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1. 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.

2. 发明申请

US20120184209A1 AREA TRACKING SYSTEMS AND METHODS OF TRACKING ELECTRONIC DEVICES 有权
标题翻译：区域跟踪系统和跟踪电子设备的方法
公开(公告)号：US20120184209A1
公开(公告)日：2012-07-19
申请号：US13006840
申请日：2011-01-14
申请人： Michael Wengrovitz , Tom Janiszewski , Qiru Zhou
发明人： Michael Wengrovitz , Tom Janiszewski , Qiru Zhou
IPC分类号： G06F15/173 , H04B7/00
CPC分类号： H04L67/125 , H04L67/12 , H04L69/28 , H04W4/02 , H04W4/029 , H04W4/04 , H04W4/20
摘要： Area tracking systems including a plurality of short range sensors that are communicatively connected to a server and methods of tracking electronic devices. The electronic devices are tracked by polling for the electronic devices, the polling including transmission of electronic addresses corresponding to the electronic devices and determining whether a reply to the poling is received. The electronic addresses are each stored on one of a plurality of lists. Each of the lists may be polled at a different frequency and may include a different number of electronic device addresses.
摘要翻译：区域跟踪系统包括通信地连接到服务器的多个短距离传感器和跟踪电子设备的方法。电子设备通过轮询电子设备进行跟踪，轮询包括与电子设备相对应的电子地址的传输，并确定是否接收到对极化的回复。电子地址各自存储在多个列表中的一个上。可以以不同的频率轮询每个列表，并且可以包括不同数量的电子设备地址。

3. 发明授权

US07701886B2 Packet loss concealment based on statistical n-gram predictive models for use in voice-over-IP speech transmission 有权
标题翻译：基于用于语音IP语音传输的统计n-gram预测模型的分组丢失隐藏
公开(公告)号：US07701886B2
公开(公告)日：2010-04-20
申请号：US10856728
申请日：2004-05-28
申请人： Minkyu Lee , Qiru Zhou , Imed Zitouni
发明人： Minkyu Lee , Qiru Zhou , Imed Zitouni
IPC分类号： H04Q11/00
CPC分类号： G10L19/04 , G10L19/005 , H04L65/80 , H04Q2213/13034 , H04Q2213/13296 , H04Q2213/13389
摘要： A method for performing packet loss concealment of lost packets in Voice over IP (Internet Protocol) speech transmission. Statistical n-gram models are initially created with use of a training speech corpus, and then, packets lost during transmission are advantageously replaced based on these models. In particular, the existence of statistical patterns in successive voice over IP (VoIP) packets is advantageously exploited by first using conventional vector quantization (VQ) techniques to quantize the parameter data for each packet with use of a corresponding VQ index, and then determining statistical correlations between consecutive sequences of such VQ indices representative of the corresponding sequences of n packets. The statistic n-gram predictive models so created are then used to predict parameter data for use in representing lost data packets.
摘要翻译：一种用于在IP语音（因特网协议）语音传输中执行丢包隐藏丢失分组的方法。最初使用训练语音语料库创建统计n-gram模型，然后基于这些模型有利地替换传输期间丢失的分组。特别地，通过首先使用常规矢量量化（VQ）技术来利用相应的VQ索引对每个分组的参数数据进行量化，有利地利用连续IP语音（VoIP）分组中的统计模式的存在，然后确定统计这种VQ索引的连续序列之间的相关性表示n个分组的相应序列。然后，如此创建的统计量n-gram预测模型用于预测用于表示丢失数据分组的参数数据。

4. 发明申请

US20110107238A1 Network-Based Collaborated Telestration on Video, Images or Other Shared Visual Content 审中-公开
标题翻译：基于网络的协作式视频，图像或其他共享视觉内容的Telestration
公开(公告)号：US20110107238A1
公开(公告)日：2011-05-05
申请号：US12608068
申请日：2009-10-29
申请人： Dong Liu , Qiru Zhou
发明人： Dong Liu , Qiru Zhou
IPC分类号： G06F15/16 , G06F3/048
CPC分类号： H04N7/15 , G06Q10/101 , G06Q50/20 , G06Q50/22 , H04L12/1831
摘要： A telestration system comprises a telestration server configured to communicate with an arbitrary number of telestration clients over a network. The telestration server is further configured to receive telestration input signals from respective ones of the clients and to send telestration output signals to the respective clients, with the telestration output signal sent to a given one of the clients comprising telestration information derived from the telestration input signal received from at least one other one of the clients. Each of the clients with the support of the telestration server can generate a combined telestration overlay for presentation with associated visual content shared between the clients.
摘要翻译：远程审计系统包括被配置为通过网络与任意数量的远程审计客户端通信的远程监控服务器。远程服务器还被配置为从相应的客户端接收远程鉴别输入信号，并且向远端区域输出信号发送远程信号输出信号到各个客户端，其中远程输出信号被发送给给定的一个客户端，包括从远程输入信号从至少另一个客户端收到。具有远程服务器支持的每个客户端都可以生成一个组合的远程复用重叠，用于呈现客户端之间共享的相关视觉内容。

