公开(公告)号：US20100008519A1

公开(公告)日：2010-01-14

申请号：US12453173

申请日：2009-04-30

申请人： Shoji Hayakawa ,  Naoshi Matsuo

发明人： Shoji Hayakawa ,  Naoshi Matsuo

IPC分类号： H04B15/00

CPC分类号： G10L21/0208 ,  G10L21/0216 ,  G10L21/0224 ,  G10L2021/02165 ,  G10L2021/02166

摘要： There is provided a noise suppressing device, for suppressing a noise component contained in a sound, including: at least two sound receiving parts receiving sounds from a plurality of directions containing a sound from a direction of a given sound source and converting the sounds to digital sound signals in a time domain, respectively; an estimating part acquiring both direction information on a direction of the given sound source and distance information on a distance from the given sound source based upon the digital sound signals converted by the sound receiving parts, and estimating a component value of a noise component contained in the signal by use of the direction information and the distance information; and a controlling part acquiring a control value of a suppression amount for controlling a range of a direction of the digital sound signals.

摘要翻译： 提供了一种用于抑制包含在声音中的噪声成分的噪声抑制装置，包括：至少两个声音接收部件，其从包含来自给定声源的方向的声音的多个方向接收声音，并将声音转换成数字 分别在时域中的声音信号; 估计部分基于由声音接收部分转换的数字声音信号，获取关于给定声源的方向的两个方向信息和距给定声源的距离的距离信息，以及估计包含在 信号通过使用方向信息和距离信息; 以及控制部，其获取用于控制数字声音信号的方向的范围的抑制量的控制值。

公开(公告)号：US07324466B2

公开(公告)日：2008-01-29

申请号：US10648319

申请日：2003-08-27

申请人： Naoshi Matsuo

发明人： Naoshi Matsuo

IPC分类号： H04B3/20

CPC分类号： H04M9/082

摘要： An echo canceling system and an echo canceling method are provided, which can deal with the case where there are a plurality of echo paths and respond to the variation in echo arrival times. An echo canceling method to be applied to a full-duplex communication system includes detecting a respective echo arrival time of one or plural echo paths based on a reference signal and an echo signal, calculating as many pseudo-echo signals as the detected arrival times, overlapping the calculated pseudo-echo signals to obtain an overall pseudo-echo signal, and subtracting the overall pseudo-echo signal from the echo signal. A FFT processing is performed with respect to the reference signal and the echo signal, and a similar canceling processing is carried out using an amplitude spectrum alone.

摘要翻译： 提供了一种回波消除系统和回波消除方法，其可以处理存在多个回波路径并响应于回波到达时间的变化的情况。 要应用于全双工通信系统的回波消除方法包括基于参考信号和回波信号来检测一个或多个回波路径的相应回波到达时间，计算与检测到的到达时间一样多的伪回波信号， 与所计算的伪回波信号重叠以获得整体伪回波信号，并从回波信号中减去总体伪回波信号。 对参考信号和回波信号执行FFT处理，并且仅使用幅度谱执行类似的消除处理。

公开(公告)号：US06553121B1

公开(公告)日：2003-04-22

申请号：US09449570

申请日：1999-11-29

申请人： Naoshi Matsuo ,  Kaori Suzuki

发明人： Naoshi Matsuo ,  Kaori Suzuki

IPC分类号： H04R500

CPC分类号： H04S1/005 ,  H04S1/002 ,  H04S1/007 ,  H04S7/302 ,  H04S2420/01

摘要： To provide a three-dimensional acoustic effect to a listener in a reproduction field, via a headphone in particular, a three-dimensional acoustic apparatus is formed by a linear synthesis filter having filter coefficients that are the linear predictive coefficients obtained by performing a linear predictive analysis on an impulse response which represents the acoustic characteristics to be added to the original signal to achieve this effect. By passing the signal through this acoustic characteristics adding filter, the desired acoustic characteristics are added to the original signal, and by dividing the power spectrum of the impulse response of these acoustic characteristics into critical bandwidths and performing this linear predictive analysis based on impulse signal determined based from power spectrum signals representing the signal sound of each of these critical bandwidths, the filter coefficients of the linear synthesis filter are determined.

