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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。

    Speech signal enhancement using visual information
    5.
    发明授权
    Speech signal enhancement using visual information 有权
    使用视觉信息的语音信号增强

    公开(公告)号：US09293151B2

    公开(公告)日：2016-03-22

    申请号：US14352016

    申请日：2011-10-17

    申请人： Tobias Herbig ,  Tobias Wolff ,  Markus Buck

    发明人： Tobias Herbig ,  Tobias Wolff ,  Markus Buck

    IPC分类号： G10L25/27 ,  G06K9/00 ,  G06T7/00 ,  H04M3/56 ,  H04R3/00 ,  G10L15/20 ,  G10L17/00 ,  H04N7/15 ,  G10L25/78 ,  G10L21/0208

    CPC分类号： G10L25/27 ,  G06K9/00624 ,  G06T7/73 ,  G06T2207/30196 ,  G10L15/20 ,  G10L17/00 ,  G10L25/78 ,  G10L2021/02082 ,  H04M3/568 ,  H04N7/15

    摘要： Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and/or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and/or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone. In addition, orientation of the speaker or the listener, relative to the microphone or the loudspeaker, can also be estimated, and this estimate may be used to adjust frequency-dependent filter weights to compensate for uneven frequency propagation of acoustic signals from a mouth, or to a human ear, about a human head.

    摘要翻译： 视觉信息用于改变或设置除波束形成器之外的音频信号处理器的操作参数。 数码相机拍摄有关包含人类扬声器和/或听众的场景的视觉信息。 分析视觉信息以确定关于房间声学的信息。 可以估计扬声器和麦克风之间的距离，并且该距离估计可以用于调整系统的整体增益。 可以估计扬声器，收听者，麦克风，扬声器和/或声音反射表面之间的距离和位置。 这些估计可以用于估计房间内的混响，并基于估计达到麦克风的直接到间接（混响）声能的估计比例来调节反混响滤波器的积极性。 此外，还可以估计扬声器或收听者相对于麦克风或扬声器的取向，并且该估计可用于调整频率依赖的滤波器权重以补偿来自口的声信号的不均匀频率传播， 或人的耳朵，关于人的头部。

    Method for determining a set of filter coefficients for an acoustic echo compensator
    6.
    发明授权
    Method for determining a set of filter coefficients for an acoustic echo compensator 有权
    用于确定声学回声补偿器的滤波器系数集合的方法

    公开(公告)号：US08787560B2

    公开(公告)日：2014-07-22

    申请号：US12708172

    申请日：2010-02-18

    申请人： Markus Buck ,  Gerhard Schmidt ,  Tobias Wolff

    发明人： Markus Buck ,  Gerhard Schmidt ,  Tobias Wolff

    IPC分类号： H04M9/08 ,  G01S15/00 ,  H04R3/00

    CPC分类号： H04R3/002 ,  G10L2021/02082 ,  H04M9/082 ,  H04R1/406 ,  H04R3/005 ,  H04R2430/20

    摘要： The invention provides a method for determining a set of filter coefficients for an acoustic echo compensator in a beamformer arrangement. The acoustic echo compensator compensates for echoes within the beamformed signal. A plurality of sets of filter coefficients for the acoustic echo compensator is provided. Each set of filter coefficients corresponds to one of a predetermined number of steering directions of the beamformer arrangement. The predetermined number of steering directions is equal to or greater than the number of microphones in the microphone array. For a current steering direction, a current set of filter coefficients for the acoustic echo compensator is determined based on the provided sets of filter coefficients.

    摘要翻译： 本发明提供一种用于确定波束形成器布置中的声学回波补偿器的滤波器系数集合的方法。 声回波补偿器补偿波束形成信号内的回波。 提供了用于声学回声补偿器的多组滤波器系数。 每组滤波器系数对应于波束形成器装置的预定数量的转向方向之一。 预定数量的转向方向等于或大于麦克风阵列中的麦克风的数量。 对于当前的转向方向，基于所提供的滤波器系数集合来确定用于声学回声补偿器的当前一组滤波器系数。

    Speaker Localization
    8.
    发明申请
    Speaker Localization 有权
    演讲者本地化

    公开(公告)号：US20110019835A1

    公开(公告)日：2011-01-27

    申请号：US12742907

    申请日：2008-11-17

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

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

    IPC分类号： G10K11/16 ,  H04R3/00

    CPC分类号： H04R29/00 ,  G10L21/0272 ,  G10L2021/02166 ,  H04R3/005

    摘要： The present invention relates to a method for localizing a sound source, in particular, a human speaker, comprising detecting sound generated by the sound source by means of a microphone array comprising more than two microphones and obtaining microphone signals, one for each of the microphones, selecting from the microphone signals a pair of microphone signals for a predetermined frequency range based on the distance of the microphones to each other and estimating the angle of the incidence of the sound on the microphone array based on the selected pair of microphone signals.

    摘要翻译： 本发明涉及一种用于定位声源，特别是人类扬声器的方法，包括通过包括两个以上麦克风的麦克风阵列检测声源产生的声音并获得麦克风信号，每个麦克风 基于麦克风彼此的距离，基于所选择的一对麦克风信号估计麦克风阵列上的声音的入射角度，从麦克风选择一个麦克风信号达到预定频率范围。

    Method for Determining a Noise Reference Signal for Noise Compensation and/or Noise Reduction
    9.
    发明申请
    Method for Determining a Noise Reference Signal for Noise Compensation and/or Noise Reduction 有权
    确定用于噪声补偿和/或降噪的噪声参考信号的方法

    公开(公告)号：US20100246851A1

    公开(公告)日：2010-09-30

    申请号：US12749066

    申请日：2010-03-29

    申请人： Markus Buck ,  Tobias Wolff ,  Toby Christian Lawin-Ore ,  Samuel Ngouoko Mboungueng ,  Gerhard Schmidt

    发明人： Markus Buck ,  Tobias Wolff ,  Toby Christian Lawin-Ore ,  Samuel Ngouoko Mboungueng ,  Gerhard Schmidt

    IPC分类号： H04B15/00

    CPC分类号： G10K11/002 ,  G10L21/0208 ,  G10L2021/02165 ,  G10L2021/02166 ,  H04R3/005 ,  H04R2499/13

    摘要： The invention provides a method for determining a noise reference signal for noise compensation and/or noise reduction. A first audio signal on a first signal path and a second audio signal on a second signal path are received. The first audio signal is filtered using a first adaptive filter to obtain a first filtered audio signal. The second audio signal is filtered using a second adaptive filter to obtain a second filtered audio signal. The first and the second filtered audio signal are combined to obtain the noise reference signal. The first and the second adaptive filter are adapted such as to minimize a wanted signal component in the noise reference signal.

    摘要翻译： 本发明提供了一种用于确定用于噪声补偿和/或降噪的噪声参考信号的方法。 接收第一信号路径上的第一音频信号和第二信号路径上的第二音频信号。 使用第一自适应滤波器对第一音频信号进行滤波，以获得第一滤波音频信号。 使用第二自适应滤波器对第二音频信号进行滤波，以获得第二滤波音频信号。 第一和第二滤波音频信号被组合以获得噪声参考信号。 第一和第二自适应滤波器适于使噪声参考信号中的有用信号分量最小化。