5. 发明授权

US07149690B2 Method and apparatus for interactive language instruction 有权
标题翻译：互动语言教学的方法和装置
公开(公告)号：US07149690B2
公开(公告)日：2006-12-12
申请号：US09392844
申请日：1999-09-09
申请人： Katherine Grace August , Nadine Blackwood , Qi P. Li , Michelle McNerney , Chi-Lin Shih , Arun Chandrasekaran Surendran , Jialin Zhong , Qiru Zhou
发明人： Katherine Grace August , Nadine Blackwood , Qi P. Li , Michelle McNerney , Chi-Lin Shih , Arun Chandrasekaran Surendran , Jialin Zhong , Qiru Zhou
IPC分类号： G09B5/06 , G09B17/00 , G10L15/22
CPC分类号： G09B19/06 , G09B5/04 , G09B19/04 , G10L13/00 , G10L15/06
摘要： A method and apparatus for interactive language instruction is provided that displays text files for processing, provide key features and functions for interactive learning, displays facial animation, and provides a workspace for language building functions. The system includes a stored set of language rules as part of the text-to-speech sub-system, as well as another stored set of rules as applied to the process of learning a language. The method implemented by the system includes digitally converting text to audible speech, providing the audible speech to a user or student (with the aid of an animated image in selected circumstances), prompting the student to replicate the audible speech, comparing the student's replication with the audible speech provided by the system, and providing feedback and reinforcement to the student by, for example, selectively recording or playing back the audible speech and the student's replication.
摘要翻译：提供了一种用于交互式语言指令的方法和装置，其显示用于处理的文本文件，提供用于交互式学习的关键特征和功能，显示面部动画，并为语言构建功能提供工作空间。该系统包括作为文本到语音子系统的一部分的存储的一组语言规则，以及应用于学习语言的过程的另一组存储的规则。该系统实施的方法包括将文本数字转换为可听话音，向用户或学生提供可听见的语音（借助于所选情况下的动画图像），提示学生复制可听见的语音，将学生的复制与由系统提供的可听话音，并通过例如选择性地录制或播放可听见的语音和学生的复制来向学生提供反馈和加强。

6. 发明授权

US06782363B2 Method and apparatus for performing real-time endpoint detection in automatic speech recognition 有权
标题翻译：用于在自动语音识别中执行实时端点检测的方法和装置
公开(公告)号：US06782363B2
公开(公告)日：2004-08-24
申请号：US09848897
申请日：2001-05-04
申请人： Chin-Hui Lee , Qi P. Li , Jinsong Zheng , Qiru Zhou
发明人： Chin-Hui Lee , Qi P. Li , Jinsong Zheng , Qiru Zhou
IPC分类号： G10L1500
CPC分类号： G10L25/87 , G10L25/24
摘要： A method and apparatus for performing real-time endpoint detection for use in automatic speech recognition. A filter is applied to the input speech signal and the filter output is then evaluated with use of a state transition diagram (i.e., a finite state machine). The filter is advantageously designed in light of several criteria in order to increase the accuracy and robustness of detection. The state transition diagram advantageously has three states. The endpoints which are detected may then be advantageously applied to the problem of energy normalization of the speech portion of the signal.
摘要翻译：一种用于执行实时端点检测以用于自动语音识别的方法和装置。将滤波器应用于输入语音信号，然后使用状态转换图（即，有限状态机）来评估滤波器输出。根据若干标准有利地设计过滤器，以便提高检测的精度和鲁棒性。状态转换图有利地具有三种状态。所检测到的端点可以有利地应用于信号的语音部分的能量归一化的问题。

7. 发明授权

US08867996B2 Area tracking systems and methods of tracking electronic devices 有权
标题翻译：跟踪电子设备的区域跟踪系统和方法
公开(公告)号：US08867996B2
公开(公告)日：2014-10-21
申请号：US13006840
申请日：2011-01-14
申请人： Michael Wengrovitz , Tom Janiszewski , Qiru Zhou
发明人： Michael Wengrovitz , Tom Janiszewski , Qiru Zhou
IPC分类号： H04B7/00 , H04L29/08 , H04W4/02 , H04W4/20 , H04W4/04 , H04L29/06
CPC分类号： H04L67/125 , H04L67/12 , H04L69/28 , H04W4/02 , H04W4/029 , H04W4/04 , H04W4/20
摘要： Area tracking systems including a plurality of short range sensors that are communicatively connected to a server and methods of tracking electronic devices. The electronic devices are tracked by polling for the electronic devices, the polling including transmission of electronic addresses corresponding to the electronic devices and determining whether a reply to the poling is received. The electronic addresses are each stored on one of a plurality of lists. Each of the lists may be polled at a different frequency and may include a different number of electronic device addresses.
摘要翻译：区域跟踪系统包括通信地连接到服务器的多个短距离传感器和跟踪电子设备的方法。电子设备通过轮询电子设备进行跟踪，轮询包括与电子设备相对应的电子地址的传输，并确定是否接收到对极化的回复。电子地址各自存储在多个列表中的一个上。可以以不同的频率轮询每个列表，并且可以包括不同数量的电子设备地址。

