摘要:
For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length.
摘要:
An sound encoder accepts a sound signal and then produces a plurality of codes which represent the sound signal on a frame-by-frame basis. The sound encoder determines the order in which the plurality of codes is to be multiplexed into a multiplexed code based on one of the plurality of codes on a frame-by-frame basis, multiplexes the plurality of codes one by one into a multiplexed code in the determined order, and acquires an error correction code for the multiplexed code. The sound encoder then outputs the multiplexed code including the acquired error correction code added to the end thereof as a sound code.
摘要:
In a method for characterizing a signal, which represents an audio content, a measure for a tonality of the signal is determined, whereupon a statement is made about the audio content of the signal based on the measure for the tonality of the signal. The measure for the tonality of the signal for the content analysis is robust against a signal distortion, such as by MP3 encoding, and has a high correlation to the content of the examined signal.
摘要:
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. By arbitrarily selecting coefficients for compression and expansion which are respectively performed in the analysis section and synthesis section , the musical tone data are controlled to be an equivalence of the target-sound data. Further, the excitation-waveform memory is designed to merely store compressed excitation-waveform data, so capacity required for the memory can be reduced.
摘要:
Compressed waveform data structure is proposed which is suited for segmentation of a plurality of samples of compressed waveform data into a plurality of frames and subsequent storage of each of the frames. The number of bits per sample of the compressed waveform data is variable between the frames, but uniform, i.e. the same among all of the samples, within each of the frames. Each of the frames has a same data storage size. Each of the frames includes, in a predetermined layout, an auxiliary information area for storing auxiliary information that includes compression-related information to be used for decompressing the compressed waveform data, and a data area for storing a plurality of samples of the compressed waveform data of the frame with each of the samples comprising a same number of bits.
摘要:
Compressed waveform data structure is proposed which is suited for segmentation of a plurality of samples of compressed waveform data into a plurality of frames and subsequent storage of each of the frames. The number of bits per sample of the compressed waveform data is variable between the frames, but uniform, i.e. the same among all of the samples, within each of the frames. Each of the frames has a same data storage size. Each of the frames includes, in a predetermined layout, an auxiliary information area for storing auxiliary information that includes compression-related information to be used for decompressing the compressed waveform data, and a data area for storing a plurality of samples of the compressed waveform data of the frame with each of the samples comprising a same number of bits. Thus, respective start positions of the frames and compressed waveform data in a memory can be fixed at predetermined positions common to the frames, so that readout control can be performed with ease.
摘要:
A synthesis of percussion musical instruments sounds is provided using a microprocessor (17) that implements an all pole lattice filter and applying either a single impulse signal to the filter or N samples of an excitation signal sequence to the filter by a memory (19). The coefficients of the filter are determined by storing digital samples (501) of desired musical note from a desired percussion instrument, generating a Fourier transform to get a spectrum (502), picking the peaks of the spectrum (503) to select the most prominent components in the spectrum and determining wanted frequencies for decaying sine waves and for the frequencies finding the time envelope and estimating therefrom the pole radius.
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 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.