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    • 1. 发明授权
    • Text dependent speaker recognition with long-term feature based on functional data analysis
    • 基于功能数据分析的具有长期特征的文本相关扬声器识别
    • US09153235B2
    • 2015-10-06
    • US13799647
    • 2013-03-13
    • Chenhao ZhangRuxin ChenThomas Fang Zheng
    • Chenhao ZhangRuxin ChenThomas Fang Zheng
    • G10L17/00G10L17/02G10L17/08G10L17/24
    • G10L17/00G10L17/02G10L17/08G10L17/24
    • One or more test features are extracted from a time domain signal. The test features are represented by discrete data. The discrete data is represented for each of the one or more test features by a corresponding one or more fitting functions, which are defined in terms of finite number of continuous basis functions and a corresponding finite number of expansion coefficients. Each fitting function is compressed through Functional Principal Component Analysis (FPCA) to generate corresponding sets of principal components. Each principal component for a given test feature is uncorrelated to each other principal component for the given test feature. A distance between a set of principal components for the given test feature and a set of principal components for one or more training features with the processing system is calculated. The test feature is classified according to the distance calculated with the processing system.
    • 从时域信号中提取一个或多个测试特征。 测试特征由离散数据表示。 通过相应的一个或多个拟合函数为一个或多个测试特征中的每一个表示离散数据,其根据有限数量的连续基函数和对应的有限数量的扩展系数来定义。 每个拟合函数通过功能主成分分析(FPCA)进行压缩,以生成相应的主成分组。 给定测试特征的每个主要组件与给定测试特征的每个主要组件不相关。 计算给定测试特征的一组主要组件与处理系统的一个或多个训练特征的一组主要组件之间的距离。 测试功能根据处理系统计算的距离进行分类。
    • 2. 发明授权
    • Source separation by independent component analysis in conjunction with source direction information
    • 源分离与独立成分分析结合源方向信息
    • US08880395B2
    • 2014-11-04
    • US13464828
    • 2012-05-04
    • Jaekwon YooRuxin Chen
    • Jaekwon YooRuxin Chen
    • G10L21/00
    • G10L21/0272
    • Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source direction information is utilized in the separation process, and independent component analysis techniques described herein use multivariate probability density functions to preserve the alignment of frequency bins in the source separation process. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.
    • 公开了用于信号处理的方法和装置。 可以通过独立分量分析来执行源分离以从源信号的混合中提取源信号。 在分离过程中使用源方向信息,并且本文描述的独立分量分析技术使用多变量概率密度函数来保持源分离过程中频率仓的对准。 要强调的是,该摘要被提供以符合要求抽象的规则,允许搜索者或其他读者快速确定技术公开内容的主题。 提交它的理解是,它不会用于解释或限制权利要求的范围或含义。
    • 3. 发明申请
    • HYBRID PERFORMANCE SCALING OR SPEECH RECOGNITION
    • 混合性能缩放或语音识别
    • US20140237277A1
    • 2014-08-21
    • US13791716
    • 2013-03-08
    • Dominic S. MallinsonRuxin Chen
    • Dominic S. MallinsonRuxin Chen
    • G06F1/32
    • G06F1/3206G06F1/3203G06F1/3231G06F1/3293G06F3/017G10L15/22G10L25/78Y02D10/122Y02D10/173
    • Aspects of the present disclosure describe methods and apparatuses for executing operations on a client device platform that is operating in a low-power state. A first analysis may be used to assign a first confidence score to a recorded non-tactile input. When the first confidence score is above a first threshold an intermediate-power state may be activated. A second more detailed analysis may then assign a second confidence score to the non-tactile input. When the second confidence score is above a second threshold, then the operation is initiated. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.
    • 本公开的方面描述了在以低功率状态运行的客户端设备平台上执行操作的方法和装置。 可以使用第一分析来将第一置信度分数分配给记录的非触觉输入。 当第一置信度高于第一阈值时,可以激活中间功率状态。 然后,第二更详细的分析可以向非触觉输入分配第二置信度分数。 当第二置信度分数高于第二阈值时,开始该操作。 要强调的是,该摘要被提供以符合要求抽象的规则,允许搜索者或其他读者快速确定技术公开内容的主题。 提交它的理解是,它不会用于解释或限制权利要求的范围或含义。
    • 4. 发明授权
    • Speech syllable/vowel/phone boundary detection using auditory attention cues
    • 语音音节/元音/电话边界检测使用听觉注意线索
    • US08756061B2
    • 2014-06-17
    • US13078866
    • 2011-04-01
    • Ozlem KalinliRuxin Chen
    • Ozlem KalinliRuxin Chen
    • G10L15/04
    • G10L15/05G10L15/04G10L15/16G10L15/24G10L15/34G10L25/03
    • In syllable or vowel or phone boundary detection during speech, an auditory spectrum may be determined for an input window of sound and one or more multi-scale features may be extracted from the auditory spectrum. Each multi-scale feature can be extracted using a separate two-dimensional spectro-temporal receptive filter. One or more feature maps corresponding to the one or more multi-scale features can be generated and an auditory gist vector can be extracted from each of the one or more feature maps. A cumulative gist vector may be obtained through augmentation of each auditory gist vector extracted from the one or more feature maps. One or more syllable or vowel or phone boundaries in the input window of sound can be detected by mapping the cumulative gist vector to one or more syllable or vowel or phone boundary characteristics using a machine learning algorithm.
