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    Method for determining a time delay for time delay compensation
    1.
    发明授权
    Method for determining a time delay for time delay compensation 有权
    用于确定时间延迟补偿的时间延迟的方法

    公开(公告)号：US08238574B2

    公开(公告)日：2012-08-07

    申请号：US12636160

    申请日：2009-12-11

    申请人： Markus Buck ,  Tobias Wolff ,  Gerhard Schmidt ,  Tim Haulick

    发明人： Markus Buck ,  Tobias Wolff ,  Gerhard Schmidt ,  Tim Haulick

    IPC分类号： H04R3/00

    CPC分类号： G01S3/807 ,  G10L21/02 ,  H04M9/082 ,  H04R1/406 ,  H04R3/005 ,  H04R3/02 ,  H04R2410/01 ,  H04R2430/23 ,  H04R2430/25 ,  H04R2499/13 ,  H04S1/00 ,  H04S3/00

    摘要： The invention provides a computer-implemented method for determining a time delay for time delay compensation of a microphone signal from a microphone array in a beamformer arrangement. For a given time, an instantaneous estimate of a position of a wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source is determined. The computer system then determines whether the instantaneous estimate deviates from a preset estimate of a position of the wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source according to a predetermined criterion. The predetermined criterion comprises a check whether the instantaneous estimate deviates from the preset estimate by at least a predetermined deviation threshold. If the predetermined criterion is fulfilled, the instantaneous estimate for the given time is set by the computer system as the preset estimate, and the computer system determines the time delay for time delay compensation of the microphone signal based on the instantaneous estimate.

    摘要翻译： 本发明提供了一种计算机实现的方法，用于确定来自波束形成器布置中的麦克风阵列的麦克风信号的时间延迟的时间延迟。 对于给定时间，确定所需声源的位置和/或源自所需声源的信号的到达方向的瞬时估计。 计算机系统然后根据预定标准确定瞬时估计是否偏离预期的所需声源的位置的预设估计和/或源自所需声源的信号的到达方向。 预定标准包括检查瞬时估计是否偏离预设估计至少预定的偏差阈值。 如果满足预定标准，则由计算机系统将给定时间的瞬时估计值设置为预设估计，并且计算机系统基于瞬时估计确定麦克风信号的时间延迟补偿的时间延迟。

    System for processing microphone signals to provide an output signal with reduced interference
    2.
    发明授权
    System for processing microphone signals to provide an output signal with reduced interference 有权
    用于处理麦克风信号的系统，以提供具有减小的干扰的输出信号

    公开(公告)号：US08189810B2

    公开(公告)日：2012-05-29

    申请号：US12125298

    申请日：2008-05-22

    申请人： Tobias Wolff ,  Markus Buck

    发明人： Tobias Wolff ,  Markus Buck

    IPC分类号： H04B15/00

    CPC分类号： H04M9/082 ,  G10L2021/02082 ,  G10L2021/02166

    摘要： A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal.

    摘要翻译： 系统减少可能影响通信的噪声或其他外部信号。 设备将来自两个或更多麦克风的声音转换为操作信号。 基于一个或两个信号，波束形成器产生中间信号。 可以通过回波消除器来估计或测量反射或其它不需要的信号。 可以通过处理回波减小信号或通过阻塞矩阵估计来测量或估计干扰。 干扰消除器可以减少可能基于阻塞矩阵和中间信号的输出修改或中断信号的干扰。

    SYSTEM FOR SPEECH SIGNAL ENHANCEMENT IN A NOISY ENVIRONMENT THROUGH CORRECTIVE ADJUSTMENT OF SPECTRAL NOISE POWER DENSITY ESTIMATIONS
    3.
    发明申请
    SYSTEM FOR SPEECH SIGNAL ENHANCEMENT IN A NOISY ENVIRONMENT THROUGH CORRECTIVE ADJUSTMENT OF SPECTRAL NOISE POWER DENSITY ESTIMATIONS 有权
    通过光谱噪声功率密度估计的正确调整，噪声环境中的语音信号增强系统

    公开(公告)号：US20090063143A1

    公开(公告)日：2009-03-05

    申请号：US12202147

    申请日：2008-08-29

    申请人： Gerhard Uwe Schmidt ,  Tobias Wolff ,  Markus Buck

    发明人： Gerhard Uwe Schmidt ,  Tobias Wolff ,  Markus Buck

    IPC分类号： G10L15/20

    CPC分类号： H04R3/00 ,  G10L21/0208 ,  G10L21/0216

    摘要： A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.

