公开(公告)号：US5602765A

公开(公告)日：1997-02-11

申请号：US278503

申请日：1994-07-21

申请人： Masashi Tanaka ,  Yutaka Kaneda ,  Shoji Makino ,  Yoichi Haneda ,  Junji Kojima

发明人： Masashi Tanaka ,  Yutaka Kaneda ,  Shoji Makino ,  Yoichi Haneda ,  Junji Kojima

IPC分类号： G05B13/04 ,  G05B23/02 ,  G06F17/10 ,  G06F17/16

CPC分类号： G06F17/16 ,  G05B13/048 ,  G06F17/10 ,  G10K2210/30232

摘要： In an adaptive estimation of an acoustic transfer function of an unknown system, a forward linear prediction coefficient vector a(k) of an input signal x(k), the sum of forward a posteriori prediction-error squares F(k), a backward linear prediction coefficient vector b(k) of the input signal x(k) and the sum of backward a posteriori prediction-error squares B(k) are computed. Letting a step size and a pre-filter deriving coefficient vector be represented by .mu. and f(k), respectively, a pre-filter coefficient vector g(k) is calculated by a recursion formula for the pre-filter coefficient vector g(h), which is composed of the following first and second equations: ##EQU1##

摘要翻译： 在未知系统的声传递函数的自适应估计中，输入信号x（k）的前向线性预测系数向量a（k），前向后验预测误差平方F（k），向后 计算输入信号x（k）的线性预测系数向量b（k）和后向后验预测误差平方B（k）的和。 通过分别用μ和f（k）表示步长和预滤波器导出系数向量，通过预滤波器系数向量g（h）的递归公式计算预滤波器系数向量g（k） ），其由以下第一和第二等式组成：

公开(公告)号：US5539731A

公开(公告)日：1996-07-23

申请号：US385989

申请日：1995-02-09

申请人： Yoichi Haneda ,  Shoji Makino ,  Masashi Tanaka ,  Yutaka Kaneda

发明人： Yoichi Haneda ,  Shoji Makino ,  Masashi Tanaka ,  Yutaka Kaneda

IPC分类号： H04M9/08

CPC分类号： H04M9/082

摘要： In an echo cancelling method of a p-order fast projection algorithm which subtracts an estimated echo signal y(k) from a microphone output signal u(k) to obtain an error signal e(k), adaptively calculates a pre-filter coefficient .beta.(k) from the auto-correlation of a received speech signal x(k) and the error signal, generating an intermediate variable z(k) updated by a coefficient s(k) obtained by smoothing the pre-filter coefficient, convolutes the received speech signal x(k) and the intermediate variable z(k), calculates the inner product of the auto-correlation of the received speech signal and the smoothed pre-filter coefficient s(k) and adding the inner product and the convoluted output to obtain the estimated echo signal, the magnitudes of the received speech signal x(k) and the error signal e(k) are compared and when the result of comparison satisfies a predetermined condition, a reset signal is generated to set the pre-filter coefficient .beta.(k) to zero for at least a period of time p, thereby preventing the accuracy of estimated echo characteristics from lowering during double-talk or send single-talk.

摘要翻译： 在从麦克风输出信号u（k）中减去估计回波信号+ E，cir y + EE（k）以获得误差信号e（k）的p阶快速投影算法的回波消除方法中，自适应地计算 来自接收到的语音信号x（k）的自相关的预滤波器系数β（k）和误差信号，生成由通过平滑预处理得到的系数s（k）更新的中间变量z（k） 滤波器系数，对接收到的语音信号x（k）和中间变量z（k）进行卷积，计算接收的语音信号的自相关和平滑的预滤波器系数s（k）的内积， 产品和卷积输出以获得估计的回波信号，接收到的语音信号x（k）和误差信号e（k）的大小进行比较，并且当比较结果满足预定条件时，产生复位信号 将预滤波器系数β（k）设置为零 一段时间p，从而防止估计的回波特性的精度在双向通话或发送单通话期间降低。

公开(公告)号：US5818945A

公开(公告)日：1998-10-06

申请号：US633295

申请日：1996-04-17

申请人： Shoji Makino ,  Yoichi Haneda ,  Akira Nakagawa ,  Masashi Tanaka ,  Suehiro Shimauchi ,  Junji Kojima

发明人： Shoji Makino ,  Yoichi Haneda ,  Akira Nakagawa ,  Masashi Tanaka ,  Suehiro Shimauchi ,  Junji Kojima

IPC分类号： H04B3/21 ,  H04M9/08 ,  H04B3/20

CPC分类号： H04B3/21 ,  H04M9/082

摘要： A received signal is output to an echo path and, at the same time, it is divided into a plurality of subbands to generate subband received signals, which are applied to estimated echo paths in the respective subbands to produce echo replicas. The echo having propagated over the echo path is divided into a plurality of subbands to generate subband echoes, from which the corresponding echo replicas are subtracted to produce misalignment signals. Based on the subband received signal in each subband and the misalignment signal corresponding thereto, a coefficient to be provided to each estimated echo path is adjusted by a projection or ES projection algorithm.