    • 在语音期间的音节或元音或电话边界检测中,可以为输入声音窗口确定听觉频谱,并且可以从听觉谱中提取一个或多个多尺度特征。 可以使用单独的二维光谱接收滤波器来提取每个多尺度特征。 可以生成与一个或多个多尺度特征相对应的一个或多个特征图,并且可以从一个或多个特征图中的每一个提取听觉要点矢量。 可以通过增加从一个或多个特征图提取的每个听觉要素矢量来获得累积的要点向量。 通过使用机器学习算法将累积的要点向量映射到一个或多个音节或元音或电话边界特征,可以检测声音的输入窗口中的一个或多个音节或元音或电话边界。
    • 8. 发明申请
    • SPEECH SYLLABLE/VOWEL/PHONE BOUNDARY DETECTION USING AUDITORY ATTENTION CUES
    • 使用审计注意事项的语音可以/ VOWEL /电话边界检测
    • US20120253812A1
    • 2012-10-04
    • US13078866
    • 2011-04-01
    • OZLEM KALINLIRuxin Chen
    • OZLEM KALINLIRuxin Chen
    • G10L15/04
    • G10L15/05G10L15/04G10L15/16G10L15/24G10L15/34G10L25/03
    • In syllable or vowel or phone boundary detection during speech, an auditory spectrum may be determined for an input window of sound and one or more multi-scale features may be extracted from the auditory spectrum. Each multi-scale feature can be extracted using a separate two-dimensional spectro-temporal receptive filter. One or more feature maps corresponding to the one or more multi-scale features can be generated and an auditory gist vector can be extracted from each of the one or more feature maps. A cumulative gist vector may be obtained through augmentation of each auditory gist vector extracted from the one or more feature maps. One or more syllable or vowel or phone boundaries in the input window of sound can be detected by mapping the cumulative gist vector to one or more syllable or vowel or phone boundary characteristics using a machine learning algorithm.
    • 在语音期间的音节或元音或电话边界检测中,可以为输入声音窗口确定听觉频谱,并且可以从听觉谱中提取一个或多个多尺度特征。 可以使用单独的二维光谱接收滤波器来提取每个多尺度特征。 可以生成与一个或多个多尺度特征相对应的一个或多个特征图,并且可以从一个或多个特征图中的每一个提取听觉要点矢量。 可以通过增加从一个或多个特征图提取的每个听觉要素矢量来获得累积的要点向量。 通过使用机器学习算法将累积的要点向量映射到一个或多个音节或元音或电话边界特征,可以检测声音的输入窗口中的一个或多个音节或元音或电话边界。
    • 9. 发明授权
    • Structure for grammar and dictionary representation in voice recognition and method for simplifying link and node-generated grammars
    • 用于语音识别中的语法和字典表示的结构以及用于简化链接和节点生成的语法的方法
    • US08190433B2
    • 2012-05-29
    • US13031104
    • 2011-02-18
    • Gustavo Hernandez AbregoRuxin Chen
    • Gustavo Hernandez AbregoRuxin Chen
    • G10L15/18
    • G10L15/193G10L15/285
    • A speech recognition engine is provided with an acoustic model and a layered grammar and dictionary library. The layered grammar and dictionary library includes a language and non-grammar layer that supplies types of rules a grammar definition layer can use and defines non-grammar the speech recognition engine should ignore. The layered grammar and dictionary library also includes a dictionary layer that defines phonetic transcriptions for word groups the speech recognition engine is meant to recognize when voice input is received. The layered grammar and dictionary library further includes a grammar definition layer that applies rules from the language and non-grammar layer to define combinations of word groups the speech recognition system is meant to recognize. Voice input is received at a speech recognition engine and is processed using the acoustic model and the layered grammar and dictionary library.
    • 语音识别引擎设有声学模型和分层语法和字典库。 分层语法和字典库包括语言和非语法层,提供语法定义层可以使用的规则类型,并定义语音识别引擎应忽略的非语法。 分层语法和字典库还包括字典层,其定义语音识别引擎在接收到语音输入时识别的单词组的语音转录。 分层语法和字典库还包括语法定义层,其应用语言和非语法层的规则来定义语音识别系统意图识别的单词组的组合。 在语音识别引擎处接收语音输入,并使用声学模型和分层语法和字典库进行处理。
    • 10. 发明授权
    • Voice recognition with parallel gender and age normalization
    • 语音识别与并行性别和年龄归一化
    • US08010358B2
    • 2011-08-30
    • US11358272
    • 2006-02-21
    • Ruxin Chen
    • Ruxin Chen
    • G10L17/00G10L15/28G10L15/14
    • G10L15/32G10L15/065
    • Methods and apparatus for voice recognition are disclosed. A voice signal is obtained and two or more voice recognition analyses are performed on the voice signal. Each voice recognition analysis uses a filter bank defined by a different maximum frequency and a different minimum frequency and wherein each voice recognition analysis produces a recognition probability ri of recognition of one or more speech units, whereby there are two or more recognition probabilities ri. The maximum frequency and the minimum frequency may be adjusted every time speech is windowed and analyzed. A final recognition probability Pf is determined based on the two or more recognition probabilities ri.
    • 公开了用于语音识别的方法和装置。 获得语音信号,并对语音信号执行两个或多个语音识别分析。 每个语音识别分析使用由不同的最大频率和不同的最小频率定义的滤波器组,并且其中每个语音识别分析产生识别一个或多个语音单元的识别概率ri,由此存在两个或更多个识别概率ri。 可以在每次打开和分析语音时调整最大频率和最小频率。 基于两个以上的识别概率ri来确定最终识别概率Pf。