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    • 2. 发明授权
    • Assistive call center interface
    • 辅助呼叫中心接口
    • US07103553B2
    • 2006-09-05
    • US10454716
    • 2003-06-04
    • Ted ApplebaumJean-Claude Junqua
    • Ted ApplebaumJean-Claude Junqua
    • G10L15/00
    • G10L15/1822
    • Unstructured voice information from an incoming caller is processed by automatic speech recognition and semantic categorization system to convert the information into structured data that may then be used to access one or more databases to retrieve associated supplemental data. The structured data and associated supplemental data are then made available through a presentation system that provides information to the call center agent and, optionally, to the incoming caller. The system thus allows a call center information processing system to handle unstructured voice input for use by the live agent in handling the incoming call and for storage and retrieval at a later time. The semantic analysis system may be implemented by a global parser or by an information retrieval technique, such as latent semantic analysis. Co-occurrence of keywords may be used to associate prior calls with an incoming call to assist in understanding the purpose of the incoming call.
    • 来自呼叫者的非结构化语音信息由自动语音识别和语义分类系统处理,以将信息转换成结构化数据,然后可以用于访问一个或多个数据库以检索相关联的补充数据。 结构化数据和相关的补充数据然后通过向呼叫中心代理提供信息并且可选地提供给传入呼叫者的呈现系统可用。 因此,该系统允许呼叫中心信息处理系统处理非结构化语音输入以供实时代理使用以处理来话呼叫并在以后的时间进行存储和检索。 语义分析系统可以由全局解析器或诸如潜在语义分析之类的信息检索技术来实现。 关键字的共现可以用于将先前的呼叫与呼入呼叫相关联,以帮助理解来话呼叫的目的。
    • 6. 发明授权
    • Robust preprocessing signal equalization system and method for normalizing to a target environment
    • 强大的预处理信号均衡系统和方法,用于归一化到目标环境
    • US06411927B1
    • 2002-06-25
    • US09148401
    • 1998-09-04
    • Philippe MorinPhilippe GelinJean-Claude Junqua
    • Philippe MorinPhilippe GelinJean-Claude Junqua
    • G10L1914
    • G10L15/065G10L15/20
    • The audio source is spectrally shaped by filtering in the time domain to approximate or emulate a standardized or target microphone input channel. The background level is adjusted by adding noise to the time domain signal prior to the onset of speech to set a predetermined background noise level based on a predetermined target. The audio source is then monitored in real time and the signal-to-noise ratio is adjusted by adding noise to the time domain signal, in real time, to maintain a signal-to-noise ratio based on a predetermined target value. The normalized audio signal may be applied to both training speech and test speech. The resultant normalization minimizes the mismatch between training and testing and also improves other speech processing functions, such as speech endpoint detection.
    • 音频源通过在时域中过滤来近似或模拟标准化或目标麦克风输入通道进行频谱成形。 通过在语音开始之前对时域信号增加噪声来调整背景电平,以基于预定目标设定预定的背景噪声电平。 然后,实时监视音频源,实时地通过对时域信号增加噪声来调整信噪比,以基于预定目标值维持信噪比。 归一化音频信号可以应用于训练语音和测试语音两者。 最终归一化使训练和测试之间的不匹配最小化,并且还改善了其他语音处理功能,例如语音端点检测。