, which decodes waveforms that are compressed by the DPCM method or the ADPCM method and repeatedly reads the decoded data, that can repetitiously read waveform data at the loop top without requiring a device for setting a decoding device.

Abstract: A digital signal processor (i.e., DSP) is employed by an electronic musical instrument. The contents of operational processes to be executed by the DSP can be arbitrarily set in accordance with filtering operations of the filters to be realized. A processing device to be embodied by the DSP at least contains a microprogram memory, a coefficient register, a delay memory and a processing portion. A desired set of coefficients are read from the coefficient register in accordance with a first address signal, while a desired program is read from the microprogram memory in accordance with a second address signal. Each program contains at least one operational command or one instruction. The value of each of the first and second address signals is periodically changed in each sampling period.

Abstract: An electronic musical instrument having a pitch designating unit capable of designating pitches of a plurality of tone signals to be generated, a plurality of tone signal generating channels for selectively generating and outputting a plurality of tone signals having a different pitch designated by the pitch designating unit, a mixing unit for mixing tone signals generated by the plurality of tone signal generating units, a plurality of resonance signal generating units each having a different resonance frequency characteristic, for receiving a mixed tone signal from the mixing unit and generating and outputting a plurality of different resonance signals, and an input level controller provided on the input side of the plurality of resonance signal generating units, for controlling a level of the mixed tone signal from the mixing unit for each resonance signal generating unit, in accordance with a pitch of each of a plurality of tone signals to be generated, and supplying the level controlled mixed tone signal

Abstract: Ratios of frequencies of frequency components of formant waveform signals are changed, and consonance of frequency components of a musical tone is controlled. Moreover, amplitudes, frequencies or numbers of formants are controlled depending upon the musical factors, making rich the contents for changing the musical tones.

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.

Abstract: In the waveform processing apparatus of the present invention, an original sound signal is divided into a plurality of bands, parameters such as amplitude and frequency are extracted for each band, and correction is made so that the amount of deviation of the parameters from the mean value thereof is gradually decreased to zero from a certain time to the time when the repetition period begins, whereby a musical tone signal is re-synthesized based on the corrected parameters. The electronic musical instrument which stores the waveforms processed by the apparatus of the present invention generates musical tones having more natural tone color change.

Abstract: There are provided an analyzer which processes a supplied original sound waveform signal so as to generate one or more analyzed waveform signal components, a synthesizer which receives and processes at least one of the analyzed waveform signal components from the analyzer so as to synthesize a waveform signal on the basis of the processing, and a control section which controls the respective parameters used in the analyzer and synthesizer independently of each other. At least output signal of the synthesizer is taken out and output as a tone signal. In order to achieve diversified control of the tone color or characteristic of a tone to be generated, a tone signal may be obtained by taking out the respective output signals of the analyzer and synthesizer for synthesis.

Abstract: A tone generator for an electronic musical instrument analyzes an original waveform signal to store results of analysis thereof into a memory. The results of the analysis read out from the memory is interpolated to generate results of the interpolation in synchronism with each sampling clock. The results of the analysis are corrected to cancel a frequency characteristic to be imparted to the results of the analysis when they are interpolated. The results of the analysis are stored into the memory after having been converted. In another form, a reading numerical value formed of an integer part and a decimal part is generated, in synchronism with a predetermined clock timing. The numerical value is variable at a rate proportional to the pitch of a musical tone to be produced. Residual waveform samples read from the memory based on the integer part are interpolated by the use of the decimal part in synchronism with a predetermined clock timing.

Abstract: A musical tone synthesizing apparatus provides a closed loop containing an adder, a delay circuit, a filter and a gain control portion. The delay circuit has a delay time which is set responsive to a tone pitch of a musical tone to be produced. An excitation signal is produced in response to the musical tone to be produced and is introduced into the closed loop through the adder, so that the excitation signal circulates through the closed loop. The gain control portion is provided to control a loop gain of the closed loop on the basis of a preset parameter and another parameter which is controlled responsive to a tone color of the musical tone to be produced. The signal circulating through the closed loop is picked up as a musical tone signal representing the musical tone to be produced. By controlling the loop gain of the closed loop, an attenuation characteristic of the musical tone signal is controlled. Hence, it is possible to freely control a sounding time of a decay sound such as a guitar sound.