摘要翻译： 接收到的信号被输出到回波路径，并且同时被分成多个子带以产生子带接收信号，这些子带被应用于相应子带中的估计的回波路径以产生回波复制。 在回波路径上传播的回波被分成多个子带以产生子带回波，从中减去对应的回波复制以产生未对准信号。 基于每个子带中的子带接收信号和与其对应的未对准信号，通过投影或ES投影算法来调整要提供给每个估计回波路径的系数。

公开(公告)号：US06553122B1

公开(公告)日：2003-04-22

申请号：US09260515

申请日：1999-03-02

申请人： Suehiro Shimauchi ,  Yoichi Haneda ,  Shoji Makino ,  Yutaka Kaneda

发明人： Suehiro Shimauchi ,  Yoichi Haneda ,  Shoji Makino ,  Yutaka Kaneda

IPC分类号： H03B320

CPC分类号： H04M9/082

摘要： Even if received signals are highly cross-correlated, echoes can be effectively cancelled and no psychoacoustical problems arise. A received signal xi(k) (where i=1, 2, . . . , N) and an additive signal ai(k) are added together, and the added output is used to drive a speaker i and input into an echo cancellation filter 405i. The received signal xi(k) and the additive signal ai(k) are input into adaptive filters 401i and 402i, respectively. The difference between the sum of the outputs from all the filters 401i and all the filters 402i and an echo ym(k) is detected as an error em(k). The coefficients of all the filters 401i and 402i are updated to reduce the error em(k). When the error em(k) is made sufficiently small, the coefficients of the filters 402i are transferred to the filters 405i. The sum of the outputs from all the filters 405i is detected as an echo replica, and the difference between the echo replica and the echo ym(k) is output.

摘要翻译： 即使接收到的信号是高度相互关联的，也可以有效地消除回波并且不产生心理声学问题。接收信号xi（k）（其中i = 1,2，...，N）和加法信号ai（k） 被添加在一起，并且所添加的输出用于驱动扬声器i并输入到回声消除滤波器405i中。 接收信号xi（k）和加法信号ai（k）分别输入到自适应滤波器401i和402i中。 检测来自所有滤波器401i和全部滤波器402i的输出和回波ym（k）之间的差值作为误差em（k）。 所有滤波器401i和402i的系数被更新以减少误差em（k）。 当误差em（k）足够小时，滤波器402i的系数被传送到滤波器405i。 检测来自所有滤波器405i的输出的和作为回波复制，并且输出回波复制和回波ym（k）之间的差。

公开(公告)号：US5187692A

公开(公告)日：1993-02-16

申请号：US856654

申请日：1992-03-20

申请人： Yoichi Haneda ,  Shoji Makino ,  Yutaka Kaneda

发明人： Yoichi Haneda ,  Shoji Makino ,  Yutaka Kaneda

IPC分类号： G01H5/00 ,  G01H17/00 ,  G10K11/178 ,  H04R3/02 ,  H04R3/04 ,  H04R29/00 ,  H04S1/00 ,  H04S7/00

CPC分类号： H04S1/005 ,  H04R29/00 ,  H04R3/02 ,  H04R3/04 ,  H04S2420/01 ,  H04S7/30 ,  Y10S367/901

摘要： A plurality of acoustic transfer functions for a plurality of sets of different positions of a loudspeaker and a microphone in an acoustic system are measured by an acoustic transfer function measuring part. The plurality of measured acoustic transfer functions are used to estimate poles of the acoustic system by a pole estimation part, and a fixed AR filter is provided with the estimated poles as fixed values. A variable MA filter is connected in series to the fixed AR filter and the acoustic transfer function of the acoustic system is simulated by the two filters. The filter coefficients of the variable MA filter are modified with a change in the acoustic transfer function of the acoustic system.

公开(公告)号：US06246760B1

公开(公告)日：2001-06-12

申请号：US08927961

申请日：1997-09-11

申请人： Shoji Makino ,  Suehiro Shimauchi ,  Yoichi Haneda ,  Akira Nakagawa ,  Junji Kojima

发明人： Shoji Makino ,  Suehiro Shimauchi ,  Yoichi Haneda ,  Akira Nakagawa ,  Junji Kojima

IPC分类号： H04M100

CPC分类号： H04M9/082 ,  H04M1/60

摘要： In a subband echo cancellation for a multichannel teleconference, received signals x1(k), x2(k), . . . , xI(k) of each channel are divided into N subband signals, an echo y(k) picked up by a microphone 16j after propagation over an echo path is divided into N subband signals y0(k), . . . ,yN−1(k), and vectors each composed of a time sequence of subband received signals x1(k), . . . , xI(k) are combined for each corresponding subband. The combined vector and an echo cancellation error signal in the corresponding subband are input into an estimation part 19n, wherein a cross-correlation variation component is extracted. The extracted component is used as an adjustment vector to iteratively adjust the impulse response of an estimated echo path. The combined vector is applied to an estimated echo path 18n formed by the adjusted value to obtain an echo replica. An echo cancellation error signal en(k) is calculated from the echo replica and a subband echo yn(k).

