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）。

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）。

公开(公告)号：US08799342B2

公开(公告)日：2014-08-05

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，对多个信号进行信号处理，从而能够以高度的 精确。

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）。

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。

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。

公开(公告)号：US08290178B2

公开(公告)日：2012-10-16

申请号：US12010553

申请日：2008-01-25

申请人： Kazuhiro Nakadai ,  Hiroshi Tsujino ,  Hirofumi Nakajima

发明人： Kazuhiro Nakadai ,  Hiroshi Tsujino ,  Hirofumi Nakajima

IPC分类号： H04R3/00 ,  G01S3/80

CPC分类号： H04S7/00 ,  H04R3/005

摘要： There is provided a sound source characteristic determining device (10) capable of being applied in an environmental where the type of a sound source is unknown. The device includes a plurality of beamformers (21-1 to 21-M) used when a sound source signal generated from a sound source at an arbitrary position in a space is inputted to a plurality of microphones (14-1 to 14-N), for weighting the acoustic signal detected by each of the microphones by using a function for correcting the difference of the sound source signals generated between the microphones and outputting a totaled signal. Each of the beamformers (21-1 to 21-M) contains a function having a unit directivity characteristic corresponding to one arbitrary direction in the space and is arranged for each of the directions corresponding to an arbitrary position in the space and the unit directivity characteristic. The sound source characteristic determining device (10) further includes means (23) for estimating the position and the direction in the space corresponding to the beamformer outputting a maximum value as the position and the direction of the sound source when the microphone (14) detects a sound source signal.

摘要翻译： 提供了能够应用于声源类型未知的环境中的声源特性确定装置（10）。 当从空间中的任意位置的声源产生的声源信号输入到多个麦克风（14-1至14-N）时，该装置包括多个波束形成器（21-1至21-M） 用于通过使用用于校正在麦克风之间产生的声源信号的差异并输出总计信号的功能来对由每个麦克风检测到的声信号进行加权。 每个波束形成器（21-1至21-M）包含具有对应于该空间中的一个任意方向的单位方向性特性的功能，并且被布置用于与空间中的任意位置相对应的每个方向，并且单位方向特性 。 声源特性确定装置（10）还包括用于估计与波束形成器相对应的空间中的位置和方向的装置（23），其输出最大值作为麦克风（14）检测到的声源的位置和方向 声源信号。

公开(公告)号：US20080199024A1

公开(公告)日：2008-08-21

申请号：US12010553

申请日：2008-01-25

申请人： Kazuhiro Nakadai ,  Hiroshi Tsujino ,  Hirofumi Nakajima

发明人： Kazuhiro Nakadai ,  Hiroshi Tsujino ,  Hirofumi Nakajima

IPC分类号： H04R3/00

CPC分类号： H04S7/00 ,  H04R3/005

摘要： There is provided a sound source characteristic determining device (10) capable of being applied in an environmental where the type of a sound source is unknown. The device includes a plurality of beamformers (21-1 to 21-M) used when a sound source signal generated from a sound source at an arbitrary position in a space is inputted to a plurality of microphones (14-1 to 14-N), for weighting the acoustic signal detected by each of the microphones by using a function for correcting the difference of the sound source signals generated between the microphones and outputting a totaled signal. Each of the beamformers (21-1 to 21-M) contains a function having a unit directivity characteristic corresponding to one arbitrary direction in the space and is arranged for each of the directions corresponding to an arbitrary position in the space and the unit directivity characteristic. The sound source characteristic determining device (10) further includes means (23) for estimating the position and the direction in the space corresponding to the beamformer outputting a maximum value as the position and the direction of the sound source when the microphone (14) detects a sound source signal.

