公开(公告)号：US5086475A

公开(公告)日：1992-02-04

申请号：US436423

申请日：1989-11-14

申请人： Ken Kutaragi ,  Makoto Furuhashi ,  Masakazu Suzuoki ,  Koji Kageyama

发明人： Ken Kutaragi ,  Makoto Furuhashi ,  Masakazu Suzuoki ,  Koji Kageyama

IPC分类号： G10H1/057 ,  G10H5/00 ,  G10H7/08 ,  G10K15/02 ,  G10L21/04

CPC分类号： G10K15/02 ,  G10H1/057 ,  G10H5/005 ,  G10H7/08 ,  G10L21/04 ,  G10H2210/066 ,  G10H2250/105 ,  G10H2250/121 ,  G10H2250/145 ,  G10H2250/235 ,  G10H2250/545 ,  G10H2250/601 ,  G10H2250/621 ,  G10H2250/631 ,  G10H2250/641

摘要： An apparatus for looping or data-compressing sampled waveform data digitized from musical sound signals (or the like) to produce sound source data, recording the sound source data on a storage medium, and reading out the sound source data from the storage medium for reproduction. To eliminate amplitude discontinuities at repetition points during looping, two connection samples of repetitive waveform portions having values closest to each other are selected from actual samples and interpolated samples. An interpolation filter performs multiple oversampling to produce the interpolated samples. The interpolation filter includes a filter for each degree of oversampling, and all the filters have the same amplitude characteristics. By asserting pulse code modulated data at the beginning portion of a looping domain, adverse compression effects can be avoided without the necessity of providing compression parameters. When reading out sound source data from the storage medium, a data start address and a looping start address are loaded, in that order, into an address generator. A discriminating flag indicating the presence or absence of the looping domain and a discriminating flag indicating the end of the sound source data can be included in the sound source data to facilitate control of looping or end of reproduction.

摘要翻译： 一种用于循环或数据压缩从音乐声音信号（或类似数字）数字化的采样波形数据的装置，以产生声源数据，将声源数据记录在存储介质上，并从存储介质读出声源数据以进行再现 。 为了消除在循环过程中的重复点处的振幅不连续性，从实际样本和内插样本中选择具有彼此最接近的值的重复波形部分的两个连接样本。 内插滤波器执行多次过采样以产生内插采样。 内插滤波器包括用于每次过采样度的滤波器，并且所有滤波器具有相同的幅度特性。 通过在循环域的开始部分断言脉冲编码调制数据，可以避免不利的压缩效应，而不需要提供压缩参数。 当从存储介质读出声源数据时，依次将数据起始地址和循环起始地址加载到地址生成器中。 指示循环域的存在或不存在的识别标志和指示声源数据结束的鉴别标记可以包括在声源数据中，以便于控制循环或再现结束。

公开(公告)号：US20160275956A1

公开(公告)日：2016-09-22

申请号：US15031274

申请日：2014-10-22

申请人： ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY ,  WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.

发明人： Taegyu LEE ,  Hyunoh OH ,  Jeongil SEO ,  Yongju LEE ,  Seungkwon BEACK ,  Kyeongok KANG ,  Daeyoung JANG ,  Youngcheol PARK ,  Daehee YOUN

IPC分类号： G10L19/008 ,  H04S3/00

CPC分类号： G10L19/008 ,  G10H2250/111 ,  G10H2250/145 ,  H04R3/00 ,  H04R5/033 ,  H04S3/00 ,  H04S3/002 ,  H04S3/004 ,  H04S3/008 ,  H04S2400/01 ,  H04S2420/03

摘要： The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor.To this end, provided are a method for generating a filter of an audio signal, including: receiving at least one proto-type filter coefficient for filtering each subband signal of an input audio signal; converting the proto-type filter coefficient into a plurality of subband filter coefficients; truncating each of the subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and generating FFT filter coefficients by fast Fourier transforming (FFT) the truncated subband filter coefficients by a predetermined block size in the corresponding subband and a parameterization unit using the same.

