Low noise differential microphone arrays
    1.
    发明授权
    Low noise differential microphone arrays 有权
    低噪声差分麦克风阵列

    公开(公告)号:US09237391B2

    公开(公告)日:2016-01-12

    申请号:US13816430

    申请日:2012-12-04

    摘要: A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.

    摘要翻译: 提供差分麦克风阵列(DMA),其包括用于将声音转换为多个电信号的麦克风传感器的数量(M),以及经配置以在一段时间内在电信号上应用线性约束的最小方差滤波器的处理器 窗口以计算多个子带上的电信号的频率响应,并且对每个子带的电信号的频率响应求和,以计算声音的估计频谱。

    Method and apparatus for receiving digital wireless transmissions using multiple-antenna communication schemes
    2.
    发明授权
    Method and apparatus for receiving digital wireless transmissions using multiple-antenna communication schemes 有权
    用于使用多天线通信方案接收数字无线传输的方法和装置

    公开(公告)号:US07136437B2

    公开(公告)日:2006-11-14

    申请号:US10196865

    申请日:2002-07-17

    IPC分类号: H04B7/10

    CPC分类号: H04L1/06

    摘要: A signal detection technique for multiple-input multiple-output (MIMO) communications systems embodied in a method and apparatus for detecting a plurality of transmitted signals with use of a plurality of receiving antennas. An iterative procedure decodes one of a plurality of transmitted signals at each iteration using an intermediate matrix at each iteration to determine the transmitted signal to be decoded. The intermediate matrix for each successive iteration is advantageously computed in a recursive manner with use of a Schur complement operation performed based on the inverse of a modified version of the intermediate matrix used in the previous iteration.

    摘要翻译: 一种在用于使用多个接收天线检测多个发射信号的方法和装置中实现的多输入多输出(MIMO)通信系统的信号检测技术。 迭代过程在每次迭代中使用中间矩阵在每次迭代中解码多个发送信号中的一个,以确定要解码的发送信号。 有利地,使用基于在前一次迭代中使用的中间矩阵的修改版本的反向执行的Schur补码操作,递归地计算每个连续迭代的中间矩阵。

    LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS
    3.
    发明申请
    LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS 有权
    低噪声差分麦克风阵列

    公开(公告)号:US20160134969A1

    公开(公告)日:2016-05-12

    申请号:US14980802

    申请日:2015-12-28

    IPC分类号: H04R3/04 H04R3/00 H04R1/40

    摘要: A differential microphone array includes a number (M) of microphone sensors for converting sound to a number of electrical signals, and a processor, operably coupled to the microphone sensors, to specify a target differential order (N) for the differential microphone array, and wherein M>N+1, specify a steering matrix D comprising N+1 steering vectors, calculate a respective one of a plurality of linearly specify a steering matrix D comprising N+1 steering vectors-constrained minimum variance filters based on the steering matrix, apply the respective one of the plurality of linearly-constrained minimum variance filters to a respective one of the electrical signals to calculate a respective frequency response of the electrical signals, wherein the respective frequency response comprises a plurality of components associated with a plurality of subbands, and sum the frequency responses of the electrical signals with respect to each subband to calculate an estimated frequency spectrum of the sound.

    摘要翻译: 差分麦克风阵列包括用于将声音转换为多个电信号的麦克风传感器数量(M),以及可操作地耦合到麦克风传感器的处理器,以指定差分麦克风阵列的目标差分顺序(N),以及 其中M> N + 1指定包括N + 1个导向矢量的导引矩阵D,计算多个线性地指定基于导引矩阵的N + 1个导向矢量约束最小方差滤波器的导引矩阵D中的相应一个, 将所述多个线性约束最小方差滤波器中的相应一个应用到所述电信号中的相应一个,以计算所述电信号的相应频率响应,其中所述相应频率响应包括与多个子带相关联的多个分量, 并且相对于每个子带对电信号的频率响应进行求和,以计算t的估计频谱 他听起来

    LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS
    4.
    发明申请
    LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS 有权
    低噪声差分麦克风阵列

    公开(公告)号:US20150163577A1

    公开(公告)日:2015-06-11

    申请号:US13816430

    申请日:2012-12-04

    IPC分类号: H04R1/08 H04R3/04

    摘要: A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.

