Coding device, coding method, decoding device, decoding method, and storage medium
    1.
    发明授权
    Coding device, coding method, decoding device, decoding method, and storage medium 有权
    编码装置,编码方法,解码装置,解码方法和存储介质

    公开(公告)号:US09165563B2

    公开(公告)日:2015-10-20

    申请号:US13727370

    申请日:2012-12-26

    发明人: Goro Sakata

    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.

    摘要翻译: 对于要编码的声音的波形数据的各个采样数据,计算预测残差值作为采样残差数据,并根据该残差波形数据计算有效位长度。 然后,对于有效比特长度数据,生成作为公共有效实际数据的处理对象之间的最大有效比特长度,以及其中表示公共有效比特长度的公共有效实际数据和信息以预定配置格式排列的编码数据 被生成。 分析包含在编码数据中的信息,并且提取多个公共有效位信息中的每一个。 然后,通过对通过添加除了公共有效位长度以外的部分的比特扩展进行解压缩的残差波形数据的分析结果进行逆线性预测处理来解码声音的波形数据。

    Sound encoder and sound decoder
    2.
    发明授权
    Sound encoder and sound decoder 有权
    声音编码器和声音解码器

    公开(公告)号:US07315817B2

    公开(公告)日:2008-01-01

    申请号:US10192563

    申请日:2002-07-11

    申请人: Hirohisa Tasaki

    发明人: Hirohisa Tasaki

    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.

    摘要翻译: 声音编码器接收声音信号,然后在逐帧的基础上产生表示声音信号的多个代码。 声音编码器在逐帧的基础上,基于多个码中的一个,将多个码多路复用的顺序决定为多路复用码,将多个码逐一复用为多路复用码 确定的顺序,并获取多路复用码的纠错码。 然后,声音编码器将包括添加到其末尾的获取的纠错码的多路复用码输出为声码。

    Electronic musical instrument with reduced storage of waveform
information
    4.
    发明授权
    Electronic musical instrument with reduced storage of waveform information 失效
    电子乐器减少了波形信息的存储

    公开(公告)号:US5777249A

    公开(公告)日:1998-07-07

    申请号:US548434

    申请日:1995-10-26

    申请人: Hideo Suzuki

    发明人: Hideo Suzuki

    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.

    摘要翻译: 电子乐器包括分析部分,激励波形存储器和合成部分。 在分析部分中,将在目标声音数据和分析循环的输出之间计算的差分数据进行压缩编码以产生压缩数据。 压缩数据作为激励波形数据存储在激励波形存储器中。 包含至少延迟电路的分析回路由通过扩展压缩数据产生的激励信号驱动。 在合成部分中,从激励波形存储器读出的激励波形数据被扩展; 扩展数据被添加到包含至少延迟电路的合成循环的输出,以产生表示要生成的乐音的乐音数据。 通过分别在分析部分和综合部分中执行的任意选择压缩和扩展的系数,将乐音数据控制为目标声音数据的等价物。 此外,激励波形存储器被设计为仅存储压缩的激励波形数据,因此可以减少存储器所需的容量。

    Compressed data structure and apparatus and method related thereto
    5.
    发明授权
    Compressed data structure and apparatus and method related thereto 有权
    压缩数据结构及其相关的装置和方法

    公开(公告)号:US07521621B2

    公开(公告)日:2009-04-21

    申请号:US11860484

    申请日:2007-09-24

    IPC分类号: G10H7/00

    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.

    摘要翻译: 提出了压缩波形数据结构,其适用于将压缩波形数据的多个样本分割成多个帧并随后存储每个帧。 每个样本的压缩波形数据的位数在帧之间是可变的,但在每个帧内是均匀的,即所有采样中的相同。 每个帧具有相同的数据存储大小。 每个帧以预定布局包括用于存储辅助信息的辅助信息区域,该辅助信息区域包括用于解压缩压缩波形数据的压缩相关信息,以及用于存储压缩波形数据的多个样本的数据区域 其中每个样本包括相同数量的比特。

    Compressed data structure and apparatus and method related thereto

    公开(公告)号:US20070240556A1

    公开(公告)日:2007-10-18

    申请号:US11810872

    申请日:2007-06-06

    IPC分类号: G10H7/00

    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.

    Synthesis of percussion musical instrument sounds
    7.
    发明授权
    Synthesis of percussion musical instrument sounds 失效
    敲击乐器合成声音

    公开(公告)号:US6111181A

    公开(公告)日:2000-08-29

    申请号:US72400

    申请日:1998-05-04

    IPC分类号: G10H1/12 G10H7/00

    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.

    摘要翻译: 使用实现全极点格子的微处理器(17)提供打击乐器声音的合成,并且通过存储器(19)向滤波器施加单个脉冲信号或向滤波器施加激励信号序列的N个样本。 通过从期望的打击乐器存储所需音符的数字样本(501)来确定滤波器的系数,产生傅立叶变换以获得频谱(502),拾取频谱的峰值(503)以选择最突出的 确定频谱中的分量,并确定衰减正弦波的有用频率以及找到时间包络的频率,并从其估计极半径。

    Musical tone generating apparatus
    9.
    发明授权
    Musical tone generating apparatus 失效
    音乐发生装置

    公开(公告)号:US5625158A

    公开(公告)日:1997-04-29

    申请号:US360208

    申请日:1994-12-20

    申请人: Tetsuji Ichiki

    发明人: Tetsuji Ichiki

    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.

    摘要翻译: 乐音产生装置具有存储原始波形数据的波形存储器。 一个或两个音调生成芯片能够固定在作为乐音产生装置的元件的打印电路板上。 设置在打印电路板上的每个音调生成芯片在时分控制下以具有预定长度的采样周期顺序地产生多个乐音的波形数据。 每个乐音产生芯片依次执行乐音产生操作,以便在通过将乐音产生部分用于音调产生时将采样周期中的每一个获得的时分信道中的原始波形数据生成波形数据。 在印刷电路板上使用一个音调生成芯片的情况下,在由音调生成芯片使用的每个时分信道中,从波形存储器中读出原始波形数据的N个样本。 在印刷电路板上使用两个音调生成芯片的情况下,在每个时分信道期间从每个音调生成芯片的波形存储器读出的原始波形数据的数目从N减小到小于N的M 。

    Signal processing method and sound source data forming apparatus
    10.
    发明授权
    Signal processing method and sound source data forming apparatus 失效
    信号处理方法和声源数据形成装置

    公开(公告)号:US5519166A

    公开(公告)日:1996-05-21

    申请号:US330329

    申请日:1994-10-27

    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.

    摘要翻译: 一种用于处理通过数字化诸如乐器声音信号的模拟信号产生的数字信号的方法和用于产生声源数据的装置。 当输入信号包含周期性重复的波形部分时,在信号处理之前通过梳状滤波器提取输入信号的基频和其高次谐波分量,这利用了输入信号的周期。 通过执行傅里叶变换以产生频率分量,相位匹配这些频率分量并执行逆傅里叶变换来检测基频或音调。 当提取重复波形部分或所谓的循环域时,选择在该域的两端附近具有最高波形相似度的循环域。 当通过选择具有由多个样本组成的块的滤波器来执行数字信号数据的位压缩时,在输入信号的起始点之前将伪信号附加到输入信号,该伪信号将导致滤波器 的最低订单选择。 循环域被设置为作为用于位压缩的单位的块的整数倍,并且循环开始块的参数基于起始和结束块的数据形成。 通过将一部分或全部的信号处理方法应用于声源数据形成装置,可以形成声源数据,其减少了由数据压缩引起的循环噪声和误差,并且具有优良的音质。