    摘要翻译： 系统估计音频信号的频谱噪声功率密度包括频谱噪声功率密度估计单元，校正项处理器和组合处理器。 频谱噪声功率密度估计单元可以提供音频信号的频谱噪声功率密度的第一估计。 校正项处理器可以至少部分地基于实际频谱噪声功率密度的频谱噪声功率密度估计误差来提供时间相关校正项。 可以确定校正项，使得谱噪声功率密度估计误差降低。 组合处理器可以将第一估计与校正项组合以获得可用于后续信号处理以增强音频信号的期望信号分量的频谱噪声功率密度的第二估计。

    Method for determining a signal component for reducing noise in an input signal
    4.
    发明授权
    Method for determining a signal component for reducing noise in an input signal 有权
    用于确定用于降低输入信号中的噪声的信号分量的方法

    公开(公告)号：US08705759B2

    公开(公告)日：2014-04-22

    申请号：US12749136

    申请日：2010-03-29

    申请人： Tobias Wolff ,  Markus Buck ,  Toby Christian Lawin-Ore

    发明人： Tobias Wolff ,  Markus Buck ,  Toby Christian Lawin-Ore

    IPC分类号： H04B3/20 ,  H04B15/00 ,  G10L21/00

    CPC分类号： G10L21/02

    摘要： The invention provides a method for determining a signal component for reducing noise in an input signal, which comprises a noise component, comprising the steps of: estimating the noise component in the input signal, estimating a reverberation component in the noise component, and removing the estimated reverberation component from the estimated noise component to obtain a modified estimate of the noise component.

    摘要翻译： 本发明提供一种确定用于降低输入信号中的噪声的信号分量的方法，包括噪声分量，包括以下步骤：估计输入信号中的噪声分量，估计噪声分量中的混响分量，以及去除 从估计的噪声分量估计混响分量，以获得噪声分量的修正估计。

    Beamforming pre-processing for speaker localization
    5.
    发明授权
    Beamforming pre-processing for speaker localization 有权
    演讲者本地化的波束成形预处理

    公开(公告)号：US08660274B2

    公开(公告)日：2014-02-25

    申请号：US12504333

    申请日：2009-07-16

    申请人： Tobias Wolff ,  Markus Buck ,  Gerhard Schmidt

    发明人： Tobias Wolff ,  Markus Buck ,  Gerhard Schmidt

    IPC分类号： H04R3/00

    CPC分类号： H04R3/005 ,  H04R2430/20 ,  H04R2430/23

    摘要： Embodiments of the present invention relate to methods, systems, and computer program products for signal processing. A first plurality of microphone signals is obtained by a first microphone array. A second plurality of microphone signals is obtained by a second microphone array different from the first microphone array. The first plurality of microphone signals is beamformed by a first beamformer comprising beamforming weights to obtain a first beamformed signal. The second plurality of microphone signals is beamformed by a second beamformer comprising the same beamforming weights as the first beamformer to obtain a second beamformed signal. The beamforming weights are adjusted such that the power density of echo components and/or noise components present in the first and second plurality of microphone signals is substantially reduced.

    摘要翻译： 本发明的实施例涉及用于信号处理的方法，系统和计算机程序产品。 第一麦克风信号由第一麦克风阵列获得。 通过与第一麦克风阵列不同的第二麦克风阵列获得第二多个麦克风信号。 第一组多个麦克风信号由包括波束成形权重的第一波束形成器波束形成，以获得第一波束形成信号。 第二组麦克风信号由包括与第一波束形成器相同的波束形成权重的第二波束形成器波束形成，以获得第二波束形成信号。 调整波束成形权重使得第一和第二多个麦克风信号中存在的回波分量和/或噪声分量的功率密度显着降低。

    Method and device for locating a sound source
    7.
    发明授权
    Method and device for locating a sound source 有权
    用于定位声源的方法和装置

    公开(公告)号：US08194500B2

    公开(公告)日：2012-06-05

    申请号：US12547681

    申请日：2009-08-26

    申请人： Tobias Wolff ,  Markus Buck ,  Gerhard Schmidt ,  Olga González Valbuena ,  Günther Wirsching

    发明人： Tobias Wolff ,  Markus Buck ,  Gerhard Schmidt ,  Olga González Valbuena ,  Günther Wirsching

    IPC分类号： G10L21/02 ,  G01S3/80 ,  H04R3/00

    CPC分类号： G01S3/8083

    摘要： A method of locating a sound source based on sound received at an array of microphones comprises the steps of determining a correlation function of signals provided by microphones of the array and establishing a direction in which the sound source is located based on at least one eigenvector of a matrix having matrix elements which are determined based on the correlation function. The correlation function has first and second frequency components associated with a first and second frequency band, respectively. The first frequency component is determined based on signals from microphones having a first distance, and the second frequency component is determined based on signals from microphones having a second distance different from the first distance.

