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1. 发明授权

US07136437B2 Method and apparatus for receiving digital wireless transmissions using multiple-antenna communication schemes 有权
标题翻译：用于使用多天线通信方案接收数字无线传输的方法和装置
公开(公告)号：US07136437B2
公开(公告)日：2006-11-14
申请号：US10196865
申请日：2002-07-17
申请人： Jacob Benesty , Jingdong Chen , Yiteng Arden Huang
发明人： Jacob Benesty , Jingdong Chen , Yiteng Arden Huang
IPC分类号： H04B7/10
CPC分类号： H04L1/06
摘要： A signal detection technique for multiple-input multiple-output (MIMO) communications systems embodied in a method and apparatus for detecting a plurality of transmitted signals with use of a plurality of receiving antennas. An iterative procedure decodes one of a plurality of transmitted signals at each iteration using an intermediate matrix at each iteration to determine the transmitted signal to be decoded. The intermediate matrix for each successive iteration is advantageously computed in a recursive manner with use of a Schur complement operation performed based on the inverse of a modified version of the intermediate matrix used in the previous iteration.
摘要翻译：一种在用于使用多个接收天线检测多个发射信号的方法和装置中实现的多输入多输出（MIMO）通信系统的信号检测技术。迭代过程在每次迭代中使用中间矩阵在每次迭代中解码多个发送信号中的一个，以确定要解码的发送信号。有利地，使用基于在前一次迭代中使用的中间矩阵的修改版本的反向执行的Schur补码操作，递归地计算每个连续迭代的中间矩阵。

2. 发明授权

US08503653B2 Method and apparatus for active speaker selection using microphone arrays and speaker recognition 有权
公开(公告)号：US08503653B2
公开(公告)日：2013-08-06
申请号：US12074276
申请日：2008-03-03
申请人： Sudhir Raman Ahuja , Jingdong Chen , Yiteng Arden Huang , Dong Liu , Qiru Zhou
发明人： Sudhir Raman Ahuja , Jingdong Chen , Yiteng Arden Huang , Dong Liu , Qiru Zhou
IPC分类号： H04M3/42 , H04M1/00 , H04L12/16 , H04Q11/00
CPC分类号： H04M3/569 , G10L17/00 , G10L2021/02166 , H04M2201/41 , H04M2203/5072
摘要： A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface. The remote participants may then select one of the identified speakers, and the speech signal selection module then selects for transmission the speech signal associated with the selected identified speaker, thereby enabling the participants at the remote location to listen to the selected speaker and neglect the speech from other active speakers.

3. 发明授权

US07945006B2 Data-driven method and apparatus for real-time mixing of multichannel signals in a media server 有权
标题翻译：用于在媒体服务器中实时混合多信道信号的数据驱动方法和装置
公开(公告)号：US07945006B2
公开(公告)日：2011-05-17
申请号：US10875553
申请日：2004-06-24
申请人： Jingdong Chen , Yiteng Arden Huang , Thomas Y Woo
发明人： Jingdong Chen , Yiteng Arden Huang , Thomas Y Woo
IPC分类号： H04B1/10
CPC分类号： H04M7/006 , H04L29/06027 , H04L65/4038 , H04L65/605 , H04M3/56 , H04M3/568
摘要： An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.
摘要翻译：一种用于在IP语音电话会议环境中混合音频信号的装置包括预处理器，混合控制器和混合处理器。预处理器被分为媒体参数估计器和媒体预处理器。媒体参数估计器估计用于控制如何混合不同信道的信号参数，例如信噪比，能级和语音活动（即信号中的语音的存在或不存在）。媒体预处理器采用静音抑制，自动增益控制，降噪等信号处理算法，优化了传入语音流的质量。基于估计的信号参数的函数，混合控制器指定特定的混合策略，并且混合处理器根据控制器提供的策略来混合预处理的语音流。

4. 发明申请

US20090220065A1 Method and apparatus for active speaker selection using microphone arrays and speaker recognition 有权
标题翻译：用于使用麦克风阵列和扬声器识别的主动扬声器选择的方法和装置
公开(公告)号：US20090220065A1
公开(公告)日：2009-09-03
申请号：US12074276
申请日：2008-03-03
申请人： Sudhir Raman Ahuja , Jingdong Chen , Yiteng Arden Huang , Dong Liu , Qiru Zhou
发明人： Sudhir Raman Ahuja , Jingdong Chen , Yiteng Arden Huang , Dong Liu , Qiru Zhou
IPC分类号： H04M3/56
CPC分类号： H04M3/569 , G10L17/00 , G10L2021/02166 , H04M2201/41 , H04M2203/5072
摘要： A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface. The remote participants may then select one of the identified speakers, and the speech signal selection module then selects for transmission the speech signal associated with the selected identified speaker, thereby enabling the participants at the remote location to listen to the selected speaker and neglect the speech from other active speakers.
摘要翻译：用于在电话会议应用中执行主动扬声器选择的方法和装置示例性地包括麦克风阵列模块，扬声器识别系统，用户界面和语音信号选择模块。麦克风阵列模块将来自每个有源扬声器的语音信号与其他有源扬声器的语音信号分离，从而提供多个单独的扬声器的语音信号。扬声器识别系统使用常规扬声器识别/识别技术识别每个当前有效的扬声器。然后将这些身份发送到远程电话会议位置，以通过用户界面向远程参与者显示。然后，远程参与者可以选择所识别的扬声器中的一个，并且语音信号选择模块然后选择用于传输与所选择的所识别的扬声器相关联的语音信号，从而使远程位置的参与者能够听取所选择的说话者并忽略语音从其他有源音箱

