公开(公告)号：US07812243B2

公开(公告)日：2010-10-12

申请号：US11321395

申请日：2005-12-29

申请人： Peter J. Celi ,  Michel A. Doidic ,  Marcus Ryle

发明人： Peter J. Celi ,  Michel A. Doidic ,  Marcus Ryle

IPC分类号： G10H3/00

CPC分类号： G10H3/186 ,  G10H1/053 ,  G10H1/125 ,  G10H3/185 ,  G10H3/188 ,  G10H2210/281 ,  G10H2230/111 ,  G10H2230/115 ,  G10H2230/121 ,  G10H2230/151 ,  G10H2250/115

摘要： Disclosed is a stringed instrument with embedded DSP modeling capabilities to model an acoustic stringed instrument. The stringed instrument has a body and a plurality of strings and each of the plurality of strings is respectively coupled to a pickup to detect a vibration signal for each string. An A/D converter converts the detected vibration signal of a string into a digital string vibration signal. A DSP is located within the body of the stringed instrument to process the digital string vibration signal and to implement an acoustic modeling system to process the digital string vibration signal in order to emulate a corresponding string tone of one of a plurality of selectable acoustic stringed instruments. Acoustic modeling includes acoustic string and body modeling, microphone placement modeling, and pick-sound modeling. The emulated acoustic digital tone signal is then converted to analog form for output to an amplification device.

摘要翻译： 公开了一种具有嵌入式DSP建模功能的弦乐器，用于对声学弦乐器进行建模。 弦乐器具有主体和多个琴弦，并且多个弦中的每一个分别耦合到拾音器以检测每个弦的振动信号。 A / D转换器将检测到的串的振动信号转换成数字串振动信号。 DSP位于弦乐器的主体内以处理数字弦振动信号并实现声学建模系统以处理数字弦振动信号，以便模拟多个可选择的弦乐器之一的对应琴弦 。 声学建模包括声学字符串和身体建模，麦克风位置建模和拾音建模。 然后将仿真的声学数字音调信号转换成模拟形式以输出到放大装置。

公开(公告)号：US06787690B1

公开(公告)日：2004-09-07

申请号：US10197363

申请日：2002-07-16

申请人： Peter J. Celi ,  Michel A. Doidic ,  David W. Fruehling ,  Marcus Ryle

发明人： Peter J. Celi ,  Michel A. Doidic ,  David W. Fruehling ,  Marcus Ryle

IPC分类号： G10H300

CPC分类号： G10H3/188 ,  G10H1/125 ,  G10H2230/111 ,  G10H2230/115 ,  G10H2230/121 ,  G10H2230/151 ,  G10H2250/115

摘要： Disclosed is a stringed instrument with embedded digital signal processing (DSP) modeling capabilities. The stringed instrument has a body and a plurality of strings and each of the plurality of strings is respectively coupled to a pickup of a polyphonic pickup. The polyphonic pickup is used to detect a vibration signal for each string. An A/D converter converts the detected vibration signal of a string into a digital string vibration signal. Further, a digital signal processor is located within the body of the stringed instrument to process the digital string vibration signal. Particularly, the digital signal processor is used to process the digital string vibration signal such that the corresponding string tone of one of a plurality of selectable stringed instruments may be emulated. The emulated digital tone signal is then converted to analog form to create an emulated analog tone signal for output to an amplification device.

摘要翻译： 公开了一种具有嵌入式数字信号处理（DSP）建模功能的弦乐器。 弦乐器具有主体和多个弦，并且多个弦中的每一个分别耦合到复音拾音器的拾音器。 复音拾音器用于检测每个弦的振动信号。 A / D转换器将检测到的串的振动信号转换成数字串振动信号。 此外，数字信号处理器位于弦乐器的主体内以处理数字弦振动信号。 特别地，数字信号处理器用于处理数字串振动信号，使得可以仿真多个可选弦乐器之一的对应串音。 仿真的数字乐音信号然后被转换成模拟形式以产生用于输出到放大装置的仿真模拟乐音信号。

公开(公告)号：US06696633B2

公开(公告)日：2004-02-24

申请号：US10330731

申请日：2002-12-27

申请人： Satoshi Miyagishima ,  Shinichi Sawara ,  Akira Miki ,  Fukushi Kawakami ,  Yasutake Suzuki ,  Takeo Shibukawa

发明人： Satoshi Miyagishima ,  Shinichi Sawara ,  Akira Miki ,  Fukushi Kawakami ,  Yasutake Suzuki ,  Takeo Shibukawa

IPC分类号： G10H102

CPC分类号： G10H1/0091 ,  G10H2210/281 ,  G10H2210/295 ,  G10H2250/115 ,  G10H2250/235

摘要： As an electronic tone is generated in response to performing operation, the electronic tone is picked up by microphones corresponding to left and right channels, and picked-up sound signals thus generated by the microphones are then subjected to signal processing, such as reverberation impartment utilizing acoustic conditions of the interior of a room. Picked-up sound signals having undergone such signal processing are audibly reproduced via rear speakers. Then, once an automatic adjustment instruction is given from a user, measuring tones are reproduced stereophonically, and contents of the signal processing of the individual channels are adjusted on the basis of measured results of picked-up sound signals generated by the microphones picking up the reproduced measuring tones.

