Hands-free voice dialing for portable and remote devices
    1.
    发明申请
    Hands-free voice dialing for portable and remote devices 审中-公开
    便携式和远程设备的免提语音拨号

    公开(公告)号:US20060009974A1

    公开(公告)日:2006-01-12

    申请号:US10888916

    申请日:2004-07-09

    IPC分类号: G10L15/18

    摘要: Dynamically constructed grammar-constraints and frequency or statistics-based constraints are used to constrain the speech recognizer and to optionally rescore the output to improve recognition accuracy. The recognition system is well adapted for hands-free operation of portable devices, such as for voice dialing operations.

    摘要翻译: 动态构造的语法约束和基于频率或基于统计的约束用于限制语音识别器,并且可选地重新输出输出以提高识别精度。 识别系统很好地适用于便携式设备的免提操作,例如语音拨号操作。

    Joint signal and model based noise matching noise robustness method for automatic speech recognition
    2.
    发明授权
    Joint signal and model based noise matching noise robustness method for automatic speech recognition 有权
    基于信号和模型的噪声匹配噪声鲁棒性自动语音识别方法

    公开(公告)号:US07729908B2

    公开(公告)日:2010-06-01

    申请号:US11369936

    申请日:2006-03-06

    IPC分类号: G10L15/20 G10L15/06 G10L21/02

    CPC分类号: G10L15/20 G10L21/0216

    摘要: A noise robustness method operates jointly in a signal domain and a model domain. For example, energy is added in the signal domain for frequency bands where an actual noise level of an incoming signal is lower than a noise level used to train models, thus obtaining a compensated signal. Also, energy is added in the model domain for frequency bands where noise level of the incoming signal or the compensated signal is higher than the noise level used to train the models. Moreover, energy is never removed, thereby avoiding problems of higher sensitivity of energy removal to estimation errors.

    摘要翻译: 噪声鲁棒性方法在信号域和模型域中共同操作。 例如,在信号域中增加能量,其中输入信号的实际噪声电平低于用于训练模型的噪声电平,从而获得补偿信号。 此外,在模型域中增加能量,其中输入信号或补偿信号的噪声电平高于用于训练模型的噪声电平的频带。 此外,能量永远不会被去除,从而避免了能量去除对估计误差的更高灵敏度的问题。

    Voice tagging, voice annotation, and speech recognition for portable devices with optional post processing
    3.
    发明申请
    Voice tagging, voice annotation, and speech recognition for portable devices with optional post processing 有权
    语音标记,语音注释和可选后置处理的便携式设备的语音识别

    公开(公告)号:US20050075881A1

    公开(公告)日:2005-04-07

    申请号:US10677174

    申请日:2003-10-02

    IPC分类号: G10L15/26 G10L21/00

    CPC分类号: G06F17/30796 G10L15/26

    摘要: A media capture device has an audio input receptive of user speech relating to a media capture activity in close temporal relation to the media capture activity. A plurality of focused speech recognition lexica respectively relating to media capture activities are stored on the device, and a speech recognizer recognizes the user speech based on a selected one of the focused speech recognition lexica. A media tagger tags captured media with generated speech recognition text, and a media annotator annotates the captured media with a sample of the user speech that is suitable for input to a speech recognizer. Tagging and annotating are based on close temporal relation between receipt of the user speech and capture of the captured media. Annotations may be converted to tags during post processing, employed to edit a lexicon using letter-to-sound rules and spelled word input, or matched directly to speech to retrieve captured media.

    摘要翻译: 媒体捕获设备具有接收与媒体捕获活动紧密相关的媒体捕获活动的用户语音的音频输入。 分别与媒体捕获活动相关的多个聚焦语音识别词典被存储在设备上,并且语音识别器基于所选择的一个焦点语音识别词典识别用户语音。 媒体标签器使用生成的语音识别文本来标记捕获的媒体,并且媒体注释器用适合于输入到语音识别器的用户语音的样本来注释所捕获的媒体。 标记和注释是基于用户语音的接收和捕获的媒体的捕获之间的紧密的时间关系。 在后期处理中,注释可以转换为标签,用于使用字母对声音规则和拼写单词输入来编辑词典,或直接与语音匹配以检索所捕获的媒体。