摘要翻译： 为此，提供了一种用于产生音频信号的滤波器的方法，包括：接收用于滤波输入音频信号的每个子带信号的至少一个原型滤波器系数; 将原型滤波器系数转换为多个子带滤波器系数; 基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息而获得的滤波器顺序信息来截断每个子带滤波器系数，至少一个截断的子带滤波器系数的长度与另一个截止的子带滤波器系数的长度不同 子带 以及通过在相应子带中以预定的块大小快速傅里叶变换（FFT）所述截断的子带滤波器系数并使用该FFT滤波器系数的参数化单元来生成FFT滤波器系数。

公开(公告)号：US20160179458A1

公开(公告)日：2016-06-23

申请号：US14579201

申请日：2014-12-22

申请人： Fractal Audio Systems

发明人： Clifford Chase

IPC分类号： G06F3/16

CPC分类号： G06F3/165 ,  G10H1/0091 ,  G10H1/125 ,  G10H1/14 ,  G10H2250/051 ,  G10H2250/145 ,  G10H2250/631 ,  H03H17/0223 ,  H03H17/0266 ,  H03H17/0283

摘要： The specification relates to the processing of a digital input signal. The digital input signal is convolved with a first time slice of an impulse response with a first direct convolution filter. In parallel, the digital input signal is divided into multiple frequency bands, and each of the multiple frequency bands of the digital input signal are convolved with a second time slice of the impulse response using an equal number of direct convolution filters. The outputs of the equal number of direct convolution filters are recombined and a delay compensation is applied to the recombined outputs. An output of the direct convolution filter is summed with the delayed and recombined outputs to form a zero-to-near-zero latency convolution output.

摘要翻译： 本说明书涉及数字输入信号的处理。 数字输入信号与具有第一直接卷积滤波器的脉冲响应的第一时间片卷积。 并行地，数字输入信号被分成多个频带，并且数字输入信号的多个频带中的每个频带与使用相等数量的直接卷积滤波器的脉冲响应的第二时间片卷积。 相等数量的直接卷积滤波器的输出被重组，并且对重组输出施加延迟补偿。 直接卷积滤波器的输出与延迟和重新组合的输出相加，以形成零到近零的等效卷积输出。

公开(公告)号：US08363843B2

公开(公告)日：2013-01-29

申请号：US11713167

申请日：2007-03-01

申请人： Steffan Diedrichsen

发明人： Steffan Diedrichsen

IPC分类号： H03G3/00

CPC分类号： H04S3/008 ,  G10H2210/281 ,  G10H2210/301 ,  G10H2250/111 ,  G10H2250/145 ,  G10K15/08 ,  G10L19/008

摘要： The present invention provides a method of generating, on a data processing system, a multi-channel audio convolution reverb, said method comprising providing a plurality of impulse responses corresponding to a desired room to be simulated; receiving, in input, multi-channel audio sample data; performing, for each respective audio channel, same channel convolution operation on said respective audio channel with a corresponding impulse response; for each audio channel other than said respective audio channel, performing cross-channel convolution operation respectively with a corresponding cross-channel impulse response; performing combination of the results of the respective convolution operations; and outputting the combination (or summation) result as said output audio channel; wherein, in said performing said cross-channel convolution operation, wherein at least one convolution operation is performed corresponding to a shorter length of impulse response than at least one other convolution operation, preferably, said cross-channel convolution operation being performed only for an initial part of said cross-channel impulse response, said initial part being defined by a definition parameter.

