Abstract: An audio signal encoding method is provided that comprises collecting audio signal samples, determining sinusoidal components in subsequent frames, estimating amplitudes and frequencies of the components for each frame, merging the obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by way of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, outputting the quantized coefficients as output data, wherein segments of different trajectories starting within a particular time are grouped into Groups of Segments, and the partitioning of trajectories into segments is synchronized with the endpoints of a Group of Segments.

Abstract: An audio signal encoding method is provided that comprises collecting audio signal samples, determining sinusoidal components in subsequent frames, estimating amplitudes and frequencies of the components for each frame, merging the obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by way of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, outputting the quantized coefficients as output data, wherein segments of different trajectories starting within a particular time are grouped into Groups of Segments, and the partitioning of trajectories into segments is synchronized with the endpoints of a Group of Segments.

Abstract: An audio signal encoding method is provided. The method comprises: collecting audio signal samples, determining sinusoidal components in subsequent frames, estimation of amplitudes and frequencies of the components for each frame, merging thus obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by means of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, outputting the quantized coefficients as output data, wherein segments of different trajectories starting within a particular time are grouped into Groups of Segments (GOS), and the partitioning of trajectories into segments is synchronized with the endpoints of a Group of Segments).

Abstract: A method is described that includes performing the following with an instruction execution pipeline of a semiconductor chip. Multiplying two vectors by: receiving a vector element multiplicand and vector element multiplier expressed in a first base system; converting the vector element multiplicand and vector element multiplier into a second lower base system to form a converted vector element multiplicand and a converted vector element multiplier; multiplying with a first execution unit of the pipeline the converted vector element multiplicand and the converted vector element multiplier to form a multiplication result; accumulating in a register a portion of the multiplication result with a portion of a result of a prior multiplication of operands expressed in the second lower base system; and, converting contents of the register into the first base system.

Abstract: Content identification methods for consumer devices determine robust audio fingerprints that are resilient to audio distortions. One method generates signatures representing audio content based on a constant Q-factor transform (CQT). A 2D spectral representation of a 1D audio signal facilitates generation of region based signatures within frequency octaves and across the entire 2D signal representation. Also, points of interest are detected within the 2D audio signal representation and interest regions are determined around selected points of interest. Another method generates audio descriptors using an accumulating filter function on bands of the audio spectrum and generates audio transform coefficients. A response of each spectral band is computed and transform coefficients are determined by filtering, by accumulating derivatives with different lags, and computing second order derivatives.

Abstract: A system and method is disclosed that facilitates the processing of a sound signal. In embodiments, an input sound signal can be processed according to a computational model using predetermined parameters. A sound signal originating from a musical instrument can be processed according to coefficients that are generated using a learning model.

Abstract: In some embodiments, an electric guitar interface system, includes a first touchpad of an electric guitar configured to detect a user input, and a control unit coupled to the first touchpad. The control unit may be configured to set a first parameter of the electric guitar's output as a function of a position of the user input along the first axis. The first parameter includes a first pickup gain. The first pickup gain may be for at least one of a bridge pickup, a middle pickup, and a neck pickup. The first parameter includes a second pickup gain.

Abstract: An acoustic effect impartment apparatus detects striking of any one of strings by a corresponding hammer in an acoustic piano like a grand piano, and vibrates a vibration section with a driving waveform signal obtained by synthesizing sine wave signals of the fundamental frequency and harmonic frequency of the hammer-struck string. Such vibration of the vibration section is transmitted to the keys via a soundboard and bridge of the piano. Thus, vibration is excited in the hammer-struck string by the striking with the hammer but also by the driving waveform signal, so that an acoustic effect corresponding to the driving waveform signal is imparted. Because the driving waveform signal is a simple signal using the sine wave signals corresponding to the fundamental frequency of the string, a natural feeling of the acoustic piano will not be lost even when the acoustic effect is imparted.

Abstract: In a coefficient measurement apparatus, a line input terminal receives a pickup signal that is generated based on a string vibration of a musical instrument. A microphone input terminal receives a microphone signal acquired by a microphone that collects sounds of the musical instrument. An adaptive filter estimates a transfer function associated to resonance of the musical instrument and a transfer function of an acoustic space formed from the musical instrument to the microphone, generates an output signal by processing the pickup signal using the estimated transfer function, and updates the transfer function using a difference between the output signal and the microphone signal as a reference signal.