    METHOD AND SYSTEM OF IDENTIFYING A USER OF A HANDHELD DEVICE
    4.
    发明申请
    METHOD AND SYSTEM OF IDENTIFYING A USER OF A HANDHELD DEVICE 审中-公开
    识别手持设备用户的方法和系统

    公开(公告)号:US20110043475A1

    公开(公告)日:2011-02-24

    申请号:US12988745

    申请日:2009-04-21

    IPC分类号: G06F3/041

    摘要: A system and method for identifying a user of a handheld device is herein disclosed. The device implementing the method and system may attempt to identify a user based on signals that are incidental to a user's handling of the device. The signals are generated by a variety of sensors dispersed along the periphery or within the housing. The sensors range may include touch sensors, inertial sensors, acoustic sensors, pulse oximiters, and a touchpad. Based on the sensors and corresponding signals, identification information is generated. The identification information is used to identify the user of the handheld device. The handheld device may implement various statistical learning and data mining techniques to increase the robustness of the system. The device may also authenticate the user based on the user drawing a circle, or other shape.

    摘要翻译: 本文公开了一种用于识别手持设备的用户的系统和方法。 实现该方法和系统的设备可以基于用户对设备的处理附带的信号来尝试识别用户。 信号由沿着周边或壳体内分散的各种传感器产生。 传感器范围可以包括触摸传感器,惯性传感器,声学传感器,脉冲嗅觉器和触摸板。 基于传感器和相应的信号,生成识别信息。 识别信息用于识别手持设备的用户。 手持设备可以实现各种统计学习和数据挖掘技术以增加系统的鲁棒性。 设备还可以基于用户绘制圆形或其他形状来认证用户。

    Focused language models for improved speech input of structured documents
    5.
    发明授权
    Focused language models for improved speech input of structured documents 有权
    用于改进结构化文档语音输入的专注语言模型

    公开(公告)号:US06901364B2

    公开(公告)日:2005-05-31

    申请号:US09951093

    申请日:2001-09-13

    CPC分类号: G10L15/1815 G10L15/30

    摘要: An e-mail message process is provided for use with a personal digital assistant which allows for the use of input speech messaging which is converted to text using a focused language model which is downloaded by a cellular phone connection to an Internet server which provides the focused language model based upon a topic for the intended e-mail message. The text that is generated from the input speech method can be summarized by the e-mail message processor and can be edited by the user. The generated e-mail message can then be transmitted again via cellular connection to an Internet e-mail server for transmitting the e-mail message to a recipient.

    摘要翻译: 提供电子邮件消息处理以与个人数字助理一起使用,该个人数字助理允许使用输入语音消息传送,其使用由通过蜂窝电话连接下载的聚焦语言模型转换为文本,该互联网服务器提供聚焦 基于预期电子邮件的主题的语言模型。 从输入语音方法生成的文本可以由电子邮件消息处理器来总结,并且可以由用户编辑。 然后可以通过蜂窝连接再次将生成的电子邮件消息发送到Internet电子邮件服务器,以将电子邮件消息发送给接收者。

    Apparatus for efficient dispatch and selection of information in law enforcement applications
    6.
    发明授权
    Apparatus for efficient dispatch and selection of information in law enforcement applications 有权
    用于在执法应用程序中高效地发送和选择信息的装置

    公开(公告)号:US06571174B2

    公开(公告)日:2003-05-27

    申请号:US09929634

    申请日:2001-08-14

    IPC分类号: G01C2134

    摘要: A navigation apparatus is disclosed which may be used by law enforcement personnel for rapid intervention to a location while adding safety and reliability to the process. The apparatus includes a computer system, having an operating system, memory and a user interface. The system further includes a positioning system, such as a GPS system for determining the position of a vehicle. The positioning system communicates with the operating system. An information database, communicating with the operating system, contains data related to routing information concerning routes for travel by the vehicle. The routing information includes safety information concerning route safety in the traveling region accessible by the vehicle. The apparatus further includes a routing system in communication with the operating system that determines a route based at least in part on the routing information. Driving directions and call information are provided multi-modally to provide the officer with critical information in an efficient and timely fashion.

