Voice tagging, voice annotation, and speech recognition for portable devices with optional post processing
    11.
    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。

    Voice tagging, voice annotation, and speech recognition for portable devices with optional post processing
    12.
    发明授权
    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:US07324943B2

    公开(公告)日:2008-01-29

    申请号:US10677174

    申请日:2003-10-02

    IPC分类号: G10L21/00 H04N5/76

    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。

    Speech data mining for call center management
    13.
    发明申请
    Speech data mining for call center management 审中-公开
    语音数据挖掘用于呼叫中心管理

    公开(公告)号:US20050010411A1

    公开(公告)日:2005-01-13

    申请号:US10616006

    申请日:2003-07-09

    IPC分类号: G10L15/26 G10L17/00 G10L15/00

    CPC分类号: G10L15/26 G10L17/00

    摘要: A speech data mining system for use in generating a rich transcription having utility in call center management includes a speech differentiation module differentiating between speech of interacting speakers, and a speech recognition module improving automatic recognition of speech of one speaker based on interaction with another speaker employed as a reference speaker. A transcript generation module generates a rich transcript based on recognized speech of the speakers. Focused, interactive language models improve recognition of a customer on a low quality channel using context extracted from speech of a call center operator on a high quality channel with a speech model adapted to the operator. Mined speech data includes number of interaction turns, customer frustration phrases, operator polity, interruptions, and/or contexts extracted from speech recognition results, such as topics, complaints, solutions, and resolutions. Mined speech data is useful in call center and/or product or service quality management.

    摘要翻译: 用于产生在呼叫中心管理中具有效用的丰富录音的语音数据挖掘系统包括区分交互式扬声器的语音的语音区分模块和改善一个扬声器的语音的自动识别的语音识别模块, 作为参考发言人。 转录本生成模块基于扬声器的识别语音生成丰富的录音。 专注的交互式语言模型通过使用适合于操作员的语音模型,在高质量频道上从呼叫中心运营商的语音提取的上下文,改善对低质量信道上客户的识别。 挖掘的语音数据包括从诸如主题,投诉,解决方案和分辨率的语音识别结果中提取的交互轮廓数量,客户沮丧短语,运营商政治,中断和/或上下文。 挖掘的语音数据在呼叫中心和/或产品或服务质量管理中是有用的。

    Methods and apparatus for blind channel estimation based upon speech correlation structure
    14.
    发明授权
    Methods and apparatus for blind channel estimation based upon speech correlation structure 有权
    基于语音相关结构的盲信道估计方法与装置

    公开(公告)号:US06687672B2

    公开(公告)日:2004-02-03

    申请号:US10099428

    申请日:2002-03-15

    IPC分类号: G10L1508

    CPC分类号: G10L21/0208

    摘要: Methods and apparatus for blind channel estimation of a speech signal corrupted by a communication channel are provided. One method includes converting a noisy speech signal into either a cepstral representation or a log-spectral representation; estimating a correlation of the representation of the noisy speech signal; determining an average of the noisy speech signal; constructing and solving, subject to a minimization constraint, a system of linear equations utilizing a correlation structure of a clean speech training signal, the correlation of the representation of the noisy speech signal, and the average of the noisy speech signal; and selecting a sign of the solution of the system of linear equations to estimate an average clean speech signal in a processing window.

    摘要翻译: 提供了由通信信道损坏的语音信号的盲信道估计的方法和装置。 一种方法包括将噪声语音信号转换成倒谱表示或对数谱表示; 估计噪声语音信号的表示的相关性; 确定噪声语音信号的平均值; 利用最小化约束,构建和求解利用清晰语音训练信号的相关结构,噪声语音信号的表示与噪声语音信号的平均值的相关性的线性方程组; 以及选择线性方程式的解的符号来估计处理窗口中的平均清洁语音信号。

    Unsupervised speech model adaptation using reliable information among N-best strings
    15.
    发明授权
    Unsupervised speech model adaptation using reliable information among N-best strings 失效
    无人监督的语音模型适应使用N最佳字符串中的可靠信息

    公开(公告)号:US06205426B1

    公开(公告)日:2001-03-20

    申请号:US09237170

    申请日:1999-01-25

    IPC分类号: G10L1514

    CPC分类号: G10L15/065

    摘要: The system performs unsupervised speech model adaptation using the recognizer to generate the N-best solutions for an input utterance. Each of these N-best solutions is tested by a reliable information extraction process. Reliable information is extracted by a weighting technique based on likelihood scores generated by the recognizer, or by a non-linear thresholding function. The system may be used in a single pass implementation or iteratively in a multi-pass implementation.