摘要翻译： 在多声道电话会议的子带回波消除中，接收信号x1（k），x2（k），... 。 。 ，每个信道的xI（k）被划分为N个子带信号，在通过回波路径传播之后由麦克风16j拾取的回波y（k）被划分为N个子带信号y0（k）。 。 。 ，yN-1（k），以及由子带接收信号x1（k）的时间序列组成的矢量。 。 。 ，对于每个对应的子带组合xI（k）。 将相应子带中的组合矢量和回波消除误差信号输入到估计部分19n中，其中提取互相关变化分量。 提取的分量被用作调整向量以迭代地调整估计回波路径的脉冲响应。 将组合矢量应用于由调整值形成的估计回波路径18n，以获得回波复制品。 从回波复制品和子带回波yn（k）计算回波消除误差信号en（k）。

公开(公告)号：US5721772A

公开(公告)日：1998-02-24

申请号：US730247

申请日：1996-10-15

申请人： Yoichi Haneda ,  Shoji Makino ,  Akira Nakagawa ,  Suehiro Shimauchi ,  Junji Kojima

发明人： Yoichi Haneda ,  Shoji Makino ,  Akira Nakagawa ,  Suehiro Shimauchi ,  Junji Kojima

IPC分类号： H03H21/00 ,  H04B3/23 ,  H04M1/60 ,  H04M9/08 ,  H04M9/00 ,  G06F17/60

CPC分类号： H04M9/082

摘要： In a subband acoustic echo canceller, FG/BG filters are provided in M ones of N subbands into which the received signal is divided, and adaptive filters are provided in the other remaining subbands. In the respective FG/BG filters, during the detection of a non-double-talk state their transfer logic parts output state signals GD-j, GD-k, . . . and their adaptive operation control parts each apply an adaptation condition signal ADP to the adaptive filter in each of the above-mentioned other remaining subbands when a predetermined number or more of the FG/BG filters output the state signals GD-j, GD-k, . . . The adaptive filter updates the subband estimated echo path coefficient only when it is supplied with the signal ADP.

摘要翻译： 在子带声回波消除器中，在接收信号被分割的N个子带的M个中提供FG / BG滤波器，并且在其它剩余子带中提供自适应滤波器。 在相应的FG / BG滤波器中，在检测非双方通话状态期间，它们的传输逻辑部分输出状态信号GD-j，GD-k。 。 。 并且当预定数量或更多的FG / BG滤波器输出状态信号GD-j，GD-k时，它们的自适应操作控制部分在自适应滤波器的每一个中将自适应条件信号ADP应用于上述其他剩余子带中的每一个 ，。 。 。 只有当信号ADP被提供时，自适应滤波器才更新子带估计回波路径系数。

公开(公告)号：US5774561A

公开(公告)日：1998-06-30

申请号：US695446

申请日：1996-08-12

申请人： Akira Nakagawa ,  Yoichi Haneda ,  Shoji Makino ,  Suehiro Shimauchi ,  Junji Kojima

发明人： Akira Nakagawa ,  Yoichi Haneda ,  Shoji Makino ,  Suehiro Shimauchi ,  Junji Kojima

IPC分类号： H04M9/08 ,  H04B3/20

CPC分类号： H04M9/082

摘要： In a subband acoustic echo canceller which generates an echo replica from a subband received signal x.sub.k (m) by an estimated echo path in each subband, subtracts the echo replica from a subband echo signal y.sub.k (m) by a subtractor to generate a subband error signal e.sub.k (m) and uses an adaptive algorithm in an echo path estimation part to estimate the transfer function of the estimated echo path from the subband error signal e.sub.k (m) and the subband received signal x.sub.k (m) so that the subband error signal e.sub.k (m) approaches zero, the stop-band attenuation of each band-pass filter of a received signal subband analysis part for generating the subband received signal x.sub.k (m) is set to be smaller than the stop-band attenuation of each band-pass filter of an echo subband analysis part for generating the subband echo signal Y.sub.k (m) to thereby flatten the frequency characteristics of the subband received signals relative to the subband echo signals.

