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。

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

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。

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.