    摘要翻译： 基于在麦克风阵列处接收到的声音来定位声源的方法包括以下步骤：确定由阵列的麦克风提供的信号的相关函数，并基于至少一个本征向量建立声源所在的方向 具有基于相关函数确定的矩阵元素的矩阵。 相关函数分别具有与第一和第二频带相关联的第一和第二频率分量。 基于来自具有第一距离的麦克风的信号确定第一频率分量，并且基于来自具有不同于第一距离的第二距离的麦克风的信号来确定第二频率分量。

    Method for Determining a Signal Component for Reducing Noise in an Input Signal
    8.
    发明申请
    Method for Determining a Signal Component for Reducing Noise in an Input Signal 有权
    确定用于降低输入信号噪声的信号分量的方法

    公开(公告)号：US20100246844A1

    公开(公告)日：2010-09-30

    申请号：US12749136

    申请日：2010-03-29

    申请人： Tobias Wolff ,  Markus Buck ,  Toby Christian Lawin-Ore

    发明人： Tobias Wolff ,  Markus Buck ,  Toby Christian Lawin-Ore

    IPC分类号： H04B3/20

    CPC分类号： G10L21/02

    摘要： The invention provides a method for determining a signal component for reducing noise in an input signal, which comprises a noise component, comprising the steps of: estimating the noise component in the input signal, estimating a reverberation component in the noise component, and removing the estimated reverberation component from the estimated noise component to obtain a modified estimate of the noise component.

    摘要翻译： 本发明提供一种确定用于降低输入信号中的噪声的信号分量的方法，包括噪声分量，包括以下步骤：估计输入信号中的噪声分量，估计噪声分量中的混响分量，以及去除 从估计的噪声分量估计混响分量，以获得噪声分量的修正估计。

    SYSTEM FOR PROCESSING MICROPHONE SIGNALS TO PROVIDE AN OUTPUT SIGNAL WITH REDUCED INTERFERENCE
    9.
    发明申请
    SYSTEM FOR PROCESSING MICROPHONE SIGNALS TO PROVIDE AN OUTPUT SIGNAL WITH REDUCED INTERFERENCE 有权
    用于处理麦克风信号以减少干扰的输出信号的系统

    公开(公告)号：US20080298602A1

    公开(公告)日：2008-12-04

    申请号：US12125298

    申请日：2008-05-22

    申请人： Tobias Wolff ,  Markus Buck

    发明人： Tobias Wolff ,  Markus Buck

    IPC分类号： H04B3/20

    CPC分类号： H04M9/082 ,  G10L2021/02082 ,  G10L2021/02166

    摘要： A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal.

    摘要翻译： 系统减少可能影响通信的噪声或其他外部信号。 设备将来自两个或更多麦克风的声音转换为操作信号。 基于一个或两个信号，波束形成器产生中间信号。 可以通过回波消除器来估计或测量反射或其它不需要的信号。 可以通过处理回波减小信号或通过阻塞矩阵估计来测量或估计干扰。 干扰消除器可以减少可能基于阻塞矩阵和中间信号的输出修改或中断信号的干扰。

    Low-delay filtering
    10.
    发明授权
    Low-delay filtering 有权
    低延迟滤波

    公开(公告)号：US09036752B2

    公开(公告)日：2015-05-19

    申请号：US14119933

    申请日：2011-05-05

    申请人： Markus Buck ,  Tobias Wolff

    发明人： Markus Buck ,  Tobias Wolff

    IPC分类号： H04B1/00 ,  H04B1/10 ,  H03H17/02 ,  H03H21/00 ,  G10L21/0232

    CPC分类号： H04B1/10 ,  G10L21/0232 ,  H03H17/0213 ,  H03H21/0027

    摘要： A method of frequency-domain filtering is provided that includes a plurality of filters, the plurality of filters including at least one constrained filter(s) W=I, I and at least one unconstrained filter(s) W=1,K− The method includes cascading the W k=i,K unconstrained filter(s). A single constraint window C is applied to the cascaded W=i,K unconstrained filter(s). The W=1,I constrained filter(s) are cascaded with the constrained cascaded Wk=1,K unconstrained filter(s) to form a resulting filter Wll=C(W 1{circle around (x)} . . . {circle around (x)} W){circle around (x)} W . . . W. The frequency domain representation of the single constraint window C may be based, at least in part, on a time domain representation of a single constraint window C that has been circularly shifted such that the frequency domain representation of the constraint window matches a property of the frequency domain representation of the cascaded W=1,K unconstrained filters.

    摘要翻译： 提供了包括多个滤波器的频域滤波的方法，所述多个滤波器包括至少一个约束滤波器W = I，I和至少一个无约束滤波器W = 1，K- 方法包括级联W k = i，K无约束滤波器。 单个约束窗口C被应用于级联的W = i，K无约束滤波器。 W = 1，约束滤波器与受限级联Wk = 1，K无约束滤波器级联，以形成滤波器W11 = C（W 1 {围绕（x）}圆圈...圆圈 around（x）} W）{circle around（x）} W。 。 。 单个约束窗口C的频域表示可以至少部分地基于已被循环移位的单个约束窗口C的时域表示，使得约束窗口的频域表示与属性 的级联W = 1的频域表示，K个无约束滤波器。