Abstract: A signal processing device is composed of a signal acquisition unit and a signal processing unit. The signal acquisition unit acquires a signal corresponding to a vibration propagated from a string attached to a stringed instrument from a pickup element that picks up the signal corresponding to the vibration. The signal processing unit includes a filter that performs convolution operation using a filter coefficient set in the filter, the signal processing unit applying the convolution operation to the acquired signal through the filter and outputting a processed signal.

Abstract: A system for timbral change, capable of changing timbres included in an existing music audio signal to arbitrary timbres. Replaced harmonic peak parameters are created by replacing a plurality of harmonic peaks included in harmonic peak parameters, which are stored in a separated audio signal analyzing and storing section 3 and indicate relative amplitudes of n-th order harmonic components of each tone generated by a musical instrument of a first kind, with harmonic peaks included in harmonic peak parameters, which are stored in a replacement parameter storing section 6 and indicate relative amplitudes of n-th order harmonic components of each tone generated by a musical instrument of a second kind and corresponding to each tone generated by the musical instrument of the first kind. A synthesized separated audio signal generating section 7 generates a synthesized separated audio signal for each tone using parameters other than the harmonic peak parameters and the replaced harmonic peak parameters.

Abstract: A method for calculating measures of similarity between time signals, which includes: acquiring and comparing data (xi, yj) of time signals (X, Y); assigning a one or a zero to every two compared data (xi, yj), depending on the result of said comparison, creating a data set; determining time sequences with said ones and zeros of the data set, each one being formed by consecutive sub-sequences of ones, separated by discontinuities of zeros; selecting the highest result of accumulated results obtained for each sub-sequence, adding for each determined point i, j of value one said one to the accumulated result of maximum value, from among the accumulated results at a point i?1, j?1 of said sub-sequence, a point i?2, j?1 of a sub-sequence of a second time sequence, and a point i?1, j?2 of a sub-sequence of a third sequence.

Abstract: A method and user interface for data sonification for representing multidimensional numerical information with a plurality of variable-timbre channels are described. The method includes providing a user interface that includes a plurality of metaphors and spatial sound rendering operations resulting in stereo audio output. In one implementation, a metaphor is used in making parameter assignments according to which audio-frequency waveforms having adjustable timbre attributes are generated. In another implementation, timbre attributes are adjustable over a range of timbre variation and are further associated with corresponding metaphors.

Abstract: A musical tone signal is synthesized based on performance information to simulate a sound generated from a musical instrument having a string and a body that supports the string by a support. There is provided a closed loop circuit having a delay element that simulates delay characteristic of vibration propagated through the string and a characteristic control element that simulates a variation in amplitude or frequency. A string model calculation circuit inputs an excitation signal based on the performance information to the closed loop circuit, and calculates first information representing a force of the string acting on the support based on a cyclic signal generated in the closed loop and representing the vibration of the string circuit. A body model calculation circuit calculates second information representing a displacement of the body or a derivative of the displacement. A musical tone signal calculation circuit calculates the musical tone signal.

Abstract: Interactive data sonification for representing multidimensional numerical information with a plurality of variable-timbre channels for use complementing data visualization is described. The method includes generation of a plurality of variable-timbre audio waveforms, each having an audio frequency parameter and at least one timbre modulation parameter having an adjustable value that affects the timbre of the audio waveform. The method includes associating aspects of multidimensional numerical data with the timbre modulation parameter of each audio frequency waveform using a mapping element. The mapping element varies values of timbre modulation parameters responsive to selected values from the multidimensional numerical data. Each audio frequency waveform can be positioned within a sonically-rendered sound field, associating information with positions within the sound field. Mapping elements can vary the positions responsive to selected values from the multidimensional numerical data.

Abstract: The Source-Dependent Instrument is a signal processing and signal generation system that uses one or more signal event generators that can be functionally activated and controlled by the analysis of an external input signal. These output generators and signal processors can be set to re-synthesize aspects of the input or synthesize a more complex or perceptually shifted output based on the input.

Abstract: In the present invention, a click sound corresponding to key depression speed is generated, and the production timings of fundamental and harmonic components respectively corresponding to each footage are changed to vary from one another in accordance with a wait time, whereby the fundamental and the harmonic components to be synthesized by additive synthesis are changed to differ from one another. Next, a click sound corresponding to key release speed is generated, and the stop timings of the fundamental and harmonic components are changed to vary from one another in accordance with a wait time, whereby the fundamental and the harmonic components to be muted are changed to differ from one another. Accordingly, by the click sound being mixed with the drawbar sound having these slight tone changes, a unique drawbar sound such as that generated by the sound producing mechanism of an actual drawbar organ is generated.

