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81. 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。

82. 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。

83. 发明申请

US20110019835A1 Speaker Localization 有权
标题翻译：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。

84. 发明申请

US20100246851A1 Method for Determining a Noise Reference Signal for Noise Compensation and/or Noise Reduction 有权
标题翻译：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。

85. 发明申请

US20090192796A1 FILTERING OF BEAMFORMED SPEECH SIGNALS 有权
标题翻译：波束语音信号的滤波
公开(公告)号：US20090192796A1
公开(公告)日：2009-07-30
申请号：US12357258
申请日：2009-01-21
申请人： Markus Buck , Klaus Scheufele
发明人： Markus Buck , Klaus Scheufele
IPC分类号： G10L15/20
CPC分类号： G10L21/0208 , G10L2021/02166
摘要： The invention relates to speech signal processing that detects a speech signal from more than one microphone and obtains microphone signals that are processed by a beamformer to obtain a beamformed signal that is post-filtered signal with a filter that employs adaptable filter weights to obtain an enhanced beamformed signal with the post-filter adapting the filter weights with previously learned filter weights.
摘要翻译：本发明涉及语音信号处理，其检测来自多于一个麦克风的语音信号，并获得由波束形成器处理的麦克风信号，以获得具有滤波后信号的波束形成信号，该滤波器采用适应滤波器权重以获得增强的波束形成的信号与后置滤波器使用先前学习的滤波器权重来适配滤波器权重。

86. 发明申请

US20090117948A1 METHOD FOR DEREVERBERATION OF AN ACOUSTIC SIGNAL 有权
标题翻译：声学信号分析方法
公开(公告)号：US20090117948A1
公开(公告)日：2009-05-07
申请号：US12263227
申请日：2008-10-31
申请人： Markus Buck , Arthur Wolf
发明人： Markus Buck , Arthur Wolf
IPC分类号： H04B1/38 , H04B3/20
CPC分类号： G10L2021/02082
摘要： A method is provided for estimating a reverberation signal component of an acoustic signal detected by a microphone where the acoustic signal is comprised of a direct sound component and a reverberation signal component. A method for dereverberation of an acoustic signal is further provided.
摘要翻译：提供一种用于估计由麦克风检测到的声信号的混响信号分量的方法，其中声信号由直接声音分量和混响信号分量组成。还提供了一种用于声信号的混响的方法。

87. 发明申请

US20080304679A1 SYSTEM FOR PROCESSING AN ACOUSTIC INPUT SIGNAL TO PROVIDE AN OUTPUT SIGNAL WITH REDUCED NOISE 有权
标题翻译：用于处理声音输入信号以减少噪声的输出信号的系统
公开(公告)号：US20080304679A1
公开(公告)日：2008-12-11
申请号：US12118205
申请日：2008-05-09
申请人： Gerhard Uwe Schmidt , Raymond Bruckner , Markus Buck , Ange Tchinda-Pockem , Mohamed Krini
发明人： Gerhard Uwe Schmidt , Raymond Bruckner , Markus Buck , Ange Tchinda-Pockem , Mohamed Krini
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L21/0232 , G10L2021/02168
摘要： An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.
摘要翻译：一种装置处理声输入信号以提供具有降低的噪声的输出信号。该装置基于频率依赖加权函数对输入信号进行加权。频率相关阈值函数从下面界定加权函数。

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