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1. 发明授权

US08509396B2 Automatic creation of complex conversational natural language call routing system for call centers 有权
标题翻译：自动创建呼叫中心的复杂对话自然语言呼叫路由系统
公开(公告)号：US08509396B2
公开(公告)日：2013-08-13
申请号：US12566369
申请日：2009-09-24
申请人： Ea-Ee Jan , Hong-Kwang Jeff Kuo , David M. Lubensky
发明人： Ea-Ee Jan , Hong-Kwang Jeff Kuo , David M. Lubensky
IPC分类号： H04M1/64
CPC分类号： G06F17/2785
摘要： A call routing system is created by receiving a set of initial target classes and a corresponding set of topic descriptions. Non-overlapping semantic tokens in the set of topic descriptions are identified. A set of clear target classes from the non-overlapping semantic tokens and the initial target classes is identified. Overlapping semantic tokens from the set of topic descriptions are identified. A set of vague classes is identified from the overlapping semantic tokens and the initial target classes. A set of disambiguation dialogues and a set of grammar prompts is generated according to the overlapping and non-overlapping semantic tokens. The call routing system is then created based on the set of clear target classes, the set of vague target classes, and the set of disambiguation dialogues.
摘要翻译：通过接收一组初始目标类别和一组相应的主题描述来创建呼叫路由系统。识别主题描述集中的非重叠语义令牌。识别来自非重叠语义令牌和初始目标类的一组清晰的目标类。确定主题描述集中的重叠语义令牌。从重叠的语义令牌和初始目标类中识别出一组模糊类。根据重叠和非重叠语义令牌生成一组消歧对话和一组语法提示。然后，基于一组明确的目标类，一组模糊的目标类和一组消歧对话来创建呼叫路由系统。

2. 发明申请

US20110069822A1 AUTOMATIC CREATION OF COMPLEX CONVERSATIONAL NATURAL LANGUAGE CALL ROUTING SYSTEM FOR CALL CENTERS 有权
标题翻译：自动创建呼叫中心的复杂对讲自然语言呼叫路由系统
公开(公告)号：US20110069822A1
公开(公告)日：2011-03-24
申请号：US12566369
申请日：2009-09-24
申请人： Ea-Ee Jan , Hong-Kwang Jeff Kuo , David M. Lubensky
发明人： Ea-Ee Jan , Hong-Kwang Jeff Kuo , David M. Lubensky
IPC分类号： H04M1/64 , G06F17/27 , G06F17/00
CPC分类号： G06F17/2785
摘要： A call routing system is created by receiving a set of initial target classes and a corresponding set of topic descriptions. Non-overlapping semantic tokens in the set of topic descriptions are identified. A set of clear target classes from the non-overlapping semantic tokens and the initial target classes is identified. Overlapping semantic tokens from the set of topic descriptions are identified. A set of vague classes is identified from the overlapping semantic tokens and the initial target classes. A set of disambiguation dialogues and a set of grammar prompts is generated according to the overlapping and non-overlapping semantic tokens. The call routing system is then created based on the set of clear target classes, the set of vague target classes, and the set of disambiguation dialogues.
摘要翻译：通过接收一组初始目标类别和一组相应的主题描述来创建呼叫路由系统。识别主题描述集中的非重叠语义令牌。识别来自非重叠语义令牌和初始目标类的一组清晰的目标类。确定主题描述集中的重叠语义令牌。从重叠的语义令牌和初始目标类中识别出一组模糊类。根据重叠和非重叠语义令牌生成一组消歧对话和一组语法提示。然后，基于一组明确的目标类，一组模糊的目标类和一组消歧对话来创建呼叫路由系统。

3. 发明授权

US07558734B1 Using web FAQ data for creating self-service speech applications 有权
标题翻译：使用Web常见问题数据创建自助语音应用程序
公开(公告)号：US07558734B1
公开(公告)日：2009-07-07
申请号：US12271912
申请日：2008-11-16
申请人： Osamuyimen T. Stewart , David M. Lubensky , Ea-Ee Jan , Xiang Li
发明人： Osamuyimen T. Stewart , David M. Lubensky , Ea-Ee Jan , Xiang Li
IPC分类号： G10L21/00
CPC分类号： G06F17/30861 , G06F17/2785 , G06F17/30684 , G10L15/26 , H04M3/4938
摘要： In one example, this invention presents a method of providing the same self-service content that is available on the web interface to users contacting by telephone, knowing that the web and telephone are fundamentally different user interfaces. In one embodiment, this patent seeks to protect the general idea of how to playback web data in real-time to the user over the speech interface. For this purpose, a method is presented comprising of the general steps through which the web data is initially sent to an automatic transformation module. Then, that transformation module refines or re-structures the web data to make it suitable for the speech interface. The algorithm in the module is predicated on the user interface principles of cognitive complexity and limitations on short term memory based on which FAQ types are classified into one of the following four classes: simple, medium, complex, and complex-complex.
摘要翻译：在一个示例中，本发明提供了一种方法，即知道网络和电话是根本上不同的用户界面的，通过电话向网络接口提供与用户联系的相同的自助服务内容。在一个实施例中，该专利旨在保护如何通过语音界面实时地向用户播放web数据的一般思想。为此，提出了一种方法，其包括最初将web数据发送到自动变换模块的一般步骤。然后，该转换模块对网络数据进行优化或重构，使其适合于语音界面。模块中的算法基于用户界面的认知复杂性原理和短期记忆的限制，基于哪些常见问题类型分为以下四个类别之一：简单，中等，复杂和复杂。

