公开(公告)号：US20050075881A1

公开(公告)日：2005-04-07

申请号：US10677174

申请日：2003-10-02

申请人： Luca Rigazio ,  Robert Boman ,  Patrick Nguyen ,  Jean-Claude Junqua

发明人： Luca Rigazio ,  Robert Boman ,  Patrick Nguyen ,  Jean-Claude Junqua

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。

公开(公告)号：US07324943B2

公开(公告)日：2008-01-29

申请号：US10677174

申请日：2003-10-02

申请人： Luca Rigazio ,  Robert Boman ,  Patrick Nguyen ,  Jean-Claude Junqua

发明人： Luca Rigazio ,  Robert Boman ,  Patrick Nguyen ,  Jean-Claude Junqua

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。

公开(公告)号：US20050010411A1

公开(公告)日：2005-01-13

申请号：US10616006

申请日：2003-07-09

申请人： Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua ,  Robert Boman

发明人： Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua ,  Robert Boman

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。

公开(公告)号：US06901364B2

公开(公告)日：2005-05-31

申请号：US09951093

申请日：2001-09-13

申请人： Patrick Nguyen ,  Luca Rigazio ,  Jean-Claude Junqua

发明人： Patrick Nguyen ,  Luca Rigazio ,  Jean-Claude Junqua

IPC分类号： G10L15/18 ,  G10L15/28 ,  G10L15/26 ,  G06F17/20 ,  G10L21/00

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。

公开(公告)号：US07089182B2

公开(公告)日：2006-08-08

申请号：US10099305

申请日：2002-03-15

申请人： Younes Souilmi ,  Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua

发明人： Younes Souilmi ,  Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua

IPC分类号： G10L15/20

CPC分类号： G10L15/20 ,  G10L15/02 ,  G10L15/063 ,  G10L21/0216

摘要： A method for performing noise adaptation of a target speech signal input to a speech recognition system, where the target speech signal contains both additive and convolutional noises. The method includes estimating an additive noise bias and a convolutional noise bias; in the target speech signal; and jointly compensating the target speech signal for the additive and convolutional noise biases in a feature domain.

公开(公告)号：US06687672B2

公开(公告)日：2004-02-03

申请号：US10099428

申请日：2002-03-15

申请人： Younes Souilmi ,  Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua

发明人： Younes Souilmi ,  Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua

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。

公开(公告)号：US06915259B2

公开(公告)日：2005-07-05

申请号：US09864838

申请日：2001-05-24

申请人： Luca Rigazio ,  Patrick Nguyen ,  David Kryze ,  Jean-Claude Junqua

发明人： Luca Rigazio ,  Patrick Nguyen ,  David Kryze ,  Jean-Claude Junqua

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。

公开(公告)号：US06907397B2

公开(公告)日：2005-06-14

申请号：US10245727

申请日：2002-09-16

申请人： David Kryze ,  Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua

发明人： David Kryze ,  Luca Rigazio ,  Patrick Nguyen ,  Jean-Claude Junqua

IPC分类号： G10L15/00 ,  G06F17/30 ,  G10L20060101 ,  G10L11/00 ,  G10L15/04 ,  G10L15/06 ,  G10L15/18 ,  G10L15/26 ,  G10L21/00

CPC分类号： G06F17/30026 ,  G10L15/183 ,  G10L15/19 ,  G10L15/26 ,  Y10S707/99933 ,  Y10S707/99934 ,  Y10S707/99935

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。

公开(公告)号：US06697778B1

公开(公告)日：2004-02-24

申请号：US09610495

申请日：2000-07-05

申请人： Roland Kuhn ,  Olivier Thyes ,  Patrick Nguyen ,  Jean-Claude Junqua ,  Robert Boman

发明人： Roland Kuhn ,  Olivier Thyes ,  Patrick Nguyen ,  Jean-Claude Junqua ,  Robert Boman

IPC分类号： G10L1506

CPC分类号： G10L17/02

摘要： Client speaker locations in a speaker space are used to generate speech models for comparison with test speaker data or test speaker speech models. The speaker space can be constructed using training speakers that are entirely separate from the population of client speakers, or from client speakers, or from a mix of training and client speakers. Reestimation of the speaker space based on client environment information is also provided to improve the likelihood that the client data will fall within the speaker space. During enrollment of the clients into the speaker space, additional client speech can be obtained when predetermined conditions are met. The speaker distribution can also be used in the client enrollment step.

摘要翻译： 扬声器空间中的客户扬声器位置用于产生用于与测试扬声器数据或测试扬声器语音模型进行比较的语音模型。 扬声器空间可以使用与客户端扬声器或客户端扬声器完全分开的训练扬声器，或者由训练和客户端扬声器组合构成。 还提供了基于客户端环境信息对扬声器空间的再估计，以提高客户端数据落入扬声器空间的可能性。 在将客户登记到扬声器空间中，当满足预定条件时，可以获得额外的客户端语音。 扬声器分配也可以在客户端注册步骤中使用。

公开(公告)号：US20050228663A1

公开(公告)日：2005-10-13

申请号：US10814960

申请日：2004-03-31

申请人： Robert Boman ,  Patrick Nguyen ,  Jean-Claude Junqua

发明人： Robert Boman ,  Patrick Nguyen ,  Jean-Claude Junqua

IPC分类号： G10L15/26

CPC分类号： G10L15/26

摘要： A media production system includes a textual alignment module aligning multiple speech recordings to textual lines of a script based on speech recognition results. A navigation module responds to user navigation selections respective of the textual lines of the script by communicating to the user corresponding, line-specific portions of the multiple speech recordings. An editing module responds to user associations of multiple speech recordings with textual lines by accumulating line-specific portions of the multiple speech recordings in a combination recording based on at least one of relationships of textual lines in the script to the combination recording, and temporal alignments between the multiple speech recordings and the combination recording.

摘要翻译： 媒体制作系统包括文本对准模块，其基于语音识别结果将多个语音记录与脚本的文本行对齐。 导航模块通过与用户对应的多个语音记录的线特定部分通信来响应相应于脚本的文本行的用户导航选择。 编辑模块通过基于脚本中的文本行的关系与组合记录中的至少一种相结合记录来组合记录中的多个语音记录的行特定部分来累积多个语音记录与文本行的响应，以及时间对齐 在多个语音记录和组合记录之间。