公开(公告)号：US20170163797A1

公开(公告)日：2017-06-08

申请号：US15320256

申请日：2014-08-13

Applicant: ZTE CORPORATION,

Inventor： Wentao JING

IPC: H04M3/22 ,  H04W12/12 ,  G10L17/08 ,  H04W8/18 ,  H04W8/20 ,  G10L17/02 ,  H04M3/42 ,  H04W4/16

CPC classification number: H04M3/2281 ,  G06F3/16 ,  G06F3/167 ,  G06K9/00536 ,  G10L17/02 ,  G10L17/08 ,  H04M1/26 ,  H04M1/57 ,  H04M1/66 ,  H04M3/42059 ,  H04M2201/405 ,  H04W4/16 ,  H04W8/18 ,  H04W8/205 ,  H04W12/00 ,  H04W12/02 ,  H04W12/12

Abstract: The present disclosure discloses a method for identifying an identity, device and a communication terminal. The method includes that: a voiceprint feature of a current call object and a mobile phone number of the current call object are extracted; the identity of the current call object is identified according to the voiceprint feature and the mobile phone number. The present disclosure solves the problem in the related art that it is difficult to effectively identify the identity of a call object, thus providing a method for effectively identifying the identity of a current call object and technically reducing the probability of phone fraud on a user.

公开(公告)号：US09263038B2

公开(公告)日：2016-02-16

申请号：US14045118

申请日：2013-10-03

Applicant: Marchex, Inc.

Inventor： Jason Flaks ,  Ziad Ismail ,  Chris Kolbegger

IPC: G06F17/27 ,  G06F17/28 ,  G06F17/20 ,  G10L15/04 ,  G10L15/00 ,  G10L15/14 ,  G10L21/00 ,  G10L15/18 ,  G10L15/08

CPC classification number: G10L15/193 ,  G10L15/142 ,  G10L15/16 ,  G10L15/1822 ,  G10L2015/088 ,  H04M3/5175 ,  H04M2201/18 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2203/6009 ,  H04M2242/12

Abstract: A facility and method for analyzing and classifying calls without transcription via keyword spotting is disclosed. The facility uses a group of calls having known outcomes to generate one or more domain- or entity-specific grammars containing keywords and related information that are indicative of particular outcome. The facility monitors telephone calls by determining the domain or entity associated with the call, loading the appropriate grammar or grammars associated with the determined domain or entity, and tracking keywords contained in the loaded grammar or grammars that are spoken during the monitored call, along with additional information. The facility performs a statistical analysis on the tracked keywords and additional information to determine a classification for the monitored telephone call.

公开(公告)号：US09245254B2

公开(公告)日：2016-01-26

申请号：US13397289

申请日：2012-02-15

Applicant: Richard T. Lord ,  Robert W. Lord ,  Nathan P. Myhrvold ,  Clarence T. Tegreene ,  Roderick A. Hyde ,  Lowell L. Wood, Jr. ,  Muriel Y. Ishikawa ,  Victoria Y. H. Wood ,  Charles Whitmer ,  Paramvir Bahl ,  Douglas C. Burger ,  Ranveer Chandra ,  William H. Gates, III ,  Paul Holman ,  Jordin T. Kare ,  Craig J. Mundie ,  Tim Paek ,  Desney S. Tan ,  Lin Zhong ,  Matthew G. Dyor

Inventor： Richard T. Lord ,  Robert W. Lord ,  Nathan P. Myhrvold ,  Clarence T. Tegreene ,  Roderick A. Hyde ,  Lowell L. Wood, Jr. ,  Muriel Y. Ishikawa ,  Victoria Y. H. Wood ,  Charles Whitmer ,  Paramvir Bahl ,  Douglas C. Burger ,  Ranveer Chandra ,  William H. Gates, III ,  Paul Holman ,  Jordin T. Kare ,  Craig J. Mundie ,  Tim Paek ,  Desney S. Tan ,  Lin Zhong ,  Matthew G. Dyor

IPC: G10L15/00 ,  G06F3/00 ,  G06Q10/10 ,  H04N5/445 ,  H04M3/56 ,  G06F17/27 ,  G06F17/28 ,  G10L17/00 ,  H04L12/18 ,  G06F3/16 ,  G10L15/26 ,  G10L13/02 ,  H04M3/42

CPC classification number: G06Q10/10 ,  G06F3/165 ,  G06F17/2765 ,  G06F17/289 ,  G10L13/02 ,  G10L15/26 ,  G10L15/265 ,  G10L17/00 ,  H04L12/1831 ,  H04L65/403 ,  H04L67/306 ,  H04M3/42221 ,  H04M3/56 ,  H04M2201/405 ,  H04M2201/41 ,  H04N5/44543

Abstract: Techniques for ability enhancement are described. Some embodiments provide an ability enhancement facilitator system (“AEFS”) configured to enhance voice conferencing among multiple speakers. Some embodiments of the AEFS enhance voice conferencing by recording, translating and presenting voice conference history information based on speaker-related information, wherein the translation is based on language identification using multiple speech recognizers and GPS information. The AEFS receives data that represents utterances of multiple speakers who are engaging in a voice conference with one another. The AEFS then determines speaker-related information, such as by identifying a current speaker, locating an information item (e.g., an email message, document) associated with the speaker, or the like. The AEFS records conference history information (e.g., a transcript) based on the determined speaker-related information. The AEFS then informs a user of the conference history information, such as by presenting a transcript of the voice conference and/or related information items on a display of a conferencing device associated with the user.