Abstract: A data sonification method for representing multidimensional numerical information with a plurality of variable-timbre channels rendered in a sound field is described. The method includes generation of a plurality of variable-timbre audio waveforms, each having an audio frequency parameter and at least one timbre modulation parameter having an adjustable value that affects the timbre of the audio waveform. The method includes associating aspects of multidimensional numerical data with the timbre modulation parameter of each audio frequency waveform using a mapping element. The mapping element varies values of timbre modulation parameters responsive to selected values from the multidimensional numerical data. The method also positions each audio frequency waveform within a sonically-rendered sound field, associating aspects of multidimensional numerical data with the sonically-rendered position within the sound field. The sound field can be stereo, two-dimensional, or three-dimensional.

Abstract: A data sonification system for representing a plurality of channels of numerical information is described. The data sonification system includes a plurality of audio waveform generator elements. Each of the audio waveform generator elements generates an associated audio frequency waveform. Each audio frequency waveform has an audio frequency parameter and at least one timbre modulation parameter having a settable value. The timbre modulation parameter affects the timbre of the audio waveform. The data sonification system includes a mapping element for associating aspects of multidimensional numerical data with the timbre modulation parameter of each audio frequency waveform. The mapping element sets the value of the timbre modulation parameter in response to multidimensional numerical data.

Abstract: An online real-time session is conducted between at least two electronic music devices each equipped with an interface connectible to a communication network and a display with a touch sensing ability. An electronic music device communicates with its counterpart device to count a time Ta1 of making an inquiry about a present time Tb of the counterpart device and a time Ta2 of receiving a response from the counterpart device while setting a time Ta3 which progresses from the time Ta2 and a time interval Td which is counted from the time Ta3. Thus, the electronic music device determines tone-generation timing at which the electronic music device is synchronized with its counterpart device in conducting an online real-time session by way of a calculation of Td+Tb+(Ta3?Ta1)?(Ta2?Ta1)/2, or Td+Tb+(Ta3?Ta2)+(Ta2?Ta1)/2.

Abstract: A method of sound-object oriented analysis and of note-object oriented processing a polyphonic digitized sound recording present in the form of a time signal F(A, t) includes the following analytical and processing steps: portion-wise readouts of the time signal F(A, t) using a window function and overlapping windows; Fourier-transforming the readout signal into frequency space, in particular by applying a discrete Fourier transform; calculating an energy value E at each bin from the frequency amplitude resulting from the Fourier transformation, in particular by squaring the real and imaginary parts or forming energy values derived from them; generating a function F(t, f, E); identifying event objects; identifying event objects; identifying note objects; comparing the temporal occurrence of event objects and note objects and associating event objects to note objects in the case of plausible time occurrences; calculating spectral proportion factors for each note object.

Abstract: The apparatus for converting an information signal from a time to a variable spectral representation includes a means for windowing the information signal, a means for converting the windowed information signal to a spectral representation, and a means for weighting a set of information signal spectral coefficients with several sets of complex base function coefficients provided from a means for providing the sets of base function coefficients. The sets of base function coefficients are derived from base functions of various frequencies by windowing and transform, wherein several sets of base function coefficients are provided for one and the same base function for base functions of higher frequencies, wherein the windows for providing these sets are related to various time portions of the base function. The variable spectral representation exhibits variable bandwidth of the variable spectral coefficients, which are efficient and accurate to calculate and especially suited for music analysis purposes.

Abstract: Methods, system and/or computer program products for detection of a note include receiving an audio signal and generating a plurality of frequency domain representations of the audio signal over time. A time domain representation is generated from the plurality of frequency domain representations. A plurality of edges are detected in the time domain representation and the note is detected by selecting one of the plurality of edges as corresponding to the note based on characteristics of the time domain representation.