摘要翻译： 由于响应于执行操作产生电子音调，所以用与左右声道相对应的麦克风拾取电子音调，然后由麦克风产生的拾取声音信号进行信号处理，例如混响室 房间内部的声学条件。 经过这种信号处理的拾取声音信号通过后置扬声器可听地再现。 然后，一旦从用户给出自动调整指令，则立体声地再现测量音，并且基于由拾取的麦克风产生的拾取声音信号的测量结果来调整各个声道的信号处理的内容 再现测量音。

公开(公告)号：US06664460B1

公开(公告)日：2003-12-16

申请号：US10037510

申请日：2002-01-04

申请人： James D. Pennock ,  Robin M. Urry ,  John D. Hanson

发明人： James D. Pennock ,  Robin M. Urry ,  John D. Hanson

IPC分类号： G10H102

CPC分类号： G10H1/125 ,  G10H1/0091 ,  G10H1/02 ,  G10H2250/095 ,  G10H2250/115

摘要： This invention provides a system for customizing musical instrument signal processing enabling users to produce different tonal characteristics in created musical pieces. In order to create such tonal characteristics, a new mathematical model of tonal characteristics may be digitally created based on two or more initial mathematical models of tonal characteristics. After simulating a first and second initial mathematical models of tonal characteristics, the new mathematical model is created by interpolating one or more coefficients of the first and second initial mathematical models. The new mathematical model may also adjust a control parameter where the control parameter may exist between two values. When the control parameter is the first value, the new mathematical model is the first initial mathematical model. When the control parameter is the second value, the new mathematical model may be the second initial mathematical model. When the control parameter is located at a point between the first and second values, the new mathematical model may represent a convergence between the first and second models.

摘要翻译： 本发明提供了一种用于定制乐器信号处理的系统，使得用户能够在创作的音乐作品中产生不同的音调特征。 为了产生这种音调特征，可以基于两个或更多个音调特征的初始数学模型数字地创建新的音调特征数学模型。 在模拟音调特性的第一和第二初始数学模型之后，通过内插第一和第二初始数学模型的一个或多个系数来创建新的数学模型。 新的数学模型还可以调整控制参数，其中控制参数可以存在于两个值之间。 当控制参数是第一个值时，新的数学模型是第一个初始数学模型。 当控制参数是第二个值时，新的数学模型可能是第二个初始数学模型。 当控制参数位于第一和第二值之间的点时，新的数学模型可以表示第一和第二模型之间的收敛。

公开(公告)号：US5771294A

公开(公告)日：1998-06-23

申请号：US725544

申请日：1996-10-03

申请人： Toshihiro Inoue ,  Hiroyuki Torimura

发明人： Toshihiro Inoue ,  Hiroyuki Torimura

IPC分类号： H04S1/00 ,  G01V1/00 ,  G10H1/00 ,  G10H1/12 ,  H03H21/00 ,  H04S3/00 ,  H04S7/00 ,  H04S5/00

CPC分类号： G10H1/125 ,  G01V1/001 ,  G10H1/0091 ,  H04S3/00 ,  G10H2210/301 ,  G10H2250/115

摘要： An acoustic image localization apparatus localizes an acoustic image of a plurality of tone color species having various timbres to a source point with respect to a listening point within a sound field. Desired ones of separate source points are specified, which are provisionally set throughout the sound field. Each of the tone color species is assigned to one or more of the specified source points such that each specified source point is assigned with a group of desired tone color species. Acoustic images of the respective groups are localized to corresponding ones of the specified source points. Further, one of separate listening points is designated, which is provisionally set relatively to the source points so that the acoustic images are localized with respect to the designated listening point.