    摘要翻译: 公开了一种导航装置,其可以被执法人员用于对位置的快速干预,同时为该过程增加安全性和可靠性。 该装置包括具有操作系统,存储器和用户界面的计算机系统。 该系统还包括诸如用于确定车辆位置的GPS系统的定位系统。 定位系统与操作系统通信。 与操作系统通信的信息数据库包含与车辆行驶路线有关的路线信息的数据。 路线信息包括关于车辆可接近的行驶区域中的路线安全的安全信息。 该装置还包括与操作系统通信的路由系统,其至少部分地基于路由信息来确定路由。 驾驶方向和通话信息以多方式提供,以有效和及时的方式向官员提供关键信息。

    Method for efficient, safe and reliable data entry by voice under adverse conditions

    公开(公告)号:US06996528B2

    公开(公告)日:2006-02-07

    申请号:US09921766

    申请日:2001-08-03

    IPC分类号: G10L15/22

    CPC分类号: G10L15/065 G10L15/22

    摘要: A method and apparatus for data entry by voice under adverse conditions is disclosed. More specifically it provides a way for efficient and robust form filling by voice. A form can typically contain one or several fields that must be filled in. The user communicates to a speech recognition system and word spotting is performed upon the utterance. The spotted words of an utterance form a phrase that can contain field-specific values and/or commands. Recognized values are echoed back to the speaker via a text-to-speech system. Unreliable or unsafe inputs for which the confidence measure is found to be low (e.g. ill-pronounced speech or noises) are rejected by the spotter. Speaker adaptation is furthermore performed transparently to improve speech recognition accuracy. Other input modalities can be additionally supported (e.g. keyboard and touch-screen). The system maintains a dialogue history to enable editing and correction operations on all active fields.

    Method for additive and convolutional noise adaptation in automatic speech recognition using transformed matrices
    9.
    发明授权
    Method for additive and convolutional noise adaptation in automatic speech recognition using transformed matrices 有权
    使用变换矩阵的自动语音识别中的加法和卷积噪声适应的方法

    公开(公告)号:US06691091B1

    公开(公告)日:2004-02-10

    申请号:US09628376

    申请日:2000-07-31

    IPC分类号: G10L1506

    摘要: A noise adaptation system and method provide for noise adaptation in a speech recognition system. The method includes the steps of generating a reference model based on a training speech signal, and compensating the reference model for additive noise in the cepstral domain. The reference model is also compensated for convolutional noise in the cepstral domain. In one embodiment, the convolutional noise is compensated for by estimating a convolutional bias between the reference model and a target speech signal. The estimated convolutional bias is transformed with a channel adaptation matrix, and the transformed convolutional bias is added to the reference model in the cepstral domain.

    摘要翻译: 噪声适应系统和方法提供语音识别系统中的噪声适应。 该方法包括以下步骤:基于训练语音信号产生参考模型,以及补偿倒谱域中加性噪声的参考模型。 参考模型也被补偿了倒谱域中的卷积噪声。 在一个实施例中,通过估计参考模型和目标语音信号之间的卷积偏差来补偿卷积噪声。 用通道自适应矩阵对估计的卷积偏差进行变换,并将变换的卷积偏差加到倒谱域中的参考模型中。

    Method for noise adaptation in automatic speech recognition using transformed matrices
    10.
    发明授权
    Method for noise adaptation in automatic speech recognition using transformed matrices 有权
    使用变换矩阵的自动语音识别中的噪声适应方法

    公开(公告)号:US06529872B1

    公开(公告)日:2003-03-04

    申请号:US09551001

    申请日:2000-04-18

    IPC分类号: G10L1506

    摘要: The improved noise adaptation technique employs a linear or non-linear transformation to the set of Jacobian matrices corresponding to an initial noise condition. An &agr;-adaptation parameter or artificial intelligence operation is employed in a linear or non-linear way to increase the adaptation bias added to the speech models. This corrects shortcomings of conventional Jacobian adaptation, which tend to underestimate the effect of noise. The improved adaptation technique is further enhanced by a reduced dimensionality, principal component analysis technique that reduces the computational burden, making the adaptation technique beneficial in embedded recognition systems.

    摘要翻译: 改进的噪声适应技术对与初始噪声条件相对应的雅可比矩阵集合采用线性或非线性变换。 以线性或非线性方式采用阿尔法适应参数或人工智能操作,以增加添加到语音模型中的适应偏差。 这纠正了常规雅各布适应的缺点,这倾向于低估噪声的影响。 改进的适应技术通过降低维度的主要成分分析技术进一步增强,主要成分分析技术降低了计算负担,使得适应技术在嵌入式识别系统中有益。