    摘要翻译: 提供差分麦克风阵列(DMA),其包括用于将声音转换为多个电信号的麦克风传感器的数量(M),以及经配置以在一段时间内在电信号上应用线性约束的最小方差滤波器的处理器 窗口以计算多个子带上的电信号的频率响应,并且对每个子带的电信号的频率响应求和,以计算声音的估计频谱。

    Method and apparatus for network echo cancellation using a proportionate normalized least mean squares algorithm
    5.
    发明授权
    Method and apparatus for network echo cancellation using a proportionate normalized least mean squares algorithm 有权
    使用比例归一化最小均方算法进行网络回波消除的方法和装置

    公开(公告)号:US06526141B2

    公开(公告)日:2003-02-25

    申请号:US09730442

    申请日:2000-12-05

    IPC分类号: H04M100

    CPC分类号: H04B3/23 H03H2021/0063

    摘要: The invention is a method and apparatus for performing adaptive filtering, and particularly echo cancellation, utilizing an efficient and effective adaptive algorithm. The invention is particularly useful in connection with network echo cancellation but is more broadly applicable to any situation where an adaptive estimate of a signal must be generated in real-time. The invention includes an improved proportionate normalized least mean squares algorithm for generating an impulse response estimate that is useful for generating an echo cancellation signal to be subtracted from the echo containing signal.

    摘要翻译: 本发明是一种利用有效且有效的自适应算法执行自适应滤波,特别是回波消除的方法和装置。 本发明在网络回波消除方面特别有用,但更广泛地适用于必须实时地产生信号的自适应估计的任何情况。本发明包括用于产生脉冲响应的改进的成比例的归一化最小均方算法 估计对于生成要从回波包含信号中减去的回波消除信号是有用的。

    Multi-channel frequency-domain adaptive filter method and apparatus
    6.
    发明授权
    Multi-channel frequency-domain adaptive filter method and apparatus 有权
    多通道频域自适应滤波方法及装置

    公开(公告)号:US07693291B2

    公开(公告)日:2010-04-06

    申请号:US11937669

    申请日:2007-11-09

    IPC分类号: H04B3/20 H04M9/08

    CPC分类号: H04M9/082

    摘要: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

    摘要翻译: 本发明是用于频域自适应滤波的方法和装置,其具有广泛的应用,例如均衡器,但是特别适用于立体声和其他多声道电话会议系统的声学回波消除电路。 该方法和装置利用使频域中的误差信号最小化的频域递归最小二乘准则。 为了降低算法的复杂度,去除约束,导致无约束的频域递归最小均方法和装置。 公开了一种用于选择UFLMS的最佳适应步骤的方法和装置。 该方法和装置被广泛应用于多信道情况,并利用所有信道中的交叉功率谱。

    Adaptive filter for network echo cancellation
    7.
    发明授权
    Adaptive filter for network echo cancellation 失效
    用于网络回波消除的自适应滤波器

    公开(公告)号:US07072465B1

    公开(公告)日:2006-07-04

    申请号:US09228772

    申请日:1999-01-06

    IPC分类号: H04B3/23

    CPC分类号: H04B3/234 H03H2021/0063

    摘要: A robust adaptive filter for use in a network echo canceller or other digital signal processing application utilizes a coefficient vector update device that, through the application of fast converging algorithms to a fast impulse response filter yields fast convergence of the adaptive filter's characteristics with the avoidance of divergence due to the onset of double talk. Robustness is also provided, via an adaptive scale non-linearity device which applies an adaptive scale non-linearity to the filter algorithms fed to the fast impulse response filter by the coefficient vector update device, so that the samples of an echo signal to be cancelled which are taken during the onset of double talk can be handled in such a manner that after the double talk detector causes adaptation to cease, the initial, potentially disturbing samples do not cause significant divergence in the filter system.

    摘要翻译: 用于网络回波消除器或其他数字信号处理应用的鲁棒自适应滤波器利用系数向量更新装置,其通过将快速收敛算法应用于快速脉冲响应滤波器产生自适应滤波器特征的快速收敛,避免了 由于双重谈话的开始,分歧。 还通过自适应标度非线性装置提供鲁棒性,其对由系数向量更新装置馈送到快速脉冲响应滤波器的滤波器算法施加自适应标度非线性,使得要消除的回波信号的样本 可以以双方通话检测器引起适应停止的方式处理在双重通话开始期间拍摄的,初始的潜在干扰的样本不会在滤波器系统中引起明显的发散。

    Stereophonic acoustic echo cancellation using non-linear transformations
    8.
    发明授权
    Stereophonic acoustic echo cancellation using non-linear transformations 失效
    使用非线性变换的立体声回声消除

    公开(公告)号:US5828756A

    公开(公告)日:1998-10-27

    申请号:US747730

    申请日:1996-11-12

    CPC分类号: H04M9/082

    摘要: A method and apparatus for estimating individual impulse responses for a stereophonic communication system, such as a teleconferencing system, which involves selectively reducing the correlation between the individual channel signals of the stereophonic system. Selective reduction of stereophonic source signal correlation advantageously results in the estimation of individual impulse responses of a receiving room of the stereophonic communication system. The selectively reduced-correlation source signals are provided to conventional adaptive filters and the receiving room loudspeakers. Automatic echo cancellation is performed in a conventional fashion, but on the selectively reduced-correlation source signals. Specifically, selective reduction of source signal correlation between two stereophonic channels of a teleconferencing system is achieved by introducing a small non-linearity into each channel in order to reduce the interchannel coherence. In accordance with certain illustrative embodiments of the present invention, each channel signal has added thereto a non-linear function of the channel signal itself, thereby reducing the interchannel coherence while preserving the quality of the signal. In one particular embodiment, the non-linear function comprises the half-wave rectifier.