摘要翻译： 为了在再现场中通过耳机提供三维声学效果，特别是通过具有滤波器系数的线性合成滤波器形成三维声学装置，所述滤波器系数是通过执行线性预测 分析脉冲响应，其表示要添加到原始信号的声学特性以实现该效果。 通过使信号通过该声学特性相加滤波器，将期望的声学特性加到原始信号上，并且通过将这些声学特性的脉冲响应的功率谱除以关键带宽并且基于确定的脉冲信号执行该线性预测分析 基于表示这些关键带宽中的每一个的信号声音的功率谱信号，确定线性合成滤波器的滤波器系数。

公开(公告)号：US06269166B1

公开(公告)日：2001-07-31

申请号：US09330017

申请日：1999-06-11

申请人： Naoshi Matsuo ,  Kaori Suzuki

发明人： Naoshi Matsuo ,  Kaori Suzuki

IPC分类号： H04R502

CPC分类号： H04S1/005 ,  H04S1/002 ,  H04S1/007 ,  H04S7/302 ,  H04S2420/01

摘要： To provide a three-dimensional acoustic effect to a listener in a reproduction field, via a headphone in particular, a three-dimensional acoustic apparatus is formed by a linear synthesis filter having filter coefficients that are the linear predictive coefficients obtained by performing a linear predictive analysis on an impulse response which represents the acoustic characteristics to be added to the original signal to achieve this effect. By passing the signal through this acoustic characteristics adding filter, the desired acoustic characteristics are added to the original signal, and by dividing the power spectrum of the impulse response of these acoustic characteristics into critical bandwidths and performing this linear predictive analysis based on impulse signal determined based from power spectrum signals representing the signal sound of each of these critical bandwidths, the filter coefficients of the linear synthesis filter are determined.

摘要翻译： 为了在再现场中通过耳机提供三维声学效果，特别是通过具有滤波器系数的线性合成滤波器形成三维声学装置，所述滤波器系数是通过执行线性预测 分析脉冲响应，其表示要添加到原始信号的声学特性以实现该效果。 通过使信号通过该声学特性相加滤波器，将期望的声学特性加到原始信号上，并且通过将这些声学特性的脉冲响应的功率谱除以关键带宽并且基于确定的脉冲信号执行该线性预测分析 基于表示这些关键带宽中的每一个的信号声音的功率谱信号，确定线性合成滤波器的滤波器系数。

公开(公告)号：US6023512A

公开(公告)日：2000-02-08

申请号：US697247

申请日：1996-08-21

申请人： Naoshi Matsuo ,  Kaori Suzuki

发明人： Naoshi Matsuo ,  Kaori Suzuki

IPC分类号： H04S1/00 ,  H04S7/00 ,  H04R5/00

CPC分类号： H04S1/005 ,  H04S1/002 ,  H04S1/007 ,  H04S2420/01 ,  H04S7/302

摘要： To provide a three-dimensional acoustic effect to a listener in a reproduction field, via a headphone in particular, a three-dimensional acoustic apparatus is formed by a linear synthesis filter having filter coefficients that are the linear predictive coefficients obtained by performing a linear predictive analysis on an impulse response which represents the acoustic characteristics to be added to the original signal to achieve this effect. By passing the signal through this acoustic characteristics adding filter, the desired acoustic characteristics are added to the original signal, and by dividing the power spectrum of the impulse response of these acoustic characteristics into critical bandwidths and performing this linear predictive analysis based on impulse signal determined based from power spectrum signals representing the signal sound of each of these critical bandwidths, the filter coefficients of the linear synthesis filter are determined.

公开(公告)号：US09099088B2

公开(公告)日：2015-08-04

申请号：US13064871

申请日：2011-04-21

申请人： Nobuyuki Washio ,  Shouji Harada ,  Akira Kamano ,  Naoshi Matsuo

发明人： Nobuyuki Washio ,  Shouji Harada ,  Akira Kamano ,  Naoshi Matsuo

IPC分类号： G10L15/00 ,  G10L17/26 ,  G10L25/48 ,  G10L21/00 ,  G10L13/00 ,  G08B23/00

CPC分类号： G10L17/26 ,  G10L25/48

摘要： An utterance state detection device includes an user voice stream data input unit that gets user voice stream data of an user, a frequency element extraction unit that extracts high frequency elements by frequency-analyzing the user voice stream data, a fluctuation degree calculation unit that calculates a fluctuation degree of the high frequency elements thus extracted every unit time, a statistic calculation unit that calculates a statistic every certain interval based on a plurality of the fluctuation degrees in a certain period of time, and an utterance state detection unit that detects an utterance state of a specified user based on the statistic obtained from user voice stream data of the specified user.

摘要翻译： 发声状态检测装置包括：用户语音流数据输入单元，其获取用户的用户语音流数据;频率元素提取单元，其通过对用户语音流数据进行频率分析来提取高频元素;波动度计算单元， 每单位时间提取的高频元件的波动程度，统计量计算单元，其基于一定时间段内的多个波动度计算每一定间隔的统计量;以及发声状态检测单元，其检测发音 基于从指定用户的用户语音流数据获得的统计量来指定用户的状态。