8. 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

9. 发明申请

US20050276235A1 Packet loss concealment based on statistical n-gram predictive models for use in voice-over-IP speech transmission 有权
标题翻译：基于用于语音IP语音传输的统计n-gram预测模型的分组丢失隐藏
公开(公告)号：US20050276235A1
公开(公告)日：2005-12-15
申请号：US10856728
申请日：2004-05-28
申请人： Minkyu Lee , Qiru Zhou , Imed Zitouni
发明人： Minkyu Lee , Qiru Zhou , Imed Zitouni
IPC分类号： G10L19/00 , G10L19/04 , H04M7/00 , H04Q11/00
CPC分类号： G10L19/04 , G10L19/005 , H04L65/80 , H04Q2213/13034 , H04Q2213/13296 , H04Q2213/13389
摘要： A method for performing packet loss concealment of lost packets in Voice over IP (Internet Protocol) speech transmission. Statistical n-gram models are initially created with use of a training speech corpus, and then, packets lost during transmission are advantageously replaced based on these models. In particular, the existence of statistical patterns in successive voice over IP (VoIP) packets is advantageously exploited by first using conventional vector quantization (VQ) techniques to quantize the parameter data for each packet with use of a corresponding VQ index, and then determining statistical correlations between consecutive sequences of such VQ indices representative of the corresponding sequences of n packets. The statistic n-gram predictive models so created are then used to predict parameter data for use in representing lost data packets.
摘要翻译：一种用于在IP语音（因特网协议）语音传输中执行丢包隐藏丢失分组的方法。最初使用训练语音语料库创建统计n-gram模型，然后基于这些模型有利地替换传输期间丢失的分组。特别地，通过首先使用常规矢量量化（VQ）技术来利用相应的VQ索引对每个分组的参数数据进行量化，有利地利用连续IP语音（VoIP）分组中的统计模式的存在，然后确定统计这种VQ索引的连续序列之间的相关性表示n个分组的相应序列。然后，如此创建的统计量n-gram预测模型用于预测用于表示丢失数据分组的参数数据。

10. 发明授权

US06336090B1 Automatic speech/speaker recognition over digital wireless channels 有权
标题翻译：数字无线频道上的自动语音/扬声器识别
公开(公告)号：US06336090B1
公开(公告)日：2002-01-01
申请号：US09201082
申请日：1998-11-30
申请人： Wu Chou , Michael Charles Recchione , Qiru Zhou
发明人： Wu Chou , Michael Charles Recchione , Qiru Zhou
IPC分类号： G10L1502
CPC分类号： G10L15/30 , G10L15/02 , G10L15/20
摘要： Automatic Speech Recognition (ASR) is achieved in wireless communications systems in which reliable ASR feature vector sequences are derived at a base station directly from digitally transmitted speech coder parameters, with no additional processing or signal modification required at the originating handset. No secondary channel need be provided for the transmission of ASR feature vectors. In operating on received speech coder parameters prior to conversion to a voice signal the present system and methods avoid the lossy conversion process and associated voice distortion. Since the received voice parameters are error protected during transmission they are received with greater accuracy. All, or a subset, of speech coding parameters, including, in appropriate cases, spectral envelope parameters, reflection coefficients, LSPs, LSFs, LPCs, LPCCs, and weighted LPCCs may be processed at a receiving base station or forwarded to another location for processing.
摘要翻译：自动语音识别（ASR）在无线通信系统中实现，其中可靠的ASR特征向量序列直接从数字传输的语音编码器参数在基站导出，而在起始手持机上不需要额外的处理或信号修改。不需要为ASR特征向量的传输提供二级信道。在转换成语音信号之前对接收到的语音编码器参数进行操作，本系统和方法避免了有损转换过程和相关的语音失真。由于接收到的语音参数在传输期间受到错误保护，所以它们以更高的精度被接收。语音编码参数的所有或子集包括在适当的情况下包括频谱包络参数，反射系数，LSP，LSF，LPC，LPCC和加权LPCC可以在接收基站处理或转发到另一个位置进行处理。

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