Abstract: Digital complex tone generators include a first tone generator configured to generate a first digital tone with selectable first characteristics including a first frequency, a first phase, and a first amplitude; a second tone generator configured to generate a second digital tone with selectable second characteristics including a second frequency, a second phase, and a second amplitude; and a generator adder configured for combining the first tone and the second tone to provide a digital complex tone with programmable characteristics. Corresponding methods include initializing a first and second tone generators based on, respective, selected frequencies, phases, and amplitudes; iteratively generating a first digital tone and a second digital tone; and combining these two tones to provide the digital complex tone.

Abstract: Audio item(s) that may be of interest to a user can be selected from a larger collection of audio items. The audio items of interest may be identified by concurrently generating audio from each item in the collection. The audio generated from individual items in the collection may be generated such that the audio is audibly and selectably differentiable from the audio generated from other items in the collection. A user-input may be detected that corresponds to a selection of a subset of the audio items. A user-input may be detected that modifies characteristics of the audio presentation in space and/or volume. A correlation between the input and the selected audio may be made through characteristics that are incorporated into the selected audio when that audio is made distinguishable.

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: Audio samples corresponding to audio extracts or whole audio titles are automatically mapped to triggers 12 in a playable sound-producing device 1, the mapping being dependent on the meta-data associated with the audio samples. Thus, a user can play the sound-producing device and generate sounds derived from his favorite audio titles. It is possible to define different mappings between the audio samples and the playable domain of the sound-producing device: an audio sample selector 50 can select between different possible samples for playback by comparing the audio properties of the samples and the play-mode and/or characteristics of the user's performance. An audio sampler/sample-processor 70 can automatically extract segments of an audio source file and map them to triggers in the sound-producing device 1.

Abstract: Based on the understanding that time-varying characteristics of a tone element, such as an amplitude and pitch, in waveform data acquired through a live performance of a musical instrument include a variation component intended or controllable by a human player and a variation component not intended or non-controllable by the human player, the present invention allows the two components to be adjusted/controlled separately and independently of each other, so as to achieve effective and high-quality control. Discrete variation value train is acquired for at least one particular tone element in original waveform data, and the acquired variation value train is separated, in accordance with a time constant factor, into a “swell” value train of a relatively great time constant and a “fluctuation” value train of a relatively small time constant. The “swell” value train and “fluctuation” value train are variably controlled independently of each other.

Abstract: Methods, system and/or computer program products for detection of a note include receiving an audio signal and generating a plurality of frequency domain representations of the audio signal over time. A time domain representation is generated from the plurality of frequency domain representations. A plurality of edges are detected in the time domain representation and the note is detected by selecting one of the plurality of edges as corresponding to the note based on characteristics of the time domain representation.

Abstract: The apparatus for converting an information signal from a time to a variable spectral representation includes a means for windowing the information signal, a means for converting the windowed information signal to a spectral representation, and a means for weighting a set of information signal spectral coefficients with several sets of complex base function coefficients provided from a means for providing the sets of base function coefficients. The sets of base function coefficients are derived from base functions of various frequencies by windowing and transform, wherein several sets of base function coefficients are provided for one and the same base function for base functions of higher frequencies, wherein the windows for providing these sets are related to various time portions of the base function. The variable spectral representation exhibits variable bandwidth of the variable spectral coefficients, which are efficient and accurate to calculate and especially suited for music analysis purposes.

Abstract: A device (1) is arranged for synthesizing sound represented by sets of parameters, each set comprising noise parameters (NP) representing noise components of the sound and optionally also other parameters representing other components, such as transients and sinusoids. Each set of parameters may correspond with a sound channel, such as a MIDI voice. In order to reduce the computational load, the device comprises a selection unit (2) for selecting a limited number of sets from the total number of sets on the basis of a perceptual relevance value, such as the amplitude or energy. The device further comprises a synthesizing unit (3) for synthesizing the noise components using the noise parameters of the selected sets only.

Abstract: The present invention implements a method and an apparatus for retrieving a sound data desired by the user on the basis of its subjective impression over the sound data. The subjective impression on the desired sound data is entered by the user and converted to a numerical value. A target sound impression value which is a numerical form of the impression on the sound data is calculated from the numerical value. The target sound impression value is then used as a retrieving key for accessing a sound database where the audio signal and the sound features of a plurality of the sound data are stored. This allows the desired sound data to be retrieved on the basis of the subjective impression of the user on the sound data.

