摘要:
A digital-to-analog conversion device which performs integration processing for integrating a difference between an input signal and a first return signal generated based on the input signal, and outputting an integration result, first quantization processing for quantizing the integration result, and outputting a first quantization signal, first return signal output processing for outputting the first return signal by adding to the first quantization signal a correction value delay signal acquired by a correction value signal outputted based on the integration result being delayed, and output processing for outputting output signals including a signal whose pulse width is asymmetrical to center of a processing period, based on the first quantization signal, in which the correction value signal includes a signal indicating a correction value for correcting a difference between a center of the pulse width asymmetrical to the center of the processing period and the center of the processing period.
摘要:
A method for the synthetic generation of a digital audio signal by means of periodic sampling of a waveform shall permit the user a particularly simple and intuitive access to the changing and creative transformation of the waveform on which the sampling is based. For this purpose, according to the invention, the waveform is specified by using control points, which, in addition to position parameters, may contain further attributes, of which the parameters and attributes can be changed individually over time by means of control signals or spontaneouseous intervention. The control-point values which result in this way can be interpreted either as direct amplitude-period phase or as magnitude-frequency or phase-frequency pairs. A continuous waveform is generated by interpolation or approximation of the control points and the parameters/attributes of the latter, which assume a time-specific value depending on the current control signals and other influences, and is used for further processing, e.g. spectral band limiting.
摘要:
Methods and apparatus for manipulating physical objects to produce sounds that are correlated with the feel of the objects being manipulated. Example devices called the PebbleBox, CrumbleBag and ScrubberGlove each use the manipulation of physical objects of arbitrary material as the basis for interacting with granular sound synthesis models. The sounds made by the objects as they are manipulated produce sound signal events that are detected and used to trigger and control the reproduction of stored sound samples in real time.
摘要:
A method includes receiving an audio signal and identifying one or more steady-state segments of the audio signal. The method also includes identifying at least one portion of the one or more segments that contains a specified frequency. Further, the method includes generating a wavetable using the at least one identified portion of the one or more segments. In addition, the method could include synthesizing an output audio signal using the wavetable. The output audio signal could represent a ringtone in a mobile telephone.
摘要:
A method for shifting the timbre and/or pitch of an input signal samples the input signal at a first rate and stores the samples in a memory buffer. A digital signal processor resamples the stored input signal at a rate that differs from the first rate at which the input note is originally sampled and stores the resampled input signal in a second memory buffer. A pitch shifter shifts the pitch of the input signal by periodically scaling the resampled input signal by a window function to create an output signal. The rate at which the resampled data is replicated by the window function determines the pitch of the output signal.
摘要:
An improved method for generating various wave functions for use in musical tone generation in FM synthesis so as to substantially reduce computational and storage requirements. The waveform desired to be used is mathematically modeled using piecewise linear approximation. A reduced number of coefficients of the piecewise linear approximation, representing the angles and offsets of the linear segments, are stored in memory.
摘要:
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.
摘要:
A musical tone generating apparatus has an waveform memory which stores original waveform data. One or two tone generating chips are be able to fixed on a print circuit board as elements of the musical tone generating apparatus. Each tone generating chip, provided on the print circuit board, sequentially generates waveform data of a plurality of musical tones at sampling periods having a predetermined length under time division control. Each tone generating chip sequentially carries out tone generating operations to generate the waveform data based on the original waveform data during time division channels which are obtained by dividing each one of the sampling periods when the musical tone generating section is used for tone generation. In the case where one tone generating chip is employed on the print circuit board, N samples of the original waveform data are read out from the waveform memory during each one of the time division channels to be used by the tone generating chip. In the case where two tone generating chips are employed on the print circuit board, the number of the original waveform data read out from the waveform memory for each tone generating chip during each time division channel is decreased from N to M which is less than N.
摘要:
A method for shifting the timbre and/or pitch of an input signal samples the input signal at a first rate and stores the samples in a memory buffer. A digital signal processor resamples the stored input signal at a rate that differs from the first rate at which the input note is originally sampled and stores the resampled input signal in a second memory buffer. A pitch shifter shifts the pitch of the input signal by periodically scaling the resampled input signal by a window function to create an output signal. The rate at which the resampled data is replicated by the window function determines the pitch of the output signal.
摘要:
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 wave form 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. When the bit compression of digital signal data is performed by selecting a filter with blocks each consisting of plural samples as units, a pseudo signal is affixed to the input signal, before the start point of the input signal, which pseudo signal will cause a filter of the lowest order to be selected. The looping domain is set so as to be a whole number multiple of the block which serves as the unit for bit compression, and the parameters of the looping start block are formed on the basis of data of the start and the end blocks. By applying a part or the whole of the signal processing method to a sound source data forming apparatus, sound source data may be formed which is reduced in the looping noise and error caused by data compression and which is of superior sound quality.