5. 发明授权

US09237391B2 Low noise differential microphone arrays 有权
标题翻译：低噪声差分麦克风阵列
公开(公告)号：US09237391B2
公开(公告)日：2016-01-12
申请号：US13816430
申请日：2012-12-04
申请人： Jacob Benesty , Jingdong Chen
发明人： Jacob Benesty , Jingdong Chen
IPC分类号： H04R3/00 , H04R1/08 , H04R3/04 , H04R1/40
CPC分类号： H04R3/04 , H04R1/08 , H04R1/406 , H04R3/005 , H04R2201/403 , H04R2430/03 , H04R2430/21
摘要： A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.
摘要翻译：提供差分麦克风阵列（DMA），其包括用于将声音转换为多个电信号的麦克风传感器的数量（M），以及经配置以在一段时间内在电信号上应用线性约束的最小方差滤波器的处理器窗口以计算多个子带上的电信号的频率响应，并且对每个子带的电信号的频率响应求和，以计算声音的估计频谱。

6. 发明申请

US20160134969A1 LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS 有权
标题翻译：低噪声差分麦克风阵列
公开(公告)号：US20160134969A1
公开(公告)日：2016-05-12
申请号：US14980802
申请日：2015-12-28
申请人： Jingdong Chen , Jacob Benesty
发明人： Jingdong Chen , Jacob Benesty
IPC分类号： H04R3/04 , H04R3/00 , H04R1/40
CPC分类号： H04R3/04 , H04R1/08 , H04R1/406 , H04R3/005 , H04R2201/403 , H04R2430/03 , H04R2430/21
摘要： A differential microphone array includes a number (M) of microphone sensors for converting sound to a number of electrical signals, and a processor, operably coupled to the microphone sensors, to specify a target differential order (N) for the differential microphone array, and wherein M>N+1, specify a steering matrix D comprising N+1 steering vectors, calculate a respective one of a plurality of linearly specify a steering matrix D comprising N+1 steering vectors-constrained minimum variance filters based on the steering matrix, apply the respective one of the plurality of linearly-constrained minimum variance filters to a respective one of the electrical signals to calculate a respective frequency response of the electrical signals, wherein the respective frequency response comprises a plurality of components associated with a plurality of subbands, and sum the frequency responses of the electrical signals with respect to each subband to calculate an estimated frequency spectrum of the sound.
摘要翻译：差分麦克风阵列包括用于将声音转换为多个电信号的麦克风传感器数量（M），以及可操作地耦合到麦克风传感器的处理器，以指定差分麦克风阵列的目标差分顺序（N），以及其中M> N + 1指定包括N + 1个导向矢量的导引矩阵D，计算多个线性地指定基于导引矩阵的N + 1个导向矢量约束最小方差滤波器的导引矩阵D中的相应一个，将所述多个线性约束最小方差滤波器中的相应一个应用到所述电信号中的相应一个，以计算所述电信号的相应频率响应，其中所述相应频率响应包括与多个子带相关联的多个分量，并且相对于每个子带对电信号的频率响应进行求和，以计算t的估计频谱他听起来

7. 发明申请

US20150163577A1 LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS 有权
标题翻译：低噪声差分麦克风阵列
公开(公告)号：US20150163577A1
公开(公告)日：2015-06-11
申请号：US13816430
申请日：2012-12-04
申请人： Jacob Benesty , Jingdong Chen
发明人： Jacob Benesty , Jingdong Chen
IPC分类号： H04R1/08 , H04R3/04
CPC分类号： H04R3/04 , H04R1/08 , H04R1/406 , H04R3/005 , H04R2201/403 , H04R2430/03 , H04R2430/21
摘要： A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.
摘要翻译：提供差分麦克风阵列（DMA），其包括用于将声音转换为多个电信号的麦克风传感器的数量（M），以及经配置以在一段时间内在电信号上应用线性约束的最小方差滤波器的处理器窗口以计算多个子带上的电信号的频率响应，并且对每个子带的电信号的频率响应求和，以计算声音的估计频谱。

8. 发明申请

US20050286664A1 Data-driven method and apparatus for real-time mixing of multichannel signals in a media server 有权
标题翻译：用于在媒体服务器中实时混合多信道信号的数据驱动方法和装置
公开(公告)号：US20050286664A1
公开(公告)日：2005-12-29
申请号：US10875553
申请日：2004-06-24
申请人： Jingdong Chen , Yiteng Huang , Thomas Woo
发明人： Jingdong Chen , Yiteng Huang , Thomas Woo
IPC分类号： H03D1/04 , H04L1/00 , H04L29/06 , H04M3/56 , H04M7/00
CPC分类号： H04M7/006 , H04L29/06027 , H04L65/4038 , H04L65/605 , H04M3/56 , H04M3/568
摘要： An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.
摘要翻译：一种用于在IP语音电话会议环境中混合音频信号的装置包括预处理器，混合控制器和混合处理器。预处理器被分为媒体参数估计器和媒体预处理器。媒体参数估计器估计用于控制如何混合不同信道的信号参数，例如信噪比，能级和语音活动（即信号中的语音的存在或不存在）。媒体预处理器采用静音抑制，自动增益控制，降噪等信号处理算法，优化了传入语音流的质量。基于估计的信号参数的函数，混合控制器指定特定的混合策略，并且混合处理器根据控制器提供的策略来混合预处理的语音流。

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