    摘要翻译: 一种用于估计诸如电话会议系统的立体声通信系统的各个脉冲响应的方法和装置,其涉及选择性地降低立体声系统的各个声道信号之间的相关性。 立体声源信号相关性的选择性降低有利地导致对立体声通信系统的接收室的各个脉冲响应的估计。 选择性的相关源信号被提供给传统的自适应滤波器和接收室扬声器。 以常规方式执行自动回波消除,但是在选择性地减小相关源信号上。 具体来说,电话会议系统的两个立体声道之间的源信号相关性的选择性降低是通过在每个通道中引入小的非线性来实现的,以便减少通道间的相干性。 根据本发明的某些说明性实施例,每个信道信号已经添加了信道信号本身的非线性函数,从而在保持信号质量的同时减少了信道间相干性。 在一个特定实施例中,非线性功能包括半波整流器。

    Method and apparatus for performing double-talk detection in acoustic echo cancellation
    9.
    发明授权
    Method and apparatus for performing double-talk detection in acoustic echo cancellation 有权
    用于在声学回声消除中执行双方通话检测的方法和装置

    公开(公告)号:US06766019B1

    公开(公告)日:2004-07-20

    申请号:US09621103

    申请日:2000-07-21

    IPC分类号: H04M908

    CPC分类号: H04M9/082

    摘要: A method and apparatus for performing double-talk detection in an acoustic echo canceller in which a detection statistic is advantageously computed based on an estimate of a cross-correlation between the far-end signal and the return signal which has been normalized with use of an estimate of a covariance matrix of the far-end signal. The estimate of the cross-correlation between the far-end signal and the return signal may be further normalized with use of either an estimate of a variance of the return signal or an estimate of a covariance matrix of the return signal. In certain illustrative embodiments of the invention, one or more of these quantities may be estimated based on signal samples sampled over a predetermined time window. And in another illustrative embodiment of the present invention, the coefficients of the adaptive filter employed in the acoustic echo canceller itself are advantageously used to compute the detection statistic. These computations may be performed in either the time domain or the frequency domain. The detection statistic so computed is compared with a predetermined threshold, which threshold may be advantageously fixed at a value close to one, in order to determine whether or not double-talk has occurred.

    摘要翻译: 一种用于在声学回声消除器中执行双方通话检测的方法和装置,其中有利地基于远端信号和返回信号之间的互相关的估计来计算检测统计量,该估计已经使用 估计远端信号的协方差矩阵。 可以使用返回信号的方差的估计或返回信号的协方差矩阵的估计来进一步归一化远端信号与返回信号之间的互相关的估计。 在本发明的某些示例性实施例中,可以基于在预定时间窗口上采样的信号样本来估计这些量中的一个或多个。 并且在本发明的另一个说明性实施例中,用于声学回声消除器本身的自适应滤波器的系数有利地用于计算检测统计量。 这些计算可以在时域或频域中执行。 将如此计算的检测统计量与预定阈值进行比较,该阈值可有利地固定在接近于1的值,以便确定是否发生双方通话。

    Robust signed regressor PNLMS method and apparatus for network echo cancellation
    10.
    发明授权
    Robust signed regressor PNLMS method and apparatus for network echo cancellation 有权
    用于网络回波消除的鲁棒签名回归器PNLMS方法和装置

    公开(公告)号:US06611594B1

    公开(公告)日:2003-08-26

    申请号:US09695787

    申请日:2000-10-25

    IPC分类号: H04M100

    CPC分类号: H04B3/234 H03H2021/0063

    摘要: The invention is a method and apparatus for performing echo cancellation utilizing a Proportionate Normalized Least Mean Squares (PNLMS) algorithm having a high convergence rate and low complexity. The invention utilizes a proportional step-size adaptive algorithm which utilizes the signs of the regressor rather than the regressor value itself in the gradient calculation.

    摘要翻译: 本发明是一种使用具有高收敛速度和低复杂度的比例归一化最小均方(PNLMS)算法进行回波消除的方法和装置。 本发明利用比例步长自适应算法,其在梯度计算中利用回归器的符号而不是回归值本身。