4. 发明授权

US07680661B2 Method and system for improved speech recognition 有权
标题翻译：改进语音识别的方法和系统
公开(公告)号：US07680661B2
公开(公告)日：2010-03-16
申请号：US12120316
申请日：2008-05-14
申请人： Raymond L. Co , Ea-Ee Jan , David M. Lubensky
发明人： Raymond L. Co , Ea-Ee Jan , David M. Lubensky
IPC分类号： G10L15/00
CPC分类号： G10L15/22 , G10L15/08 , G10L2015/223
摘要： A method for speech recognition includes: prompting a user with a first query to input speech into a speech recognition engine; determining if the inputted speech is correctly recognized; wherein in the event the inputted speech is correctly recognized proceeding to a new task; wherein in the event the inputted speech is not correctly recognized, prompting the user repeatedly with the first query to input speech into the speech recognition engine, and determining if the inputted speech is correctly recognized until a predefined limit on repetitions has been met; wherein in the event the predefined limit has been met without correctly recognizing the inputted user speech, prompting speech input from the user with a secondary query for redundant information; and cross-referencing the user's n-best result from the first query with the n-best result from the second query to obtain a top hypothesis.
摘要翻译：一种用于语音识别的方法包括：提示用户进行第一次查询以将语音输入到语音识别引擎中; 确定输入的语音是否被正确识别; 其中在所输入的语音被正确识别进行到新任务的情况下; 其中在所输入的语音未被正确识别的情况下，用所述第一询问反复提示所述用户向所述语音识别引擎输入语音，并且确定所输入的语音是否被正确识别，直到已经满足预定的重复限制为止; 其中在没有正确地识别所输入的用户语音的情况下满足所述预定义限制的情况下，用冗余信息的辅助查询提示来自用户的语音输入; 并用第二个查询的n个最佳结果交叉引用第一个查询中用户的最佳结果，以获得最高假设。

5. 发明授权

US07467090B1 Using web FAQ data for creating self-service speech applications 有权
标题翻译：使用Web常见问题数据创建自助语音应用程序
公开(公告)号：US07467090B1
公开(公告)日：2008-12-16
申请号：US12037996
申请日：2008-02-27
申请人： Osamuyimen T. Stewart , David M. Lubensky , Ea-Ee Jan , Xiang Li
发明人： Osamuyimen T. Stewart , David M. Lubensky , Ea-Ee Jan , Xiang Li
IPC分类号： G10L21/00
CPC分类号： G06F17/30861 , G06F17/2785 , G06F17/30684 , G10L15/26 , H04M3/4938
摘要： In one example, this invention presents a method of providing the same self-service content that is available on the web interface to users contacting by telephone, knowing that the web and telephone are fundamentally different user interfaces. In one embodiment, this seeks to protect the general idea of how to playback web data in real-time to the user over the speech interface. For this purpose, a method is presented comprising of the general steps through which the web data is initially sent to an automatic transformation module. Then, that transformation module refines or re-structures the web data to make it suitable for the speech interface. The algorithm in the module is predicated on the user interface principles of cognitive complexity and limitations on short term memory based on which FAQ types are classified into one of the following four classes: simple, medium, complex, and complex-complex.
摘要翻译：在一个示例中，本发明提供了一种方法，即知道网络和电话是根本上不同的用户界面的，通过电话向网络接口提供与用户联系的相同的自助服务内容。在一个实施例中，这旨在保护如何通过语音界面实时地向用户播放web数据的一般思想。为此，提出了一种方法，其包括最初将web数据发送到自动变换模块的一般步骤。然后，该转换模块对网络数据进行优化或重构，使其适合于语音界面。模块中的算法基于用户界面的认知复杂性原理和短期记忆的限制，基于哪些常见问题类型分为以下四个类别之一：简单，中等，复杂和复杂。

