公开(公告)号：US20170076707A1

公开(公告)日：2017-03-16

申请号：US15265820

申请日：2016-09-14

申请人： CASIO COMPUTER CO., LTD.

发明人： Goro SAKATA

IPC分类号： G10H7/00 ,  G10H7/04 ,  G06F17/30

CPC分类号： G10H7/008 ,  G06F17/30778 ,  G10H1/0041 ,  G10H7/02 ,  G10H7/04 ,  G10H2240/071 ,  G10H2240/131 ,  G10H2240/161 ,  G10H2250/595 ,  G10H2250/601 ,  G10L19/167 ,  G11B20/1251

摘要： A waveform data structure includes a plurality of types of frames having different data sizes. Each of the plurality of types of frames includes an auxiliary information area and a data area. The auxiliary information area includes an area for storing common effective-bit length data for a section of waveform samples, and an area for storing an identifier for identifying one of the plurality of types of frames. The data area is an area for storing extracted waveform samples which are extracted from the waveform samples based on the common effective-bit length. The number of the extracted waveform samples is determined based on the common effective-bit length.

摘要翻译： 波形数据结构包括具有不同数据大小的多种类型的帧。 多个类型的帧中的每一个包括辅助信息区和数据区。 辅助信息区域包括用于存储一部分波形采样的公共有效位长度数据的区域和用于存储用于识别多种类型的帧之一的标识符的区域。 数据区域是用于存储基于公共有效位长度从波形采样中提取的提取的波形样本的区域。 基于公共有效位长度来确定提取的波形样本的数量。

公开(公告)号：US20130246073A1

公开(公告)日：2013-09-19

申请号：US13727370

申请日：2012-12-26

申请人： CASIO COMPUTER CO., LTD.

发明人： Goro SAKATA

IPC分类号： G10L19/04

CPC分类号： G10L19/04 ,  G10H1/0041 ,  G10H7/002 ,  G10H2230/041 ,  G10H2250/601 ,  G10L19/08 ,  H03M7/30

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。

公开(公告)号：US07378586B2

公开(公告)日：2008-05-27

申请号：US10676205

申请日：2003-09-30

申请人： Masatsugu Okazaki

发明人： Masatsugu Okazaki

IPC分类号： G10H7/00

CPC分类号： G10H7/12 ,  G10H7/02 ,  G10H2250/585 ,  G10H2250/595 ,  G10H2250/601 ,  G10H2250/621 ,  G10H2250/641

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。

公开(公告)号：US20040267522A1

公开(公告)日：2004-12-30

申请号：US10484513

申请日：2004-08-09

发明人： Eric Allamanche ,  Jurgen Herre ,  Oliver Hellmuth ,  Thorsten Kastner

IPC分类号： G10L019/14

CPC分类号： G10H1/0008 ,  G10H2240/061 ,  G10H2240/135 ,  G10H2250/235 ,  G10H2250/601 ,  G10L15/04 ,  G10L19/0204

摘要： In a method for characterizing a signal representing an audio content a measure is determined for a tonality of the signal, whereupon a statement is made about the audio content of the signal on the basis of the measure for the tonality of the signal. The measure for the tonality is derived from a quotient whose numerator is the mean of the summed values of spectral components of the signal exponentiated with a first power and whose denominator is the mean of the summed values of spectral components exponentiated with a second power, the first and second powers differing from each other. The measure for the tonality of the signal for the content analysis is robust in relation to a signal distortion, due e.g. to MP3 coding, and has a high correlation with the content of the analysed signal.

摘要翻译： 在表征音频内容的信号的表征方法中，针对信号的音调确定了一个度量，然后根据该信号音调的度量，对该信号的音频内容做出声明。 音调的度量来自商，其分子是以第一功率取幂的信号的频谱分量的总和值的平均值，其分母是用第二功率指数的频谱分量的总和值的平均值， 第一和第二权力彼此不同。 用于内容分析的信号的音调的度量相对于信号失真是鲁棒的，例如。 到MP3编码，并且与分析的信号的内容具有高度的相关性。

公开(公告)号：US20030019348A1

公开(公告)日：2003-01-30

申请号：US10192563

申请日：2002-07-11

发明人： Hirohisa Tasaki

IPC分类号： G10H001/18

CPC分类号： G10H7/00 ,  G10H2240/191 ,  G10H2250/601

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。

公开(公告)号：US5872727A

公开(公告)日：1999-02-16

申请号：US752014

申请日：1996-11-19

申请人： Chih-Chung Kuo

发明人： Chih-Chung Kuo

IPC分类号： G10H1/12 ,  G10H1/20 ,  G06F17/10 ,  G10H1/06 ,  G10H7/00

CPC分类号： G10H1/20 ,  G10H1/125 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/255 ,  G10H2250/601 ,  G10H2250/621 ,  G10H2250/631

摘要： 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.

摘要翻译： 公开了一种改变音调音高的方法。 它包括：（a）使用全零滤波器（AZF）对数字化的原始波形进行白化处理以获得白化波形; （b）以期望的比例比对白化的波形进行重采样以获得标度和白化的波形; （c）使用全极滤波器（APF）对经缩放和白化的波形进行着色处理，以获得合成波形。 在一个优选实施例中，全零滤波器执行以下变换函数：＆lt; IMAGE＆gt;全极滤波器执行以下变换函数：＆lt; IMAGE＆gt;其中，ai和bi是线性预测系数。 白化的波形可以压缩并存储为波形图，可以在重新采样之前随后检索和解压缩。

公开(公告)号：US5693901A

公开(公告)日：1997-12-02

申请号：US627366

申请日：1996-04-04

申请人： Kaoru Matsunaga

发明人： Kaoru Matsunaga

IPC分类号： G10H1/12 ,  G10H7/02 ,  G10H7/00 ,  G10H7/10

CPC分类号： G10H7/02 ,  G10H1/125 ,  G10H2250/591 ,  G10H2250/595 ,  G10H2250/601 ,  Y10S84/09

摘要： 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. In the second embodiment, a waveform that is to be repeatedly read is coded by a prediction filter, for which a prediction filter coefficient is set so that the result of decoding at the repeated reading head portion matches each time.