摘要翻译： 在通过每个子带中的估计回波路径从子带接收信号xk（m）生成回波复制品的子带声回波消除器中，通过减法器从子带回波信号yk（m）中减去回波复制品，以产生子带误差 信号ek（m），并且在回波路径估计部分中使用自适应算法来估计来自子带误差信号ek（m）和子带接收信号xk（m）的估计回波路径的传递函数，使得子带误差信号 ek（m）接近零时，用于产生子带接收信号xk（m）的接收信号子带分析部分的每个带通滤波器的阻带衰减被设置为小于每个频带衰减的阻带衰减， 用于产生子带回波信号Y k（m）的回波子带分析部件的通过滤波器，从而相对于子带回波信号使子带接收信号的频率特性变平。

公开(公告)号：US5272695A

公开(公告)日：1993-12-21

申请号：US756622

申请日：1991-09-09

申请人： Shoji Makino ,  Yoichi Haneda ,  Yutaka Kanesa

发明人： Shoji Makino ,  Yoichi Haneda ,  Yutaka Kanesa

IPC分类号： H03H21/00 ,  H04B3/23 ,  H04M9/08 ,  H04J1/00

CPC分类号： H04M9/082

摘要： A received input signal and an echo signal resulting from the passage of the received input signal through an echo path are both analyzed or divided into a plurality of common subbands. The received input signal in each subband is supplied to an estimated echo path provided in the subband, by which it is rendered into an echo replica signal. The echo replica signal is subtracted, by a subtractor provided in each subband, from the echo signal in the same subband as the echo replica signal to obtain a residual echo signal. The residual echo signals in the respective subbands are synthesized into a full-band residual echo signal. The estimated echo path in each subband is formed by a digital FIR filter and its filter coefficients are calculated by a coefficient calculation part in the subband, based on the received input signal, the residual echo signal and a step size matrix. The filter coefficients are iteratively updated so that the residual echo signal in each subband may be minimized. The step size matrix is used to define the step size of the filter coefficients and is determined by an acoustic field characteristics calculation part, based on the variation characteristics of an impulse response of the echo path in each subband.

摘要翻译： 接收到的输入信号和由接收到的输入信号通过回波路径而产生的回波信号都被分析或分成多个公共子带。 每个子带中的接收到的输入信号被提供给在子带中提供的估计回波路径，由此将其呈现为回波复制信号。 通过在每个子带中提供的减法器从与回波复制信号相同子带中的回波信号中减去回波复制信号，以获得残留回波信号。 各个子带中的残余回波信号被合成为全带残留回波信号。 每个子带中的估计回波路径由数字FIR滤波器形成，并且其滤波器系数由子带中的系数计算部分基于接收到的输入信号，残余回波信号和步长矩阵来计算。 滤波器系数被迭代地更新，使得每个子带中的残余回波信号可以被最小化。 步长矩阵用于定义滤波器系数的步长，并且由声场特性计算部分基于每个子带中的回波路径的脉冲响应的变化特性来确定。

公开(公告)号：US09191738B2

公开(公告)日：2015-11-17

申请号：US13996302

申请日：2011-12-19

申请人： Kenta Niwa ,  Sumitaka Sakauchi ,  Kenichi Furuya ,  Yoichi Haneda

发明人： Kenta Niwa ,  Sumitaka Sakauchi ,  Kenichi Furuya ,  Yoichi Haneda

IPC分类号： H04R3/00 ,  G10L21/0232 ,  G10L21/0208 ,  G10L21/0216

CPC分类号： H04R3/00 ,  G10L21/0232 ,  G10L2021/02082 ,  G10L2021/02166 ,  H04R3/005 ,  H04R2430/03

摘要： A sound enhancement technique that uses transfer functions ai,g of sounds that come from each of one or more positions/directions that are assumed to be sound sources arriving at each microphone to obtain a filter for a position that is a target of sound enhancement, where i denotes a direction and g denotes a distance for identifying each of the positions. Each of the transfer functions ai,g is represented by sum of a transmission characteristic of a direct sound that directly arrives from the position determined by the direction i and the distance g and a transmission characteristic of one or more reflected sounds produced by reflection of the direct sound off an reflective object. A filter that corresponds to the position that is the target of sound enhancement is applied to frequency-domain signals transformed from M picked-up sounds picked up with M microphones to obtain a frequency-domain output signal.

摘要翻译： 一种声音增强技术，其使用传送函数ai，g，来自每个被认为是声源的一个或多个位置/方向的声音到达每个麦克风，以获得用于声音增强的目标的位置的滤波器， 其中i表示方向，g表示用于识别每个位置的距离。 传递函数ai，g中的每一个由从由方向i确定的位置和距离g直接到达的直接声音的传输特性和通过反射产生的一个或多个反射声音的传输特性的和来表示 直接关闭反光物体。 对应于作为声音增强的目标的位置的滤波器被应用于从用M个麦克风拾取的M个拾取的声音变换的频域信号，以获得频域输出信号。