6. 发明授权

US5745649A Automated speech recognition using a plurality of different multilayer perception structures to model a plurality of distinct phoneme categories 失效
标题翻译：使用多个不同的多层感知结构来自动语音识别来模拟多个不同的音素类别
公开(公告)号：US5745649A
公开(公告)日：1998-04-28
申请号：US820937
申请日：1997-03-19
申请人： David M. Lubensky
发明人： David M. Lubensky
IPC分类号： G10L15/14 , G10L15/16 , G10L5/04
CPC分类号： G10L15/16 , G10L15/142
摘要： For speech recognition systems a method for modeling context-dependent phonetic categories using artificial neural nets has been described. First, linguistically motivated context-clustering is employed to reduce the number of context-dependent categories. Second, phone-specific MLP structures are used where the number of outputs in each MLP is based on the number of left and right contexts occurring in a training database. The structure of each MLP can be automatically determined using the cascade-correlation learning algorithm.
摘要翻译：对于语音识别系统，已经描述了使用人造神经网络来对上下文相关语音分类建模的方法。首先，采用语言动机的上下文聚类来减少上下文相关类别的数量。第二，使用电话特定的MLP结构，其中每个MLP中的输出数量基于在训练数据库中发生的左右上下文的数量。可以使用级联相关学习算法自动确定每个MLP的结构。

7. 发明授权

US5719921A Methods and apparatus for activating telephone services in response to speech 失效
标题翻译：响应语音激活电话服务的方法和装置
公开(公告)号：US5719921A
公开(公告)日：1998-02-17
申请号：US609029
申请日：1996-02-29
申请人： George J. Vysotsky , Ayman O. Asadi , David M. Lubensky , Vijay R. Raman , Jayant M. Naik
发明人： George J. Vysotsky , Ayman O. Asadi , David M. Lubensky , Vijay R. Raman , Jayant M. Naik
IPC分类号： G10L15/00 , G10L15/06 , G10L15/20 , G10L15/22 , G10L15/26 , G10L15/28 , H04M3/42 , H04M3/44 , G10L9/08 , H04M1/30 , H04M1/66
CPC分类号： H04M1/271 , G10L15/065 , G10L15/20 , G10L15/22 , G10L15/26 , G10L15/34 , H04M3/42204 , H04M3/44 , G10L2015/088 , H04M2201/40 , H04M3/42
摘要： Methods and apparatus for activating telephone services in response to speech are described. A directory including names is maintained for each customer. A speaker dependent speech template and a telephone number for each name, is maintained as part of each customer's directory. Speaker independent speech templates are used for recognizing commands. The present invention has the advantage of permitting a customer to place a call by speaking a person's name which serves as a destination identifier without having to speak an additional command or steering word to place the call. This is achieved by treating the receipt of a spoken name in the absence of a command as an implicit command to place a call. Explicit speaker independent commands are used to invoke features or services other than call placement. Speaker independent and speaker dependent speech recognition are performed on a customer's speech in parallel. An arbiter is used to decide which function or service should be performed when an apparent conflict arises as a result of both the speaker dependent and speaker independent speech recognition step outputs. Stochastic grammars, word spotting and/or out-of-vocabulary rejection are used as part of the speech recognition process to provide a user friendly interface which permits the use of spontaneous speech. Voice verification is performed on a selective basis where security is of concern.
摘要翻译：描述了响应于语音激活电话服务的方法和装置。为每个客户维护包含名称的目录。每个名字的说话者依赖语音模板和电话号码都作为每个客户目录的一部分进行维护。扬声器独立语音模板用于识别命令。本发明的优点在于，允许客户通过说出作为目的地标识符的人的姓名来进行呼叫，而不用说另外的命令或指导词来进行呼叫。这是通过在没有命令的情况下处理接收到口语名称作为发出呼叫的隐式命令来实现的。独立于显示扬声器的命令用于调用除呼叫位置之外的功能或服务。扬声器独立和扬声器相关语音识别是在客户演讲中并行执行的。当由于说话人依赖和说话者独立的语音识别步骤输出而产生明显的冲突时，仲裁器用于决定应该执行哪个功能或服务。语音识别过程的一部分使用随机语法，单词发音和/或超出词汇拒绝，以提供允许使用自发语音的用户友好界面。语音验证是在安全性受到关注的基础上进行的。