Abstract: The present invention pertains to audio equipment simulative of musical instruments. The invention allows a user to simulate the playing of music reproduced from an external audio source. The present invention allows a user to quickly modify the amplification of a given audio band pass frequency by selecting a first trigger associated with that band pass frequency, and subsequently activating a second trigger to modify the selected band pass frequency. The present invention has an decorative and functional appearance which may simulate any musical instrument, such as a guitar.

Abstract: A method and apparatus for analyzing and reducing or increasing the dissonance of an electronic audio input signal are realized by identifying the partials of the audio input signal by frequency and amplitude. The dissonance of the input partials is calculated with respect to a set of reference partials according to a procedure disclosed herein. One or more of the input partials is then shifted, and the dissonance re-calculated. If the dissonance changes in the desired manner, the shifted partial may replace the input partial from which it was derived. An output signal is produced comprising the shifted input partials, so that the output signal is more or less dissonant that the input signal, as desired. The method may be used with computerized sound processing equipment, e.g., MIDI-based equipment. The input signal and reference partials may come from different sources, e.g.

Abstract: A filter device has a variable frequency cut-off characteristic as well as a variable gain characteristic. To obtain these characteristics, a filtering operation is performed on data input thereto on the basis of a transmission function, which is defined by gain G for predetermined frequencies and a frequency fo corresponding to a phase at an intersection of a unit circle and a straight line intersecting at right angles a real axis passing a midpoint between a polar and a zero point on the real axis on Z-plane. An electronic musical instrument employing the filter device can change amplitudes of frequency components involved in musical tones on the basis of performance data, generating musical tones involving complex harmonics and having various tone colors, as a conventional acoustic instrument does.

Abstract: A tone generator for an electronic musical instrument is fundamentally configured by a drive-waveform creating portion and a closed loop. The drive-waveform creating portion creates a drive-waveform signal by mixing an excitation waveform and a noise waveform together. The drive-waveform signal is applied to the closed loop through an adder. The closed loop contains a plurality of feedback paths, each of which at least contains a delay circuit and an all-pass filter. The adder adds all of output signals of the feedback paths together with the drive-waveform signal so as to produce a musical tone signal. The number of delay stages, representing an amount of delay to be used in each delay circuit provided in each feedback path, is designated in response to a delay ratio which is arbitrarily set.

Abstract: An electronic musical instrument reads data of a musical tone from a waveform memory thereof according to pitch information input thereto, to form a tone signal containing a tone amplitude signal indicative of amplitude of the musical tone. The tone signal is passed into a plurality of signal paths to be formed into a plurality of analog tone signals. The plurality of signals paths each amplify the tone amplitude signal according to a note on/off velocity, and attenuate a plurality of modifications of the tone signal, depending on or not depending on the note on/off velocity and the pitch information on the musical tone, to generate a musical tone suitably reflecting the note on/off velocity and the pitch.

Abstract: A filter device has a variable frequency cut-off characteristic as well as a variable gain characteristic. To obtain these characteristics, a filtering operation is performed on data input thereto on the basis of a transmission function, which is defined by gain G for predetermined frequencies and a frequency fo corresponding to a phase at an intersection of a unit circle and a straight line intersecting at right angles a real axis passing a midpoint between a polar and a zero point on the real axis on Z-plane. An electronic musical instrument employing the filter device can change amplitudes of frequency components involved in musical tones on the basis of performance data, generating musical tones involving complex harmonics and having various tone colors, as a conventional acoustic instrument does.

Abstract: A tone generation system includes one or more digital waveguide networks coupled to one or more junctions, one of which receives a control signal for controlling tone generation. The control signal initiates and interacts with a wave signal propagating through the waveguide networks to form a tone signal. A non-linear junction may be employed which receives a signal from a waveguide, converts it in accordance with a non-linear function based upon the value of the control signal and provides it back to the waveguide. A tone signal whose pitch is determined by the wave transmission characteristics of the waveguide network is thereby produced.

