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1. 发明授权

US08238574B2 Method for determining a time delay for time delay compensation 有权
标题翻译：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。

2. 发明授权

US08189810B2 System for processing microphone signals to provide an output signal with reduced interference 有权
标题翻译：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。

3. 发明申请

US20090063143A1 SYSTEM FOR SPEECH SIGNAL ENHANCEMENT IN A NOISY ENVIRONMENT THROUGH CORRECTIVE ADJUSTMENT OF SPECTRAL NOISE POWER DENSITY ESTIMATIONS 有权
标题翻译：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。

4. 发明授权

US09036752B2 Low-delay filtering 有权
标题翻译：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。

5. 发明授权

US08364479B2 System for speech signal enhancement in a noisy environment through corrective adjustment of spectral noise power density estimations 有权
标题翻译：通过频谱噪声功率密度估计的校正调整，在噪声环境中进行语音信号增强的系统
公开(公告)号：US08364479B2
公开(公告)日：2013-01-29
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： G10L21/02
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。

6. 发明申请

US20100215184A1 Method for Determining a Set of Filter Coefficients for an Acoustic Echo Compensator 有权
标题翻译：用于确定声学回波补偿器的滤波器系数集的方法
公开(公告)号：US20100215184A1
公开(公告)日：2010-08-26
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： H04B3/20
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。

7. 发明授权

US09338549B2 Acoustic localization of a speaker 有权
标题翻译：扬声器的声学定位
公开(公告)号：US09338549B2
公开(公告)日：2016-05-10
申请号：US13478941
申请日：2012-05-23
申请人： Tim Haulick , Gerhard Uwe Schmidt , Markus Buck , Tobias Wolff
发明人： Tim Haulick , Gerhard Uwe Schmidt , Markus Buck , Tobias Wolff
IPC分类号： G01S3/80 , H04R3/00 , H04R1/40 , G01S3/808 , G01S5/30 , G01S15/00 , G01S7/52 , G01S15/42 , G01S15/87 , H04S7/00 , G01S13/00 , G01S7/292 , H04R29/00 , H04M1/60 , G01S13/42 , G01S5/02 , G01S15/06 , G01S3/802 , G01S7/523 , H04B7/08 , G01S3/04 , G01S5/18 , G10K11/34 , G01S3/00 , G10K15/08
CPC分类号： H04R3/005 , G01S3/00 , G01S3/043 , G01S3/802 , G01S3/808 , G01S3/8083 , G01S5/0252 , G01S5/0273 , G01S5/18 , G01S5/30 , G01S7/292 , G01S7/52003 , G01S7/523 , G01S13/003 , G01S13/42 , G01S15/00 , G01S15/003 , G01S15/06 , G01S15/42 , G01S15/876 , G10K11/34 , G10K15/08 , H04B7/086 , H04M1/6075 , H04R1/40 , H04R1/406 , H04R29/00 , H04R29/001 , H04R2201/401 , H04R2201/403 , H04R2203/12 , H04R2410/01 , H04R2430/20 , H04R2499/11 , H04S7/00 , H04S7/30 , H04S7/301 , H04S7/302 , H04S7/303 , H04S7/305
摘要： A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.
摘要翻译：系统将扬声器定位在包含扬声器和麦克风阵列的房间中。扬声器传输部分由扬声器反射的声音。麦克风阵列基于麦克风信号的特性来检测反射的声音并将声音转换成麦克风阵列，扬声器距离麦克风阵列或两者的距离。

8. 发明授权

US09293151B2 Speech signal enhancement using visual information 有权
标题翻译：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。

9. 发明授权

US08787560B2 Method for determining a set of filter coefficients for an acoustic echo compensator 有权
标题翻译：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。

10. 发明申请

US20120294118A1 Acoustic Localization of a Speaker 有权
标题翻译：演讲者声学定位
公开(公告)号：US20120294118A1
公开(公告)日：2012-11-22
申请号：US13478941
申请日：2012-05-23
申请人： Tim Haulick , Gerhard Uwe Schmidt , Markus Buck , Tobias Wolff
发明人： Tim Haulick , Gerhard Uwe Schmidt , Markus Buck , Tobias Wolff
IPC分类号： G01S3/80
CPC分类号： H04R3/005 , G01S3/00 , G01S3/043 , G01S3/802 , G01S3/808 , G01S3/8083 , G01S5/0252 , G01S5/0273 , G01S5/18 , G01S5/30 , G01S7/292 , G01S7/52003 , G01S7/523 , G01S13/003 , G01S13/42 , G01S15/00 , G01S15/003 , G01S15/06 , G01S15/42 , G01S15/876 , G10K11/34 , G10K15/08 , H04B7/086 , H04M1/6075 , H04R1/40 , H04R1/406 , H04R29/00 , H04R29/001 , H04R2201/401 , H04R2201/403 , H04R2203/12 , H04R2410/01 , H04R2430/20 , H04R2499/11 , H04S7/00 , H04S7/30 , H04S7/301 , H04S7/302 , H04S7/303 , H04S7/305
摘要： A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.
摘要翻译：系统将扬声器定位在包含扬声器和麦克风阵列的房间中。扬声器传输部分由扬声器反射的声音。麦克风阵列基于麦克风信号的特性来检测反射的声音并将声音转换成麦克风阵列，扬声器距离麦克风阵列或两者的距离。

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