摘要翻译： 提供了能够应用于声源类型未知的环境中的声源特性确定装置（10）。 当从空间中的任意位置的声源产生的声源信号输入到多个麦克风（14-1至14-N）时，该装置包括多个波束形成器（21-1至21-M） 用于通过使用用于校正在麦克风之间产生的声源信号的差异并输出总计信号的功能来对由每个麦克风检测到的声信号进行加权。 每个波束形成器（21-1至21-M）包含具有对应于该空间中的一个任意方向的单位方向特性的功能，并且被布置用于与空间中的任意位置相对应的每个方向，并且单位方向特性 。 声源特性确定装置（10）还包括用于估计与波束形成器相对应的空间中的位置和方向的装置（23），其输出最大值作为麦克风（14）检测到的声源的位置和方向 声源信号。

公开(公告)号：US07999168B2

公开(公告)日：2011-08-16

申请号：US12503448

申请日：2009-07-15

申请人： Kazuhiro Nakadai ,  Yuji Hasegawa ,  Hiroshi Tsujino ,  Kazumasa Murata ,  Ryu Takeda ,  Hiroshi Okuno

发明人： Kazuhiro Nakadai ,  Yuji Hasegawa ,  Hiroshi Tsujino ,  Kazumasa Murata ,  Ryu Takeda ,  Hiroshi Okuno

IPC分类号： G10H1/00

CPC分类号： G10H1/36 ,  A63H3/28 ,  G10H1/40 ,  G10H2240/135 ,  G10H2250/371 ,  G10H2250/455

摘要： A robot includes: a sound collecting unit collecting and converting a musical sound into a musical acoustic signal; a voice signal generating unit generating a self-vocalized voice signal; a sound outputting unit converting the self-vocalized voice signal into a sound and outputting the sound; a self-vocalized voice regulating unit receiving the musical acoustic signal and the self-vocalized voice signal; a filtering unit performing a filtering process; a beat interval reliability calculating unit performing a time-frequency pattern matching process and calculating a beat interval reliability; a beat interval estimating unit estimating a beat interval; a beat time reliability calculating unit calculating a beat time reliability; a beat time estimating unit estimating a beat time on the basis of the calculated beat time reliability; a beat time predicting unit predicting a beat time before the current time; and a synchronization unit synchronizing the self-vocalized voice signal.

摘要翻译： 机器人包括：收集单元，收集并将音乐声音转换成音乐声信号; 产生自发声音信号的语音信号产生单元; 声音输出单元，将自发声音信号转换成声音并输出声音; 接收音乐声音信号和自发声音信号的自发声音调节单元; 执行滤波处理的滤波单元; 节拍间隔可靠性计算单元，执行时间频率模式匹配处理并计算节拍间隔可靠性; 估计拍子间隔的拍子间隔估计单元; 节拍时间可靠性计算单元，计算节拍时间可靠性; 拍子时间估计单元，基于所计算的拍子时间可靠性来估计拍子时间; 拍子时间预测单元预测当前时间之前的拍子时间; 以及使自发声音信号同步的同步单元。

公开(公告)号：US08594846B2

公开(公告)日：2013-11-26

申请号：US12503431

申请日：2009-07-15

申请人： Kazuhiro Nakadai ,  Yuji Hasegawa ,  Hiroshi Tsujino ,  Kazumasa Murata ,  Ryu Takeda ,  Hiroshi Okuno

发明人： Kazuhiro Nakadai ,  Yuji Hasegawa ,  Hiroshi Tsujino ,  Kazumasa Murata ,  Ryu Takeda ,  Hiroshi Okuno

IPC分类号： G05B15/00 ,  G05B19/00

CPC分类号： G10H1/36 ,  A63H3/28 ,  G10H1/40 ,  G10H2240/135 ,  G10H2250/371 ,  G10H2250/455

摘要： A beat tracking apparatus includes: a filtering unit configured to perform a filtering process on an input acoustic signal and to accentuate an onset; a beat interval reliability calculating unit configured to perform a time-frequency pattern matching process employing a mutual correlation function on the acoustic signal of which the onset is accentuated and to calculate a beat interval reliability; and a beat interval estimating unit configured to estimate a beat interval on the basis of the calculated beat interval reliability.

摘要翻译： 拍子跟踪装置包括：滤波单元，被配置为对输入的声信号执行滤波处理并加重起点; 拍子间隔可靠性计算单元，被配置为对所述开始强调的声学信号执行采用相互关联函数的时间 - 频率模式匹配处理，并且计算拍子间隔可靠性; 以及拍摄间隔估计单元，被配置为基于所计算的拍子间隔可靠性来估计拍子间隔。