公开(公告)号：US5648778A

公开(公告)日：1997-07-15

申请号：US683754

申请日：1996-07-17

申请人： Alfredo R. Linz ,  Carlin Dru Cabler ,  Glen W. Brown ,  Martin P. Soques

发明人： Alfredo R. Linz ,  Carlin Dru Cabler ,  Glen W. Brown ,  Martin P. Soques

IPC分类号： F02B75/02 ,  G06F3/16 ,  G10H1/00 ,  G10H1/12 ,  G10H7/00 ,  H03K23/68 ,  H03M7/32 ,  H03M1/00

CPC分类号： H03K23/68 ,  G06F3/162 ,  G10H1/0058 ,  G10H1/0066 ,  G10H1/125 ,  G10H7/002 ,  F02B2075/027 ,  G10H2230/035 ,  G10H2240/311 ,  G10H2250/115 ,  G10H2250/191 ,  G10H2250/545 ,  G10H2250/571 ,  G10H2250/611

摘要： A stereo audio CODEC, including means for performing D/A and A/D conversions, means for reducing digitally induced noise during attenuation/gain changes, data format conversion means, analog and digital filtering means, analog mixing means, on-chip 16-sample, 32-bit wide record and playback FIFOs, serial interface with external serial DSP, large local memory for supplementing on-chip record and playback FIFOs, control registers, register data bus and synthesizer DAC.

摘要翻译： 立体声音频CODEC，包括用于执行D / A和A / D转换的装置，用于在衰减/增益变化期间减少数字感应噪声的装置，数据格式转换装置，模拟和数字滤波装置，模拟混合装置，片上16- 采样，32位宽记录和播放FIFO，与外部串行DSP的串行接口，用于补充片上记录和播放FIFO，控制寄存器，寄存器数据总线和合成器DAC的大型本地存储器。

公开(公告)号：US5534844A

公开(公告)日：1996-07-09

申请号：US415266

申请日：1995-04-03

申请人： David Norris

发明人： David Norris

IPC分类号： G06F7/02 ,  G10H1/00 ,  G10H1/12 ,  G10H7/00 ,  H03K23/68 ,  H03K23/70

CPC分类号： G06F7/026 ,  H03K23/68 ,  H03K23/70 ,  G10H2220/315 ,  G10H2230/035 ,  G10H2240/311 ,  G10H2250/115 ,  G10H2250/191 ,  G10H2250/545 ,  G10H2250/571 ,  G10H2250/605 ,  G10H2250/611

摘要： A static-type comparator, which compares the magnitude of a first binary number with a second binary number and determines if the first binary number is equal to, greater than, or less than the second binary number, is described. The comparator comprises a carry chain of comparison cells. Each comparison cell in the carry chain compares the magnitude of a different bit position of the first number with a corresponding bit position in the second number. The comparison cells input a first voltage signal, representing a binary value of a bit position of the first number, and a second voltage signal, representing a binary value of a corresponding bit position of the second number. A voltage signal from a voltage source Vcc is input into the carry chain and propagates through the chain until a comparison cell detects that there is a difference in magnitudes for a particular bit position. If a difference in magnitudes is detected, the comparison cell will output a voltage signal indicating whether the first number is greater than or less than the second number. If the numbers are equal, voltage signal Vcc propagates through the entire carry chain and is output. In the preferred embodiment, the comparator includes decode circuitry for decoding the carry chain's outputs. The comparator preferably also inputs a timing signal to time the comparator's operations.

摘要翻译： 静态型比较器，其将第一二进制数的大小与第二二进制数进行比较，并确定第一二进制数是否等于，大于或小于第二二进制数。 比较器包括比较单元的进位链。 进位链中的每个比较单元将第一数字的不同位位置的大小与第二数目中的对应位位置进行比较。 比较单元输入表示第一数字的位位置的二进制值的第一电压信号和表示第二数字的对应位位置的二进制值的第二电压信号。 来自电压源Vcc的电压信号被输入到进位链中并且通过链传播，直到比较单元检测到特定位位置的幅度差异为止。 如果检测到大小差异，则比较单元将输出指示第一数字是大于还是小于第二数字的电压信号。 如果数字相等，则电压信号Vcc通过整个进位链传播并被输出。 在优选实施例中，比较器包括用于对进位链的输出进行解码的解码电路。 比较器优选地还输入定时信号以对比较器的操作进行计时。

公开(公告)号：US5247130A

公开(公告)日：1993-09-21

申请号：US734279

申请日：1991-07-22

申请人： Hideo Suzuki ,  Kazuhisa Okamura

发明人： Hideo Suzuki ,  Kazuhisa Okamura

IPC分类号： G10H1/12 ,  H03H17/00 ,  H03H17/02

CPC分类号： G10H1/125 ,  G10H2250/095 ,  G10H2250/115 ,  G10H2250/545 ,  Y10S84/09

摘要： A series of delay circuits are provided which receive tone signal sample data of plural channels supplied on a time shared basis, hold plural tone signal sample data of each channel and hold these plural sample data while sequentially delaying them. The series of delay circuits are connected in an endless manner to form a circulating loop and plural data selection circuits are provided at predetermined delay stages. Each data selection circuit performs a selection control as to whether new tone signal sample data should be loaded in the delay circuit loop or data in the delay circuit should be circulated. Tone signal sample data which are provided sequentially from the delay circuit loop after delay are sequentially operated with filter coefficients of plural orders and results of the operation are accumulated to obtain a filter operation output. This construction enables tone signal sample data of a necessary channel to be loaded in the delay circuit loop at a point of a necessary data selection circuit in a time period which is shorter than one cycle of the entire delay circuit loop. Therefore, the time division period of supplied tone signal sample data of each channel can be made different from the time division period of each channel in the delay circuit loop.

