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1. 发明授权

US08391505B2 Reverberation suppressing apparatus and reverberation suppressing method 有权
标题翻译：混响抑制装置和混响抑制方法
公开(公告)号：US08391505B2
公开(公告)日：2013-03-05
申请号：US12791428
申请日：2010-06-01
申请人： Kazuhiro Nakadai , Hirofumi Nakajima , Hiroshi Okuno , Ryu Takeda
发明人： Kazuhiro Nakadai , Hirofumi Nakajima , Hiroshi Okuno , Ryu Takeda
IPC分类号： H04B3/20
CPC分类号： H04M9/082
摘要： A reverberation suppressing apparatus separating sound source signals based on input signals output from microphones collecting the plurality of sound source signals, includes a sound signal output unit generating sound signals and outputting the generated sound signals, a sound acquiring unit acquiring the input signals from microphones, a first evaluation function calculation unit calculating a separation matrix, the input signals, and the sound source signals, and calculating a first evaluation function, a reverberation component suppressing unit calculating an optimal separation matrix, and suppressing a reverberation component by separating the sound source signals other than the generated sound signals, and a separation matrix updating unit dividing a step-size function, approximating each segment to a linear function, calculating step sizes based on the approximated linear functions, and repeatedly updating the separation matrix so that the degree of separation of the sound source signals exceeds the predetermined value.
摘要翻译：基于从收集多个声源信号的麦克风输出的输入信号分离声源信号的混响抑制装置包括产生声音信号并输出所生成的声音信号的声音信号输出单元，从麦克风获取输入信号的声音获取单元，第一评估函数计算单元，计算分离矩阵，输入信号和声源信号，并计算第一评估函数，混响分量抑制单元计算最佳分离矩阵，并通过分离声源信号来抑制混响分量除了产生的声音信号之外，还有分离矩阵更新单元，将步长函数除以近似线性函数，根据近似线性函数计算步长，并重复更新分离矩阵，使得分离度的声音很酸 ce信号超过预定值。

2. 发明授权

US08867754B2 Dereverberation apparatus and dereverberation method 有权
标题翻译：混频器和混频器
公开(公告)号：US08867754B2
公开(公告)日：2014-10-21
申请号：US12704582
申请日：2010-02-12
申请人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa
发明人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa
IPC分类号： H04R29/00 , H04R27/00 , H04R1/40
CPC分类号： H04R27/00 , H04R1/406 , H04R2227/009
摘要： A dereverberation apparatus includes a signal selecting unit which selects a sound signal to be used for dereverberation process from a plurality of sound signals, and a dereverberation processing unit which performs the dereverberation process for the selected sound signal.
摘要翻译：解混乱装置包括：信号选择单元，其从多个声音信号中选择要用于去混响处理的声音信号;以及混响处理单元，其对所选择的声音信号执行去混响处理。

3. 发明授权

US08265290B2 Dereverberation system and dereverberation method 有权
标题翻译：混沌系统和混响方法
公开(公告)号：US08265290B2
公开(公告)日：2012-09-11
申请号：US12548871
申请日：2009-08-27
申请人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
发明人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
IPC分类号： H04B3/20
CPC分类号： H04S1/002
摘要： Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R+) of a non-square matrix (R) as a correlation matrix of input signals (x). On the basis of the inverse filter (h) and an estimated correlation matrix (R^) generated according to a window function (w), an error cost (J(h) between a correlation value of the input signals (x) and output signals (y) and a desired correlation value (d) is calculated. On the basis of the error cost (J(h)), the inverse filter (h) is adaptively updated according to a gradient method.
摘要翻译：提供了一种能够灵活地处理任意条件并且能够识别声音或声源信号的混响系统等。根据混响系统，通过使用非正方形矩阵（R）的伪逆矩阵（R +）作为输入信号（x）的相关矩阵来设置逆滤波器（h）。根据逆滤波器（h）和根据窗函数（w）产生的估计相关矩阵（R ^），输入信号（x）的相关值和输出信号计算信号（y）和期望的相关值（d），根据误差成本（J（h）），根据梯度法自适应地更新逆滤波器（h）。

4. 发明申请

US20090063605A1 SIGNAL PROCESSING DEVICE 有权
标题翻译：信号处理装置
公开(公告)号：US20090063605A1
公开(公告)日：2009-03-05
申请号：US12198488
申请日：2008-08-26
申请人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
发明人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
IPC分类号： G06F17/15
CPC分类号： G06F17/15
摘要： A device capable of improving the convergence rate and estimation accuracy in estimating a correlation value. According to a signal processing device, since a window length is adjusted in such a manner to reduce an estimated error of a correlation matrix, the convergence rate and estimation accuracy in estimating the correlation matrix and the correlation value as its off-diagonal element can be improved. Then, in such a high-probability condition that the correlation of plural output signals according to a state is estimated with a high degree of precision, signal processing is performed on the plural signals, so that the state can be estimated with a high degree of precision.
摘要翻译：一种在估计相关值时能够提高收敛速度和估计精度的装置。根据信号处理装置，由于以减小相关矩阵的估计误差的方式调整窗口长度，所以估计相关矩阵的收敛速度和估计精度以及相关值作为其对角线元素可以是改进。然后，在以高精度估计出多个根据状态的输出信号的相关性的高概率条件下，对多个信号执行信号处理，从而能够以高度的精确。