公开(公告)号：US07772481B2

公开(公告)日：2010-08-10

申请号：US11196815

申请日：2005-08-03

申请人： Roberto M. Aimi

发明人： Roberto M. Aimi

IPC分类号： G10H3/00

CPC分类号： G10H7/002 ,  G10H3/146 ,  G10H2220/525 ,  G10H2230/275 ,  G10H2250/145

摘要： Methods and apparatus for simulating the sound of an acoustic percussion instrument. A first stored waveform signal representative of the impulse response of an acoustic percussion instrument is convolved with a waveform produced by a sensor circuit attached to a physical playing surface. Undesirable response characteristics of the playing surface and sensor circuit may be filtered out by deconvolving sensor output, or the stored waveform representing the acoustic instrument, with the impulse response of the combination of the playing surface and the sensor circuit.

摘要翻译： 用于模拟声学打击乐器声音的方法和装置。 表示声学打击乐器的脉冲响应的第一存储的波形信号与由附接到物理播放表面的传感器电路产生的波形卷积。 播放面和传感器电路的不期望的响应特性可以通过解卷积传感器输出或代表声学乐器的存储波形与播放表面和传感器电路的组合的脉冲响应一起滤除。

公开(公告)号：US20090010460A1

公开(公告)日：2009-01-08

申请号：US11713167

申请日：2007-03-01

申请人： Steffan Diedrichsen

发明人： Steffan Diedrichsen

IPC分类号： H04R5/02

CPC分类号： H04S3/008 ,  G10H2210/281 ,  G10H2210/301 ,  G10H2250/111 ,  G10H2250/145 ,  G10K15/08 ,  G10L19/008

摘要： The present invention provides a method of generating, on a data processing system, a multi-channel audio convolution reverb, said method comprising providing a plurality of impulse responses corresponding to a desired room to be simulated; receiving, in input, multi-channel audio sample data; performing, for each respective audio channel, same channel convolution operation on said respective audio channel with a corresponding impulse response; for each audio channel other than said respective audio channel, performing cross-channel convolution operation respectively with a corresponding cross-channel impulse response; performing combination of the results of the respective convolution operations; and outputting the combination (or summation) result as said output audio channel; wherein, in said performing said cross-channel convolution operation, wherein at least one convolution operation is performed corresponding to a shorter length of impulse response than at least one other convolution operation, preferably, said cross-channel convolution operation being performed only for an initial part of said cross-channel impulse response, said initial part being defined by a definition parameter.

摘要翻译： 本发明提供了一种在数据处理系统上产生多声道音频卷积混响的方法，所述方法包括提供对应于要被仿真的所需房间的多个脉冲响应; 在输入中接收多声道音频采样数据; 对于每个相应的音频通道，对所述相应的音频通道执行与相应的脉冲响应相同的信道卷积运算; 对于除了所述各个音频信道之外的每个音频通道，分别以对应的交叉通道脉冲响应执行交叉通道卷积运算; 执行各卷积运算结果的组合; 并输出组合（或求和）结果作为所述输出音频通道; 其中，在所述执行所述交叉通道卷积运算中，其中对应于比至少一个其它卷积运算的较短的脉冲响应长度执行至少一个卷积运算，优选地，所述交叉通道卷积运算仅针对初始 所述交叉通道脉冲响应的一部分，所述初始部分由定义参数定义。

公开(公告)号：US20030202667A1

公开(公告)日：2003-10-30

申请号：US10424045

申请日：2003-04-25

申请人： Yamaha Corporation

发明人： Satoshi Sekine

IPC分类号： H03G003/00 ,  G10H001/02 ,  G10H007/00

CPC分类号： G10H1/0091 ,  G10H2210/281 ,  G10H2210/301 ,  G10H2250/145

摘要： An apparatus estimates an impulse response for use in reproduction of a sound in a desired acoustic space. In the apparatus, a first acquisition section acquires space information indicating a spatial shape of the acoustic space and an acoustic reflectivity of a boundary surface enclosing the acoustic space. A second acquisition section acquires point information indicating positions of a sound generation point and a sound reception point set in the acoustic space. An estimation section estimates a set of acoustic ray paths of the sound traveling from the sound generation point to the sound reception point based on the acquired space information and the point information. A third acquisition section acquires directivity information indicating an acoustic directivity of the sound generation point and the sound reception point. A weighting section estimates an acoustic intensity of each acoustic ray path, and weights each acoustic intensity by the acquired directivity information. A determination section determines the impulse response based on directions of the respective acoustic ray paths toward the sound reception point and the weighed acoustic intensities of the respective acoustic ray paths.