摘要翻译： 根据第一发明，提供一种电子乐器，其对通过DPCM方法或ADPCM方法压缩的波形进行解码和读取，该波形存储与每个波形相符合的预测滤波器系数，并且通过使用预测来再现乐音 滤波系数。 在第一发明中，存储在电子乐器中的波形与准备了波形时使用的预测滤波器系数一起存储，并且对于每个波形采用最佳预测滤波器系数来再现波形。 根据第二发明，提供了一种电子乐器，其对由DPCM方法或ADPCM方法压缩的波形进行解码并重复读取解码数据，其可以在环路顶部重复读取波形数据，而不需要设备设置 解码装置。 在第二实施例中，要重复读取的波形由预测滤波器编码，预测滤波器系数被设置为使得重复读头部分的解码结果与每次匹配。

公开(公告)号：US5430241A

公开(公告)日：1995-07-04

申请号：US438088

申请日：1989-11-16

申请人： Makoto Furuhashi ,  Masakazu Suzuoki ,  Ken Kutaragi

发明人： Makoto Furuhashi ,  Masakazu Suzuoki ,  Ken Kutaragi

IPC分类号： G10H3/12 ,  G10H7/08 ,  G10K15/02 ,  G10L19/02 ,  G10H7/00

CPC分类号： G10K15/02 ,  G10H3/125 ,  G10H7/08 ,  G10L19/02 ,  G10H2210/066 ,  G10H2250/105 ,  G10H2250/235 ,  G10H2250/281 ,  G10H2250/601 ,  Y10S84/09

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 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. 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，由梳状滤波器提取输入信号的基频及其高次谐波分量，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。

公开(公告)号：US5119425A

公开(公告)日：1992-06-02

申请号：US780836

申请日：1991-10-23

申请人： Paul A. Rosenstrach ,  Michael A. Deaett

发明人： Paul A. Rosenstrach ,  Michael A. Deaett

IPC分类号： G10H7/00 ,  G10K15/02 ,  H03M7/30

CPC分类号： G10H7/00 ,  G10K15/02 ,  H03M7/3082 ,  G10H2250/015 ,  G10H2250/211 ,  G10H2250/601

摘要： A pseudo-random or transient synthesized signal is provided by analysis of a plurality of related signals by vector quantization of linear predictive coding coefficients (cluster representatives) of time blocks of the signals and providing cumulative probability matrices for the transition from one cluster representative for one block to a cluster representative of the next successive block of each of the signals. Synthesis of the pseudo-random signal is provided by randomly selecting according to a cumulative transition probability, the cluster representative of a next successive block given the selected cluster representative of the previous block, the coefficient of each block time being applied to a noise-excited recursive filter to generate the pseudo-random synthesized signal. Synthesis includes probabalistic models using Markov transitions, to produce transient sounds such as sonar, hatch closings, and hull groans.

摘要翻译： 通过对信号的时间块的线性预测编码系数（簇代表）的矢量量化，通过分析多个相关信号来提供伪随机或瞬态合成信号，并提供用于从一个簇代表转换的累积概率矩阵 块代表代表每个信号的下一个连续块的簇。 通过随机选择累积转移概率来提供伪随机信号的合成，表示给定表示先前块的所选择的簇的下一个连续块的簇，将每个块时间的系数应用于噪声激励 递归滤波器生成伪随机合成信号。 综合包括使用马尔可夫转换的概率模型，产生瞬时声音，如声纳，舱口关闭和船体呻吟声。

公开(公告)号：US20110231193A1

公开(公告)日：2011-09-22

申请号：US13151660

申请日：2011-06-02

申请人： Yao Qian ,  Frank Soong

发明人： Yao Qian ,  Frank Soong

IPC分类号： G10L13/08

CPC分类号： G10H1/06 ,  G10H7/12 ,  G10H2210/201 ,  G10H2240/056 ,  G10H2250/015 ,  G10H2250/455 ,  G10H2250/471 ,  G10H2250/601

摘要： Various technologies for generating a synthesized singing voice waveform. In one implementation, the computer program may receive a request from a user to create a synthesized singing voice using the lyrics of a song and a digital file containing its melody as inputs. The computer program may then dissect the lyrics' text and its melody file into its corresponding sub-phonemic units and musical score respectively. The musical score may be further dissected into a sequence of musical notes and duration times for each musical note. The computer program may then determine a fundamental frequency (F0), or pitch, of each musical note.

摘要翻译： 用于生成合成歌唱声音波形的各种技术。 在一个实现中，计算机程序可以使用歌曲的歌词和包含其旋律的数字文件作为输入来接收来自用户的请求以创建合成歌唱声音。 然后，计算机程序可以分别将歌词的文本及其旋律文件分解成其对应的子音素单元和乐谱。 乐谱可以进一步解剖为每个音符的一系列音符和持续时间。 然后，计算机程序可以确定每个音符的基本频率（F0）或音高。