Abstract: Systems (100 or 300) and methods (400 or 500) are provided for selecting a post-compression waveform from a post-compression waveform table (106) and supplying it to a synthesis engine (108). The post-compression waveform is based upon a set of post-compression coefficients determined by generating a frequency-domain representation of a periodic signal, the representation including at least one pre-compression frequency-domain sample (204), and performing a threshold-based compression of the pre-compression frequency-domain samples. Systems and methods also include indexing and storing (502) post-compression coefficients in a post-compression coefficient table (102), generating (506) a post-compression waveform based upon the set of post-compression coefficients, and placing (508) the post-compression waveform in the table prior to the selecting (510).

Abstract: There are provided a singing voice-synthesizing method and apparatus capable of performing synthesis of natural singing voices close to human singing voices based on performance data being input in real time. Performance data is inputted for each phonetic unit constituting a lyric, to supply phonetic unit information, singing-starting time point information, singing length information, etc. Each performance data is inputted in timing earlier than the actual singing-starting time point, and a phonetic unit transition time length is generated. By using the phonetic unit transition time, the singing-starting time point information, and the singing length information, the singing-starting time points and singing duration times of the first and second phonemes are determined. In the singing voice synthesis, for each phoneme, a singing voice is generated at the determined singing-starting time point and continues to be generated for the determined singing duration time.

Abstract: Partial waveform data representative of a waveform shape variation are extracted from supplied waveform data, and the extracted partial waveform data are stored along with time position information indicative of their respective time positions. In reproduction, the partial waveform data and time position information are read out, then the partial waveform data are arranged on the time axis in accordance with the time position information, and a waveform is produced on the basis of the waveform data arranged on the time axis. In another implementation, sets of sample identification information and time position information are obtained in accordance with a performance tone waveform to be reproduced, and sample data are obtained from a database in accordance with the sample identification information. The thus-obtained sample data are arranged on the time axis in accordance with the time position information, and the desired waveform is produced on the basis of the sample data arranged on the time axis.

Abstract: A quadratic phase interpolation method for synthesis of musical tones incorporates both phase and frequency measurements at the boundaries of a data frame using a weighted least square algorithm approach. The approach assumes that the true frequency and phase at the two ends of a data frame conform to a quadratic phase model and that exact match between measured phase and frequency with the quadratic model is not necessary because of the noise in the measurements.

Abstract: The present invention is directed to classifying a musical piece based on determined characteristics for each of plural notes contained within the piece. Exemplary embodiments accommodate the fact that in a continuous piece of music, the starting and ending points of a note may overlap previous notes, the next note, or notes played in parallel by one or more instruments. This is complicated by the additional fact that different instruments produce notes with dramatically different characteristics. For example, notes with a sustaining stage, such as those produced by a trumpet or flute, possess high energy in the middle of the sustaining stage, while notes without a sustaining stage, such as those produced by a piano or guitar, posses high energy in the attacking stage when the note is first produced.

Abstract: A synthesized strong hit waveform data is generated by synthesizing a strong hit amplitude component separated from a strong hit waveform data generated based on tone of a strong play and a weak hit phase component separated from a weak hit waveform data generated based on tone of a weak play. This synthesized strong hit waveform data is stored in a memory together with a weak hit waveform data. A musical tone signal is generated by interpolating the weak hit waveform data and the synthesized strong hit waveform data read out from the memory, in response to a signal supplied from external portion.

Abstract: Harmonic signals of frequencies have a relation fn=f1*n* .sqroot.1+C*n.sup.2 /.sqroot.1+C where C is a constant and n is a harmonic number relative to the fundamental frequency f1, are generated and synthesized to produce a modified sound that can be heard comfortably by human ears.

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: For use in a synthesizer having a wave source that produces a periodic wave, frequency shifting circuitry for frequency-shifting the periodic wave and waveshaping circuitry for transforming the periodic wave into a waveform containing a formant, the frequency-shifting causing displacement of the formant, a circuit for, and method of, compensating for the displacement and a synthesizer employing the circuit or the method. In one embodiment, the circuit includes bias circuitry, coupled to the wave source and the frequency shifting circuitry, that introduces a bias into the periodic wave based on a degree to which the frequency shifting circuitry frequency shifts the periodic wave, the bias reducing a degree to which the formant is correspondingly frequency-shifted.

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 musical sound synthesizer generates a predetermined singing sound based on performance data. A compression device determines whether each of a plurality of phonemes forming the predetermined singing sound is a first phoneme to be sounded in accordance with a note-on signal indicative of a note-on of the performance data, and compresses a rise time of the first phoneme when the first phoneme is sounded in accordance with occurrence of the note-on signal of the performance data.