摘要翻译： 提供了一系列延迟电路，其接收在时间共享的基础上提供的多个频道的音调信号采样数据，保持每个通道的多个音调信号采样数据并保持这些多个采样数据，同时依次延迟它们。 一系列延迟电路以无限方式连接形成循环回路，并以预定的延迟阶段提供多个数据选择电路。 每个数据选择电路执行关于是否应该在延迟电路环路中加载新的音调信号采样数据或延迟电路中的数据应该循环的选择控制。 由延迟后的延迟电路循环依次提供的音调信号采样数据按多个阶的滤波器系数顺序运行，累加运算结果以获得滤波器运算输出。 这种结构使得必要通道的音调信号采样数据能够在短于整个延迟电路环路的一个周期的时间段内在必要的数据选择电路的一点加载在延迟电路回路中。 因此，可以使每个信道的提供的音调信号采样数据的时分周期与延迟电路回路中每个通道的时分周期不同。

公开(公告)号：US4338581A

公开(公告)日：1982-07-06

申请号：US146966

申请日：1980-05-05

申请人： Nelson H. Morgan

发明人： Nelson H. Morgan

IPC分类号： G10H1/00 ,  G10H1/02 ,  H03H15/02 ,  H03H17/02

CPC分类号： H03H15/02 ,  G10H1/0091 ,  G10H1/02 ,  H03H17/0291 ,  G10H2250/105 ,  G10H2250/115 ,  G10H2250/531 ,  Y10S84/26

摘要： Room acoustics simulation is achieved by digitally defining delay and weighting parameters and generating sampled analog signal responses. In the early portion of a response term an accurate impulse response is provided and subsequently high echo density is simulated using sampled analog signal Schroeder sections. A sampled data analog signal device is used which employs charge transfer devices as programmable delay media, multiplying digital to analog converters to generate weighting of the analog signal and a general purpose microprocessor as a parameter calculator. The early impulse response characteristic is accomplished by providing a time domain finite impulse response (FIR) section in a transversal filter arrangement which directly feeds to an ultimate output and to parallel comb filter sections which in turn input to at least one all pass section of the so-called Schroeder configuration. The charge transfer devices are employed as programmable delays to propagate the analog signals. All weighting coefficient products are generated by multiplying digital-to-analog converters.

摘要翻译： 通过数字定义延迟和加权参数并产生采样的模拟信号响应来实现房间声学仿真。 在响应项的早期部分，提供精确的脉冲响应，随后使用采样的模拟信号施罗德分段来模拟高回波密度。 使用采样的数据模拟信号装置，其采用电荷传输装置作为可编程延迟介质，乘以数模转换器以产生模拟信号的加权和通用微处理器作为参数计算器。 早期脉冲响应特性通过在横向滤波器装置中提供时域有限脉冲响应（FIR）部分来实现，该横向滤波器装置直接馈送到最终输出端，并且并行梳状滤波器部分，该梳状滤波器部分依次输入到至少一个全部通过部分 所谓的施罗德配置。 电荷转移装置被用作可编程延迟来传播模拟信号。 所有加权系数乘积都是通过乘数字模拟转换器产生的。

公开(公告)号：US11694663B2

公开(公告)日：2023-07-04

申请号：US17191286

申请日：2021-03-03

申请人： CASIO COMPUTER CO., LTD.

发明人： Masuo Yokota

IPC分类号： G10H1/00 ,  G10K15/08

CPC分类号： G10H1/0091 ,  G10K15/08 ,  G10H2250/105 ,  G10H2250/115

摘要： An effect addition device includes at least one processor that executes a time domain convolution process of convolving a first time domain data part of impulse response of sound effects with a time domain data on an original sound, a frequency domain convolution process of convoluting a second time domain data part of the impulse response data with the time domain data on the original sound, a convolution extension process of extending a convolved state(s) of an output signal(s) resulting from the time domain convolution process and/or the frequency domain convolution process by arithmetic processing which corresponds to an all-pass filter and/or arithmetic processing which corresponds to a comb filter, and a synthesized sound effect addition process of adding a sound effect which is synthesized by execution of the time domain convolution process, the frequency domain convolution process and the convolution extension process to the original sound.