Abstract: The musical tone synthesizing apparatus utilizes a loop circuit receptive of an input signal for looping the same with a given delay time to produce an output signal representative of a musical tone. The loop circuit includes a delay element for delaying the input signal and a feedback path for feeding back the delayed input signal to the delay element. A control unit is provided for controlling the delay element according to a designated pitch of a musical tone so as to set the delay time of the loop circuit to thereby determine the pitch of the musical tone. A generator generates the input signal containing various frequency components. A high-pass filter is connected precedingly to the loop circuit for selectively attenuating relatively lower frequency components of the input signal, which have periods longer than the set delay time of the loop circuit.

Abstract: The musical tone synthesizing apparatus is mainly configured by a closed loop which at least provides a delay circuit and an all-pass filter. The delay circuit delays an input signal (e.g., excitation wave signal) by a first delay time corresponding to a certain integral number of sampling periods. The all-pass filter functions to at least delay an output of the delay circuit by a second delay time corresponding to a decimal fraction of the sampling period, so that an output of the all-pass filter is fed back to the delay circuit. The whole delay time of the closed loop consists of the first and second delay times which can be respectively controlled. Thus, a musical tone signal representing a synthesized musical tone (e.g., an attenuating sound which is produced from an percussion instrument) is picked up from the closed loop. Incidentally, the whole configuration of the closed loop can be embodied by a digital signal processor (DSP).

Abstract: Reading of a waveform memory is controlled and one of different processing is performed depending upon which of a first mode and a second mode has been designated. When the first mode has been designated, an interpolation operation is performed on the basis of plural sample value data read from the waveform memory by using a predetermined number of processing time slots and a resulting one waveform sample value data is produced. When the second mode, has been designated, a processing for producing one waveform sample value data is performed on the basis of one sample value data read from the waveform memory by using one processing time slot. By using a predetermined plural number of processing time slots, one waveform sample value data produced with a high accuracy in the first mode whereas plural waveform sample value data are produced in the second mode whereby generation of a waveform can be performed efficiently by properly utilizing the mode.

Abstract: A method for processing a digital signal produced by digitizing an analog signal such as a musical instrument sound signal, and an apparatus for producing sound source data. When the input signal contains a periodically repetitive waveform portion, the fundamental frequency and its high harmonic components of the input signal is extracted by a comb filter prior to signal processing which takes advantage of the periodicity of the input signal. The fundamental frequency or pitch is detected by performing Fourier transform to produce frequency components, phase matching these frequency components and performing inverse Fourier transform. When extracting a repetitive waveform portion or so-called looping domain, such looping domain having the highest similarity in waveform in the vicinity of both ends of the domain is selected.

Abstract: A tone signal generating apparatus, including a memory for storing wave data containing frequency components within a full band; a tone generator for reading the wave data from the memory in accordance with key ON/OFF data to generate a tone signal; a velocity generator for generating a velocity value based on key ON/OFF data; a first coefficient processor for producing a first coefficient to control the amplitude in accordance with the velocity value generated by the velocity generator; a first operator for performing an operation on the first coefficient produced by the first coefficient processor and the tone signal generated by the tone generator, a filter for extracting frequency components within a specific band from the tone signal output from the first operator to produce a first tone signal and outputting the first tone signal; a second coefficient processor for producing a second coefficient to control the amplitude in accordance with the velocity value generated by the velocity generator; and a sec

Abstract: A circuit for adding a resonance tone to a tone signal to be generated is provided. When a damper operator is not operated, the resonance tone is not added but an ordinary tone signal is generated. When the damper operator has been operated, the resonance tone is added so that a tone signal including the resonance tone is generated. An effect of a damper operator, i.e., loud pedal, in a piano, a natural musical instrument, is thereby simulated with high fidelity. The resonance tone may be produced by passing an ordinary tone signal through a filter. Alternatively, data obtained by sampling an actually produced tone of a piano, a natural musical instrument, when a damper operator, i.e., loud pedal, is ON may be stored in a memory and a resonance tone may be generated by reading out the stored data from the memory. The signal of the generated resonance tone may be sounded by itself or after mixing with an ordinary tone signal at a suitable mixing ratio.