8. 发明申请

US20130231916A1 METHOD AND APPARATUS FOR FAST TRANSLATION MEMORY SEARCH 有权
标题翻译：用于快速翻译记忆搜索的方法和装置
公开(公告)号：US20130231916A1
公开(公告)日：2013-09-05
申请号：US13412104
申请日：2012-03-05
申请人： Juan M. Huerta , David M. Lubensky , Cheng Wu
发明人： Juan M. Huerta , David M. Lubensky , Cheng Wu
IPC分类号： G06F17/28
CPC分类号： G06F17/2827 , G06F17/2836
摘要： Methods and systems for fast translation memory search include, in response to an input query string, identifying a plurality of hypothesis strings stored in a translation memory as candidates to match the query string. One or more candidates are eliminated, using a processor, where string lengths between the candidates and the query string are at least a cutoff value representing a string edit distance. One or more candidates are eliminated where differences in word frequency distributions between the candidates and the query string are at least the cutoff value. One or more candidates are eliminated by employing a dynamic programming matrix where string edit distances between the candidates and the query string are at least the cutoff value. A number of remaining candidates are outputted as matches to the query string.
摘要翻译：用于快速翻译存储器搜索的方法和系统包括响应于输入查询字符串，将存储在翻译存储器中的多个假设字符串识别为与查询字符串匹配的候选。消除一个或多个候选者，使用处理器，候选者和查询字符串之间的串长度至少是表示字符串编辑距离的截止值。删除一个或多个候选者，其中候选和查询字符串之间的字频率分布的差异至少为截止值。通过采用动态编程矩阵来消除一个或多个候选者，其中候选和查询串之间的字符串编辑距离至少为截止值。剩余的多个候选者作为匹配输出到查询字符串。

9. 发明申请

US20120136646A1 Data Security System 有权
标题翻译：数据安全系统
公开(公告)号：US20120136646A1
公开(公告)日：2012-05-31
申请号：US12956739
申请日：2010-11-30
申请人： Carl J. Kraenzel , David M. Lubensky , Baiju Dhirajlal Mandalia , Cheng Wu
发明人： Carl J. Kraenzel , David M. Lubensky , Baiju Dhirajlal Mandalia , Cheng Wu
IPC分类号： G06F17/28
CPC分类号： G06F17/289 , G06F17/2854 , G10L13/02 , G10L15/26
摘要： A method, computer system, and computer program product for translating information. The computer system receives the information for a translation. The computer system identifies portions of the information based on a set of rules for security for the information in response to receiving the information. The computer system sends the portions of the information to a plurality of translation systems. In response to receiving translation results from the plurality of translation systems for respective portions of the information, the computer system combines the translation results for the respective portions to form a consolidated translation of the information.
摘要翻译：一种用于翻译信息的方法，计算机系统和计算机程序产品。计算机系统接收翻译信息。计算机系统基于用于响应于接收信息的信息的安全性的一组规则来识别信息的部分。计算机系统将该部分信息发送到多个翻译系统。响应于从信息的各个部分接收来自多个翻译系统的翻译结果，计算机系统组合各个部分的翻译结果以形成信息的综合翻译。

10. 发明申请

US20090287483A1 METHOD AND SYSTEM FOR IMPROVED SPEECH RECOGNITION 有权
标题翻译：改进语音识别的方法和系统
公开(公告)号：US20090287483A1
公开(公告)日：2009-11-19
申请号：US12120316
申请日：2008-05-14
申请人： Raymond L. Co , Ee-ee Jan , David M. Lubensky
发明人： Raymond L. Co , Ee-ee Jan , David M. Lubensky
IPC分类号： G10L15/00
CPC分类号： G10L15/22 , G10L15/08 , G10L2015/223
摘要： A method for speech recognition includes: prompting a user with a first query to input speech into a speech recognition engine; determining if the inputted speech is correctly recognized; wherein in the event the inputted speech is correctly recognized proceeding to a new task; wherein in the event the inputted speech is not correctly recognized, prompting the user repeatedly with the first query to input speech into the speech recognition engine, and determining if the inputted speech is correctly recognized until a predefined limit on repetitions has been met; wherein in the event the predefined limit has been met without correctly recognizing the inputted user speech, prompting speech input from the user with a secondary query for redundant information; and cross-referencing the user's n-best result from the first query with the n-best result from the second query to obtain a top hypothesis.
摘要翻译：一种用于语音识别的方法包括：提示用户进行第一次查询以将语音输入到语音识别引擎中; 确定输入的语音是否被正确识别; 其中在所输入的语音被正确识别进行到新任务的情况下; 其中在所输入的语音未被正确识别的情况下，用所述第一询问反复提示所述用户向所述语音识别引擎输入语音，并且确定所输入的语音是否被正确识别，直到已经满足预定的重复限制为止; 其中在没有正确地识别所输入的用户语音的情况下满足所述预定义限制的情况下，用冗余信息的辅助查询提示来自用户的语音输入; 并用第二个查询的n个最佳结果交叉引用第一个查询中用户的最佳结果，以获得最高假设。

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