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81. 发明申请

US20100150375A1 Determination of the Coherence of Audio Signals 有权
标题翻译：确定音频信号的一致性
公开(公告)号：US20100150375A1
公开(公告)日：2010-06-17
申请号：US12636432
申请日：2009-12-11
申请人： Markus Buck , Timo Matheja
发明人： Markus Buck , Timo Matheja
IPC分类号： H04B15/00
CPC分类号： G10L25/78 , G10L2021/02165
摘要： Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals.
摘要翻译：本发明的实施例公开了用于估计信号一致性的计算机实现的方法，系统和计算机程序产品。首先，由第一麦克风检测由声源产生的声音以获得第一麦克风信号，并由第二麦克风检测第二麦克风信号。第一麦克风信号被第一自适应有限脉冲响应滤波器滤波以获得第一滤波信号。第二麦克风信号被第二自适应有限脉冲响应滤波器滤波，以获得第二滤波信号。基于经滤波的信号确定第一滤波信号和第二滤波信号的相干性。第一麦克风信号和第二麦克风信号被滤波，以便在声音传输到第一麦克风的声音传递功能与声音从声源传输到第二麦克风之间的差异被补偿在第一和第二滤波信号。

82. 发明申请

US20090067642A1 NOISE REDUCTION THROUGH SPATIAL SELECTIVITY AND FILTERING 有权
标题翻译：通过空间选择性和过滤减少噪音
公开(公告)号：US20090067642A1
公开(公告)日：2009-03-12
申请号：US12189545
申请日：2008-08-11
申请人： Markus Buck , Tobias Wolff
发明人： Markus Buck , Tobias Wolff
IPC分类号： H04B15/00
CPC分类号： H04R3/005 , H04R2430/25
摘要： A signal processor uses input devices to detect speech or aural signals. Through a programmable set of weights and/or time delays (or phasing) the output of the input devices may be processed to yield a combined signal. The noise contributions of some or each of the outputs of the input devices may be estimated by a circuit element or a controller that processes the outputs of the respective input devices to yield power densities. A short-term measure or estimate of the noise contribution of the respective outputs of the input devices may be obtained by processing the power densities of some or each of the outputs of the respective input devices. Based on the short-term measure or estimate, the noise contribution of the combined signal may be estimated to enhance the combined signal when processed further. An enhancement device or post-filter may reduce noise more effectively and yield robust speech based on the estimated noise contribution of the combined signal.
摘要翻译：信号处理器使用输入设备来检测语音或听觉信号。通过一组可编程的权重和/或时间延迟（或定相），可以处理输入设备的输出以产生组合信号。可以由处理各个输入装置的输出以产生功率密度的电路元件或控制器估计输入装置的一些或每个输出的噪声贡献。可以通过处理各个输入设备的一些或每个输出的功率密度来获得输入设备的各个输出的噪声贡献的短期测量或估计。基于短期测量或估计，可以估计组合信号的噪声贡献，以在进一步处理时增强组合信号。增强装置或后置滤波器可以更有效地降低噪声，并且基于组合信号的估计的噪声贡献产生鲁棒的语音。

83. 发明授权

US09888316B2 System and method for identifying suboptimal microphone performance 有权
公开(公告)号：US09888316B2
公开(公告)日：2018-02-06
申请号：US14778643
申请日：2013-03-21
申请人： NUANCE COMMUNICATIONS, INC. , Timo Matheja , Markus Buck
发明人： Timo Matheja , Markus Buck
IPC分类号： H04R3/00 , G06F3/16 , H04M1/60 , H04M1/03
CPC分类号： H04R3/00 , G06F3/165 , H04M1/03 , H04M1/6008 , H04R3/005 , H04R2430/03 , H04R2430/23 , H04R2499/11
摘要： Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.

84. 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。

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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