5. 发明授权

US08131542B2 Sound source separation system which converges a separation matrix using a dynamic update amount based on a cost function 有权
标题翻译：声源分离系统，其使用基于成本函数的动态更新量收敛分离矩阵
公开(公告)号：US08131542B2
公开(公告)日：2012-03-06
申请号：US12133691
申请日：2008-06-05
申请人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
发明人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
IPC分类号： G10L19/00 , H03F1/26
CPC分类号： G06K9/6242 , G06K9/6245 , G10L21/028 , G10L2021/02166
摘要： A system capable of separating sound source signals with high precision while improving a convergence rate and convergence precision. A process of updating a current separation matrix Wk to a next separation matrix Wk+1 such that a next value J(Wk+1) of a cost function is closer to a minimum value J(W0) than a current value J(Wk) is iteratively performed. An update amount ΔWk of the separation matrix is increased as the current value J(Wk) of the cost function is increased and is decreased as a current gradient ∂J(Wk)/∂W of the cost function is rapid. On the basis of input signals x from a plurality of microphones Mi and an optimal separation matrix W0, it is possible to separate sound source signals y(=W0·x) with high precision while improving a convergence rate and convergence precision.
摘要翻译：一种能够在提高收敛速度和收敛精度的同时高精度地分离声源信号的系统。将当前分离矩阵Wk更新为下一个分离矩阵Wk + 1的过程，使得成本函数的下一个值J（Wk + 1）比当前值J（Wk）更接近最小值J（W0）被迭代地执行。随着成本函数的当前值J（Wk）增加，并且随着成本函数的当前梯度∂J（Wk）/∂W）快速地减小，分离矩阵的更新量＆Dgr; Wk增加。基于来自多个麦克风Mi的输入信号x和最优分离矩阵W0，可以在提高收敛速度和收敛精度的同时高精度地分离声源信号y（= W0·x）。

6. 发明申请

US20080306739A1 SOUND SOURCE SEPARATION SYSTEM 有权
标题翻译：声源分离系统
公开(公告)号：US20080306739A1
公开(公告)日：2008-12-11
申请号：US12133691
申请日：2008-06-05
申请人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
发明人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
IPC分类号： G10L11/00
CPC分类号： G06K9/6242 , G06K9/6245 , G10L21/028 , G10L2021/02166
摘要： A system capable of separating sound source signals with high precision while improving a convergence rate and convergence precision. A process of updating a current separation matrix Wk to a next separation matrix Wk+1 such that a next value J(Wk+1) of a cost function is closer to a minimum value J(W0) than a current value J(Wk) is iteratively performed. An update amount ΔWk of the separation matrix is increased as the current value J(Wk) of the cost function is increased and is decreased as a current gradient ∂J(Wk)/∂W of the cost function is rapid. On the basis of input signals x from a plurality of microphones Mi and an optimal separation matrix W0, it is possible to separate sound source signals y(=W0·x) with high precision while improving a convergence rate and convergence precision.
摘要翻译：一种能够在提高收敛速度和收敛精度的同时高精度地分离声源信号的系统。将当前分离矩阵Wk更新为下一个分离矩阵Wk + 1的过程，使得成本函数的下一个值J（Wk + 1）比当前值J（Wk）更接近最小值J（W0）被迭代地执行。随着成本函数的当前值J（Wk）增加，并且随着成本函数的当前梯度∂J（Wk）/∂W）快速地减小，分离矩阵的更新量DeltaWk增加。基于来自多个麦克风Mi和最佳分离矩阵W0的输入信号x，可以在提高收敛速度和收敛精度的同时高精度地分离声源信号y（= W0.x）。

