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41. 发明申请

US20050182630A1 Multilingual text-to-speech system with limited resources 有权
标题翻译：具有有限资源的多语言文字到语音系统
公开(公告)号：US20050182630A1
公开(公告)日：2005-08-18
申请号：US10771256
申请日：2004-02-02
申请人： Xavier Miro , Peter Veprek , Jean-Claude Junqua
发明人： Xavier Miro , Peter Veprek , Jean-Claude Junqua
IPC分类号： G10L13/00 , G10L13/08
CPC分类号： G10L13/08
摘要： A multilingual text-to-speech system includes a source datastore of primary source parameters providing information about a speaker of a primary language. A plurality of primary filter parameters provides information about sounds in the primary language. A plurality of secondary filter parameters provides information about sounds in a secondary language. One or more secondary filter parameters is normalized to the primary filter parameters and mapped to a primary source parameter.
摘要翻译：多语言文本到语音系统包括提供关于主要语言的说话者的信息的主要源参数的源数据库。多个主要滤波器参数提供关于主要语言的声音的信息。多个次级滤波器参数提供关于次要语言的声音的信息。将一个或多个二次滤波器参数归一化为主滤波器参数并映射到主要源参数。

42. 发明申请

US20050075881A1 Voice tagging, voice annotation, and speech recognition for portable devices with optional post processing 有权
标题翻译：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
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。

43. 发明授权

US06792407B2 Text selection and recording by feedback and adaptation for development of personalized text-to-speech systems 有权
标题翻译：通过反馈和适应的文本选择和记录来开发个性化的文本到语音系统
公开(公告)号：US06792407B2
公开(公告)日：2004-09-14
申请号：US09821973
申请日：2001-03-30
申请人： Nicholas Kibre , Steven Pearson , Brian Hanson , Jean-Claude Junqua
发明人： Nicholas Kibre , Steven Pearson , Brian Hanson , Jean-Claude Junqua
IPC分类号： G10L1306
CPC分类号： G10L13/047 , G10L13/04
摘要： A new speaker provides speech from which comparison snippets are extracted. The comparison snippets are compared with initial snippets stored in a recorded snippet database that is associated with a concatenative synthesizer. The comparison of the snippets to the initial snippets produces required sound units. A greedy selection algorithm is performed with the required sound units for identifying the smallest subset of the input text that contains all of the text for the new speaker to read. The new speaker then reads the optimally selected text and sound units are extracted from the human speech such that the recorded snippet database is modified and the speech synthesized adopts the voice quality and characteristics of the new speaker.
摘要翻译：一个新的演讲者提供了从中提取比较片段的演讲。将比较片段与存储在与级联合成器相关联的记录片段数据库中的初始片段进行比较。代码段与初始代码段的比较会产生所需的声音单位。使用所需的声音单元执行贪婪选择算法，用于识别包含要被读取的新发言者的所有文本的输入文本的最小子集。然后，新的扬声器读取最佳选择的文本，并且从人类语音中提取声音单元，使得记录的片段数据库被修改并且合成的语音采用新的说话者的语音质量和特征。

44. 发明授权

US06496801B1 Speech synthesis employing concatenated prosodic and acoustic templates for phrases of multiple words 有权
标题翻译：语音合成采用多个词短语的连接韵律和声学模板
公开(公告)号：US06496801B1
公开(公告)日：2002-12-17
申请号：US09432876
申请日：1999-11-02
申请人： Peter Veprek , Steve Pearson , Jean-Claude Junqua
发明人： Peter Veprek , Steve Pearson , Jean-Claude Junqua
IPC分类号： G10L1308
CPC分类号： G10L13/08
摘要： A speech synthesis system for generating voice dialog for a message frame having a fixed and a variable portion. A prosody module selects a prosodic template for each of the fixed and variable portions wherein at least one portion comprises a phrase of multiple words. An acoustic module selects an acoustic template for each of the fixed and variable portions wherein at least one portion comprises a phrase of multiple words. A frame generator concatenates the respective prosodic templates and acoustic templates. A sound module generates the voice dialog in accordance with the concatenated prosodic and acoustic templates.
摘要翻译：一种用于产生具有固定和可变部分的消息帧的语音对话的语音合成系统。韵律模块为每个固定和可变部分选择韵律模板，其中至少一个部分包括多个词的短语。声学模块为每个固定和可变部分选择声学模板，其中至少一个部分包括多个单词的短语。帧发生器连接相应的韵律模板和声学模板。声音模块根据连接的韵律和声学模板产生语音对话。

45. 发明授权

US07089182B2 Method and apparatus for feature domain joint channel and additive noise compensation 有权
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.