Abstract: A parameter setting apparatus is provided in an electronic music instrument for editing a configure of parameters in accordance with a manipulation amount to set a tone generator which creates a desired timbre of musical tones upon setting. In the apparatus, a first memory device stores a first subset of parameters containing a Boolean parameter having a first Boolean value and a numerical parameter having a first numerical value. A second memory device stores a second subset of parameters containing the Boolean parameter having a second Boolean value and the numerical parameter having a second numerical value. An operating device provides a manipulation amount for editing parameters. An interpolating device interpolates the first numerical value and the second numerical value in accordance with the manipulation amount to determine an edited numerical value of the numerical parameter.

Abstract: A musical sound synthesizing apparatus synthesizes plural-series overtones having different inharmonicity and amplitude level so as to generate a musical sound close to a natural sound. Inharmonicity coefficients K1 and K2 are read out for each overtone of different series in accordance with a timbre and a pitch, preferably modified according to a touch velocity, and then inputted to the first and second overtone frequency generators 19 and 20. Two series of overtones respectively computed in first and second overtone generators 21 and 22 on the basis of fundamental and overtone frequencies computed using of the coefficients K1 and K2, are controlled in its amplitude by the multipliers 25 and 26, and then synthesized so as to be outputted as a musical sound signal. In the multipliers 25 and 26, respective overtones are multiplied by amplitude coefficients Lk1 and Lk2 determined for each series in accordance with the touch velocity so that their amplitude levels are modified.

Abstract: An improved control structure for music synthesis is provided in which: 1) the sound representation provided to the adaptive function mapper allows for a greatly increased degree of control over the sound produced; and 2) training of the adaptive function mapper is performed using an error measure, or error norm, that greatly facilitates learning while ensuring perceptual identity of the produced sound with the training example. In accordance with one embodiment of the invention, sound data is produced by applying to an adaptive function mapper control parameters including: at least one parameter selected from the set of time and timbre space coordinates; and at least one parameter selected from the set of pitch, .DELTA.pitch, articulation and dynamic. Using an adaptive function mapper, mapping is performed from the control parameters to synthesis parameters to be applied to a sound synthesizer.

Abstract: An improved method for shifting the pitches of a tone is disclosed. It comprises: (a) subjecting a digitized original waveform to a whitening process using an all-zero filter (AZF) to obtain a whitened waveform; (b) resampling the whitened waveform at a desired scaling ratio to obtain a scaled and whitened waveform; (c) subjecting the scaled and whitened waveform to a coloring process using an all-pole filter (APF) to obtain a synthesized waveform. In a preferred embodiment, the all-zero filter performs the transformation function of: ##EQU1## and the all-pole filter performs the transformation function of: ##EQU2## wherein the a.sub.i 's and b.sub.i 's are linear predictive coefficients. The whitened waveforms can be compressed and stored as wavetables, which can be subsequently retrieved and decompressed before resampling.

Abstract: An electronic musical instrument comprises an analysis section, an excitation-waveform memory and a synthesis section. In the analysis section, difference data, which are calculated between target-sound data and output of an analysis loop, are subjected to compressive coding to produce compressed data. The compressed data are stored in the excitation-waveform memory as excitation-waveform data. The analysis loop, containing at least a delay circuit, is driven by an excitation signal which is produced by expanding the compressed data. In the synthesis section, the excitation-waveform data, read out from the excitation-waveform memory, are expanded; and expanded data are added to output of a synthesis loop, containing at least a delay circuit, so as to produce musical tone data representative of a musical tone to be generated.

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 MIDI playback system is described in which a sequencer is characterized by: logic for generating a sequence of events, the sequence comprising groups of events, each group comprising events to be executed within a first time interval, the groups being separated in the sequence by marker events for indicating the time intervals, the sequencer being arranged to send the sequence of events together to a synthesizer. A synthesizer is described comprising storage means to receive and store the sequences of events; logic for reading an event from the storage; and logic for determining if the event is a marker event, the synthesizer being arranged to wait, if the event is a marker event, a time equal to the first time interval before reading another event from storage. Using this arrangement, the sequencer is no longer required to schedule the MIDI commands precisely on time and therefore the timing services that a software implementation of such a sequencer might ask from a computer system are reduced.