公开(公告)号：US09020820B2

公开(公告)日：2015-04-28

申请号：US13446019

申请日：2012-04-13

申请人： Shoji Hayakawa ,  Naoshi Matsuo

发明人： Shoji Hayakawa ,  Naoshi Matsuo

IPC分类号： G10L15/14 ,  G10L17/26 ,  G10L15/08 ,  G10L25/24 ,  G10L25/63 ,  G10L25/90

CPC分类号： G10L17/26 ,  G10L15/08 ,  G10L25/24 ,  G10L25/63 ,  G10L25/90

摘要： A state detecting apparatus includes: a processor to execute acquiring utterance data related to uttered speech, computing a plurality of statistical quantities for feature parameters regarding features of the utterance data, creating, on the basis of the plurality of statistical quantities regarding the utterance data and another plurality of statistical quantities regarding reference utterance data based on other uttered speech, pseudo-utterance data having at least one statistical quantity equal to a statistical quantity in the other plurality of statistical quantities, computing a plurality of statistical quantities for synthetic utterance data synthesized on the basis of the pseudo-utterance data and the utterance data, and determining, on the basis of a comparison between statistical quantities of the synthetic utterance data and statistical quantities of the reference utterance data, whether the speaker who produced the uttered speech is in a first state or a second state; and a memory.

摘要翻译： 一种状态检测装置，包括：处理器，用于执行获取与发话语音有关的话语数据，计算关于话语数据特征的特征参数的多个统计量，基于关于话语数据的多个统计量，以及 关于基于其他语音的参考话语数据的另外多个统计量，具有至少一个等于其他多个统计量中的统计量的统计量的伪话语数据，计算合成语音数据的多个统计量 伪话音数据和发声数据的基础，并且基于合成发音数据的统计量与参考语音数据的统计量之间的比较，确定产生发言语音的演讲者是否处于 第一状态或第二状态; 和记忆。

公开(公告)号：US20120275620A1

公开(公告)日：2012-11-01

申请号：US13425717

申请日：2012-03-21

申请人： Naoshi MATSUO

发明人： Naoshi MATSUO

IPC分类号： H04R3/00

CPC分类号： H04R3/005 ,  H04R1/406 ,  H04R2410/07

摘要： A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation.

摘要翻译： 麦克风阵列装置包括：获取单元，被配置为以预定的时间间隔从从多个麦克风中的每一个输入的声音信号获取样本; 操作单元，被配置为基于从所述多个麦克风输入的每个声音信号，基于由多个样本所拥有的声音信号的音量来计算值; 相关系数计算器，被配置为基于针对各声音信号计算的值来计算声音信号之间的相关系数; 以及增益计算器，被配置为基于所述相关系数来计算从所述多个麦克风输入的声音信号的缩小增益。

公开(公告)号：US07941315B2

公开(公告)日：2011-05-10

申请号：US11385653

申请日：2006-03-22

申请人： Naoshi Matsuo

发明人： Naoshi Matsuo

IPC分类号： G10L21/02

CPC分类号： G10L21/0208

摘要： Accepting the speech having the noise superimposed thereon and converting it into a signal on a time axis of the speech, an amplitude component of a speech for each predetermined frequency band of the converted signal on the frequency axis is calculated. Calculating a noise reduction coefficient, the noise component is reduced by multiplying the signal on the frequency axis of the original signal by the calculated noise reduction coefficient. By estimating the target value of the remaining noise for each frequency band, a signal on a frequency axis in which a signal corresponding to a frequency band of which target value estimated by the noise target value is larger than the value of the amplitude component of the signal on the frequency axis of which noise component is reduced is corrected to a signal corresponding to the target value is restored, into a signal on a time axis.

摘要翻译： 接收具有叠加在其上的噪声并将其转换成语音的时间轴上的信号的语音，计算频率轴上的转换信号的每个预定频带的语音的振幅分量。 通过计算噪声降低系数，通过将原始信号的频率轴上的信号乘以所计算的噪声降低系数来降低噪声分量。 通过估计每个频带的剩余噪声的目标值，频率轴上的信号，其中与由噪声目标值估计的目标值的频带相对应的信号大于该噪声目标值的振幅分量的值 噪声分量减小的频率轴上的信号被校正为对应于目标值的信号被恢复为时间轴上的信号。

公开(公告)号：US20100232620A1

公开(公告)日：2010-09-16

申请号：US12788107

申请日：2010-05-26

申请人： Naoshi MATSUO

发明人： Naoshi MATSUO

IPC分类号： H04R3/00

CPC分类号： H04S7/30

摘要： A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

摘要翻译： 声音处理装置包括：多个声音输入单元; 检测单元，用于检测输入到多个声音信号单元的每个声音的频率分量，每个声音从大致垂直于由多个声音输入单元中的两个声音输入单元的布置位置确定的线的方向到达; 校正系数单元，用于获得用于校正由两个声音输入单元从输入声音产生的至少一个声音信号的电平的校正系数，以便基于声音的声音来匹配声音信号的电平 检测频率分量; 校正单元，用于使用所获得的校正系数校正至少一个声音信号的电平; 以及处理单元，用于基于具有校正电平的声音信号执行声音处理。