46. 发明授权

US07062339B2 Method and apparatus for communicating between a portable device and a server 有权
公开(公告)号：US07062339B2
公开(公告)日：2006-06-13
申请号：US10141340
申请日：2002-05-09
申请人： John K. Howard , Dwayne Escola , Jim Pollock , Jean-Claude Junqua
发明人： John K. Howard , Dwayne Escola , Jim Pollock , Jean-Claude Junqua
IPC分类号： G10L21/00 , G10L11/00 , G06F17/00
CPC分类号： G05B19/042 , G05B2219/23363 , G05B2219/23386 , G10L15/26 , G10L2015/223
摘要： A control system including a portable device and a server. The portable device includes: (1) a body; (2) a microphone for receiving a first audio data; (3) an audio coder for converting the first audio data to first audio data signals; (4) an optical sensor for reading a first optical data; (5) an optical coder for converting the first optical data to first optical data signals; and (6) a transmitter for transmitting at least the first audio data signals or the first optical data signals to the server. The server includes: (1) a communication device for receiving at least first audio data signals or the first optical data signals transmitted by the transmitter; (2) a data analyzer for extracting semantic information from the first audio data signal and for converting the semantic information into an audio command; (3) an optical analyzer for converting the first optical data to an optical command; (4) a memory device for storing programs to be performed; and (5) a CPU for receiving the audio command and the optical command, retrieving a program associated with the audio command or the optical command and for executing the retrieved program.

47. 发明授权

US06996528B2 Method for efficient, safe and reliable data entry by voice under adverse conditions 失效
公开(公告)号：US06996528B2
公开(公告)日：2006-02-07
申请号：US09921766
申请日：2001-08-03
申请人： Philippe R. Morin , Jean-Claude Junqua , Luca Rigazio , Robert C. Boman , Peter Veprek
发明人： Philippe R. Morin , Jean-Claude Junqua , Luca Rigazio , Robert C. Boman , Peter Veprek
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.

48. 发明申请

US20050286705A1 Intelligent call routing and call supervision method for call centers 有权
标题翻译：呼叫中心的智能呼叫路由和呼叫监控方法
公开(公告)号：US20050286705A1
公开(公告)日：2005-12-29
申请号：US10869810
申请日：2004-06-16
申请人： Matteo Contolini , Jean-Claude Junqua
发明人： Matteo Contolini , Jean-Claude Junqua
IPC分类号： H04M1/64 , H04M3/00 , H04M3/523 , H04M5/00
CPC分类号： H04M3/5232 , H04M2203/2061
摘要： A call routing and supervising system includes an input receiving customer speech from a remote location, and a voice characteristics extractor extracting voice characteristics from the customer speech, such as language/dialect/accent, age group, gender, and eigendimension coordinates. A customer service representative selector selects one or more customer service representatives based on profiles of the customer service representatives respective of customers having voice characteristics similar to the extracted voice characteristics. In other aspects, a call monitor automatically analyzes dialogue between the customer and the customer service representative, such as detected interruptions, tracked dialogue turns, and recognized key phrases indicating frustration, polity, and/or resolution characteristics of dialogue. The call monitor records performance of the customer service representative respective of customers having the voice characteristics. Automatic call rerouting and/or real-time instruction of call center personnel can also be accomplished based on analysis results.
摘要翻译：呼叫路由和监督系统包括从远程位置接收客户语音的输入和从客户语音提取语音特征的语音特征提取器，例如语言/方言/口音，年龄组，性别和特征坐标。客户服务代表选择器基于具有与所提取的语音特征相似的语音特征的客户的客户服务代表的简档来选择一个或多个客户服务代表。在其他方面，呼叫监视器自动分析客户和客户服务代表之间的对话，诸如检测到的中断，跟踪的对话转弯以及表示对话的挫折，政治和/或解析特征的识别的关键短语。呼叫监视器记录具有语音特征的客户的客户服务代表的性能。呼叫中心人员的自动呼叫转接和/或实时指令也可以基于分析结果完成。

49. 发明申请

US20050228663A1 Media production system using time alignment to scripts 审中-公开
标题翻译：媒体制作系统使用时间对齐脚本
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。

50. 发明授权

US06691091B1 Method for additive and convolutional noise adaptation in automatic speech recognition using transformed matrices 有权
标题翻译：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
申请人： Christophe Cerisara , Luca Rigazio , Robert Boman , Jean-Claude Junqua
发明人： Christophe Cerisara , Luca Rigazio , Robert Boman , Jean-Claude Junqua
IPC分类号： G10L1506
CPC分类号： G10L15/063 , G10L15/20 , G10L21/0216
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。

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