7. 发明授权

US08867755B2 Sound source separation apparatus and sound source separation method 有权
标题翻译：声源分离装置和声源分离方法
公开(公告)号：US08867755B2
公开(公告)日：2014-10-21
申请号：US13211002
申请日：2011-08-16
申请人： Kazuhiro Nakadai , Hirofumi Nakajima
发明人： Kazuhiro Nakadai , Hirofumi Nakajima
IPC分类号： H04B3/20 , G10L21/028 , G10L21/0216
CPC分类号： G10L21/028 , G10L2021/02166
摘要： A sound source separation apparatus includes a transfer function storage unit that stores a transfer function from a sound source, a sound change detection unit that generates change state information indicating a change of the sound source on the basis of an input signal input from a sound input unit, a parameter selection unit that calculates an initial separation matrix on the basis of the change state information generated by the sound change detection unit, and a sound source separation unit that separates the sound source from the input signal input from the sound input unit using the initial separation matrix calculated by the parameter selection unit.
摘要翻译：声源分离装置包括存储来自声源的传递函数的传递函数存储单元，声音变化检测单元，其基于从声音输入输入的输入信号生成指示声源的变化的改变状态信息单元，参数选择单元，其基于由声音变化检测单元生成的改变状态信息来计算初始分离矩阵;以及声源分离单元，其将声源与从声音输入单元输入的输入信号分离，使用由参数选择单元计算的初始分离矩阵。

8. 发明授权

US09055356B2 Sound source localization apparatus and sound source localization method 有权
标题翻译：声源定位装置和声源定位方法
公开(公告)号：US09055356B2
公开(公告)日：2015-06-09
申请号：US12791423
申请日：2010-06-01
申请人： Kazuhiro Nakadai , Hirofumi Nakajima , Keisuke Nakamura
发明人： Kazuhiro Nakadai , Hirofumi Nakajima , Keisuke Nakamura
IPC分类号： H04B15/00 , H04R3/00 , B25J9/00
CPC分类号： H04R3/005 , B25J9/00
摘要： A sound source localization apparatus for localizing a sound source using an eigenvector, includes, a sound signal input unit inputting a sound signal, a correlation matrix calculation unit calculating a correlation matrix of the input sound signal, and an eigenvector calculation unit calculating an eigenvalue of the correlation matrix using the calculated correlation matrix, wherein the eigenvector calculation unit calculates the eigenvector using the correlation matrix of the input sound signal and one or more predetermined correlation matrices.
摘要翻译：使用特征向量来定位声源的声源定位装置包括：输入声音信号的声音信号输入单元，计算输入声音信号的相关矩阵的相关矩阵计算单元和计算特征值的特征值的特征向量计算单元使用所计算的相关矩阵的相关矩阵，其中特征向量计算单元使用输入声音信号和一个或多个预定相关矩阵的相关矩阵来计算特征向量。

9. 发明授权

US08831237B2 Sound source separation apparatus and sound source separation method 有权
公开(公告)号：US08831237B2
公开(公告)日：2014-09-09
申请号：US13211002
申请日：2011-08-16
申请人： Kazuhiro Nakadai , Hirofumi Nakajima
发明人： Kazuhiro Nakadai , Hirofumi Nakajima
IPC分类号： H04B3/20 , G10L21/028 , G10L21/0216
摘要： A sound source separation apparatus includes a transfer function storage unit that stores a transfer function from a sound source, a sound change detection unit that generates change state information indicating a change of the sound source on the basis of an input signal input from a sound input unit, a parameter selection unit that calculates an initial separation matrix on the basis of the change state information generated by the sound change detection unit, and a sound source separation unit that separates the sound source from the input signal input from the sound input unit using the initial separation matrix calculated by the parameter selection unit.

10. 发明申请

US20100054489A1 DEREVERBERATION SYSTEM AND DEREVERBERATION METHOD 有权
标题翻译： DEREVERBERATION系统和DEREVERBERATION方法
公开(公告)号：US20100054489A1
公开(公告)日：2010-03-04
申请号：US12548871
申请日：2009-08-27
申请人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
发明人： Hirofumi Nakajima , Kazuhiro Nakadai , Yuji Hasegawa , Hiroshi Tsujino
IPC分类号： H04B3/20
CPC分类号： H04S1/002
摘要： Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R+) of a non-square matrix (R) as a correlation matrix of input signals (x). On the basis of the inverse filter (h) and an estimated correlation matrix (R̂) generated according to a window function (w), an error cost (J(h) between a correlation value of the input signals (x) and output signals (y) and a desired correlation value (d) is calculated. On the basis of the error cost (J(h)), the inverse filter (h) is adaptively updated according to a gradient method.
摘要翻译：提供了一种能够灵活地处理任意条件并且能够识别声音或声源信号的混响系统等。根据混响系统，通过使用非正方形矩阵（R）的伪逆矩阵（R +）作为输入信号（x）的相关矩阵来设置逆滤波器（h）。根据逆滤波器（h）和根据窗函数（w）产生的估计相关矩阵（R），输入信号（x）的相关值和输出信号（x）之间的误差成本（J（h）（y）和期望的相关值（d），基于误差成本（J（h）），逆滤波器（h）根据梯度法自适应地更新。

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