    摘要翻译: 该系统使用识别器执行无监督的语音模型自适应,以产生用于输入语音的N最佳解。 这些N最佳解决方案中的每一个都通过可靠的信息提取过程进行测试。 通过基于由识别器生成的似然分数的加权技术或非线性阈值函数来提取可靠信息。 该系统可以在单遍实现中或在多遍实现中迭代地使用。

    Speaker and environment adaptation based on linear separation of variability sources
    16.
    发明授权
    Speaker and environment adaptation based on linear separation of variability sources 有权
    基于可变性来源线性分离的扬声器和环境适应

    公开(公告)号:US06915259B2

    公开(公告)日:2005-07-05

    申请号:US09864838

    申请日:2001-05-24

    IPC分类号: G10L15/06 G10L21/02

    CPC分类号: G10L15/07 G10L21/0208

    摘要: Linear approximation of the background noise is applied after feature extraction and prior to speaker adaptation to allow the speaker adaptation system to adapt the speech models to the enrolling user without distortion from background noise. The linear approximation is applied in the feature domain, such as in the cepstral domain. Any adaptation technique that is commutative in the feature domain may be used.

    摘要翻译: 背景噪声的线性近似在特征提取之后并且在说话者适配之前被应用,以允许扬声器适配系统将语音模型适应于登记用户,而不会从背景噪声失真。 线性近似应用于特征域,如倒谱域。 可以使用在特征域中可交换的任何适配技术。

    System and method of media file access and retrieval using speech recognition
    17.
    发明授权
    System and method of media file access and retrieval using speech recognition 有权
    使用语音识别的媒体文件访问和检索的系统和方法

    公开(公告)号:US06907397B2

    公开(公告)日:2005-06-14

    申请号:US10245727

    申请日:2002-09-16

    摘要: An embedded device for playing media files is capable of generating a play list of media files based on input speech from a user. It includes an indexer generating a plurality of speech recognition grammars. According to one aspect of the invention, the indexer generates speech recognition grammars based on contents of a media file header of the media file. According to another aspect of the invention, the indexer generates speech recognition grammars based on categories in a file path for retrieving the media file to a user location. When a speech recognizer receives an input speech from a user while in a selection mode, a media file selector compares the input speech received while in the selection mode to the plurality of speech recognition grammars, thereby selecting the media file.

    摘要翻译: 用于播放媒体文件的嵌入式设备能够基于来自用户的输入语音来生成媒体文件的播放列表。 它包括产生多个语音识别语法的索引器。 根据本发明的一个方面,索引器基于媒体文件的媒体文件头的内容生成语音识别语法。 根据本发明的另一方面,索引器基于用于将媒体文件检索到用户位置的文件路径中的类别来生成语音识别语法。 当语音识别器在选择模式下从用户接收到输入语音时,媒体文件选择器将选择模式下接收到的输入语音与多个语音识别语法进行比较,从而选择媒体文件。

    Bubble splitting for compact acoustic modeling
    20.
    发明授权
    Bubble splitting for compact acoustic modeling 有权
    气泡分裂用于紧凑的声学建模

    公开(公告)号:US07328154B2

    公开(公告)日:2008-02-05

    申请号:US10639974

    申请日:2003-08-13

    IPC分类号: G10L15/00

    摘要: An improved method is provided for constructing compact acoustic models for use in a speech recognizer. The method includes: partitioning speech data from a plurality of training speakers according to at least one speech related criteria (i.e., vocal tract length); grouping together the partitioned speech data from training speakers having a similar speech characteristic; and training an acoustic bubble model for each group using the speech data within the group.

    摘要翻译: 提供了一种用于构建用于语音识别器中的紧凑声学模型的改进方法。 该方法包括:根据至少一个语音相关标准(即,声道长度)来分割来自多个训练说话者的语音数据; 将具有类似语音特征的训练说话者的分割语音数据分组在一起; 并使用组内的语音数据为每个组训练声音气泡模型。