摘要翻译： 一种装置估计用于在所需声学空间中再现声音的脉冲响应。 在该装置中，第一获取部获取表示声空间的空间形状的空间信息和包围声学空间的边界面的声反射率。 第二获取部分获取指示在声学空间中设置的声音产生点和声音接收点的位置的点信息。 估计部基于所获取的空间信息和点信息来估计从声音生成点传播到声音接收点的声音的一组声线路径。 第三获取部分获取指示声音产生点和声音接收点的声方向性的方向性信息。 加权部分估计每个声射线路径的声强度，并且通过获取的方向性信息对每个声强进行加权。 确定部分基于各个声光路径朝向声音接收点的方向和各个声线路径的称重声强来确定脉冲响应。

公开(公告)号：US20030033338A1

公开(公告)日：2003-02-13

申请号：US10145782

申请日：2002-05-16

发明人： Ulf Lindgren ,  Alberto Jimenez Feltstrom ,  Thomas Jacobsson

IPC分类号： G06F017/10

CPC分类号： G10H7/045 ,  G10H2210/225 ,  G10H2230/015 ,  G10H2240/056 ,  G10H2240/251 ,  G10H2250/145 ,  G10H2250/281 ,  G10H2250/285 ,  G10H2250/291 ,  G10H2250/545 ,  G10H2250/631

摘要： A method and apparatus are provided for changing the pitch of a tabulated waveform in wavetable based synthesizers. Harmonics that normally would be aliased before a transposition process are removed by a discrete time low pass filter at the same time that the tabulated waveform is reconstruction and resampling.

摘要翻译： 提供了一种用于改变基于波表的合成器中的列表波形的音调的方法和装置。 在转置处理之前通常将被别名的谐波通过离散时间低通滤波器去除，同时制表波形是重建和重采样。

公开(公告)号：US5814750A

公开(公告)日：1998-09-29

申请号：US555537

申请日：1995-11-09

申请人： Avery L. Wang ,  Brooks S. Read

发明人： Avery L. Wang ,  Brooks S. Read

IPC分类号： G10H1/12 ,  G10H7/08

CPC分类号： G10H1/125 ,  G10H7/08 ,  G10H2250/111 ,  G10H2250/145 ,  G10H2250/285 ,  G10H2250/291 ,  G10H2250/545 ,  G10H2250/631 ,  Y10S84/09

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.

摘要翻译： 重采样的方法包括使用低通滤波器的脉冲响应函数来卷积给定的采样集合。 在该方法中，在从接近脉冲响应的分段多项式进行重采样时计算卷积计算所需的脉冲响应的值。 在一个实施例中，该方法被应用于从存储的波形提供各种音调的乐音。

公开(公告)号：US4643067A

公开(公告)日：1987-02-17

申请号：US631131

申请日：1984-07-16

申请人： Ralph Deutsch

发明人： Ralph Deutsch

IPC分类号： G10H7/08 ,  G10H1/06 ,  G10H7/00 ,  G10H7/10 ,  G10H1/00

CPC分类号： G10H1/06 ,  G10H7/105 ,  G10H2250/145

摘要： A keyboard operated electronic musical instrument is disclosed which has a number of tone generators each of which is assigned to an actuated keyswitch. The generated musical waveshapes are transformed to produce tones having a time variant spectra by processing the waveshapes with a time variant masking function.

摘要翻译： 公开了一种键盘操作的电子乐器，其具有多个音调发生器，每个音调发生器分配给致动的钥匙开关。 所产生的音乐波形被变换以产生具有时变频谱的音调，通过处理具有时变掩蔽功能的波形形状。