Abstract translation: 描述了能力提升的技术。 一些实施例提供了一种配置用于增强多个扬声器之间的语音会议的能力增强辅助系统（“AEFS”）。 AEFS的一些实施例通过基于扬声器相关信息记录，翻译和呈现语音会议历史信息来增强语音会议，其中翻译是基于使用多个语音识别器和GPS信息的语言识别。 AEFS接收的数据表示多个演讲者正在进行语音会议的演讲。 然后，AEFS确定说话者相关的信息，例如通过识别当前的说话者，定位与说话者相关联的信息项（例如，电子邮件消息，文档）等。 AEFS基于所确定的说话者相关信息记录会议历史信息（例如，抄本）。 然后，AEFS向用户通知会议历史信息，例如通过在与用户相关联的会议设备的显示器上呈现语音会议和/或相关信息项的抄本。

公开(公告)号：US20150100302A1

公开(公告)日：2015-04-09

申请号：US14045118

申请日：2013-10-03

Applicant: Marchex, Inc.

Inventor： Jason Flaks ,  Ziad Ismail ,  Chris Kolbegger

IPC: G06F17/28

CPC classification number: G10L15/193 ,  G10L15/142 ,  G10L15/16 ,  G10L15/1822 ,  G10L2015/088 ,  H04M3/5175 ,  H04M2201/18 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2203/6009 ,  H04M2242/12

Abstract: A facility and method for analyzing and classifying calls without transcription via keyword spotting is disclosed. The facility uses a group of calls having known outcomes to generate one or more domain- or entity-specific grammars containing keywords and related information that are indicative of particular outcome. The facility monitors telephone calls by determining the domain or entity associated with the call, loading the appropriate grammar or grammars associated with the determined domain or entity, and tracking keywords contained in the loaded grammar or grammars that are spoken during the monitored call, along with additional information. The facility performs a statistical analysis on the tracked keywords and additional information to determine a classification for the monitored telephone call.

Abstract translation: 公开了一种通过关键词检测来分析和分类无转录的设施和方法。 该设施使用具有已知结果的一组呼叫来生成包含关键字和相关信息的一个或多个域或实体特定语法，这些语法指示特定的结果。 该设施通过确定与呼叫相关联的域或实体来监视电话呼叫，加载与所确定的域或实体相关联的适当语法或语法，以及跟踪包含在被监视呼叫期间所述的加载的语法或语法中的关键字以及 附加信息。 该设施对跟踪的关键字和附加信息进行统计分析，以确定被监视的电话呼叫的分类。

公开(公告)号：US08787534B1

公开(公告)日：2014-07-22

申请号：US13185095

申请日：2011-07-18

Applicant: Silke Witt-ehsani ,  Aaron Scott Fisher

Inventor： Silke Witt-ehsani ,  Aaron Scott Fisher

IPC: H04M1/64

CPC classification number: H04M3/5232 ,  G10L2015/088 ,  H04M3/4936 ,  H04M3/5166 ,  H04M2201/18 ,  H04M2201/405 ,  H04M2203/158 ,  H04M2203/556

Abstract: A method and apparatus of processing caller experiences is disclosed. One example method may include determining a call event type occurring during a call and assigning a weight to the call event type via a processing device. The method may also include calculating a caller experience metric value representing a caller's current call status responsive to determining the at least one call event type, the caller experience metric being a function of the current event type weight and a discounting variable that discounts a value of past events. The method may also provide comparing the caller experience metric to a predefined threshold value and determining whether to perform at least one of transferring the call to a live agent and switching from a current caller modality to a different caller modality.

Abstract translation: 公开了一种处理呼叫者体验的方法和装置。 一个示例性方法可以包括确定在呼叫期间发生的呼叫事件类型，并且经由处理设备向呼叫事件类型分配权重。 该方法还可以包括响应于确定至少一个呼叫事件类型来计算表示呼叫者当前呼叫状态的呼叫者体验度量值，呼叫者体验度量是当前事件类型权重的函数，以及折扣值 过去的事。 该方法还可以提供将呼叫者体验度量与预定义的阈值进行比较，并且确定是否执行将呼叫转移到实时代理和从当前呼叫者模式切换到不同呼叫者模式中的至少一个。

公开(公告)号：US20130144603A1

公开(公告)日：2013-06-06

申请号：US13397289

申请日：2012-02-15

Applicant: Richard T. Lord ,  Robert W. Lord ,  Nathan P. Myhrvold ,  Clarence T. Tegreene ,  Roderick A. Hyde ,  Lowell L. Wood, JR. ,  Muriel Y. Ishikawa ,  Victoria Y.H. Wood ,  Charles Whitmer ,  Paramvir Bahl ,  Douglas C. Burger ,  Ranveer Chandra ,  William H. Gates, III ,  Paul Holman ,  Jordin T. Kare ,  Craig J. Mundie ,  Tim Paek ,  Desney S. Tan ,  Lin Zhong ,  Matthew G. Dyor

Inventor： Richard T. Lord ,  Robert W. Lord ,  Nathan P. Myhrvold ,  Clarence T. Tegreene ,  Roderick A. Hyde ,  Lowell L. Wood, JR. ,  Muriel Y. Ishikawa ,  Victoria Y.H. Wood ,  Charles Whitmer ,  Paramvir Bahl ,  Douglas C. Burger ,  Ranveer Chandra ,  William H. Gates, III ,  Paul Holman ,  Jordin T. Kare ,  Craig J. Mundie ,  Tim Paek ,  Desney S. Tan ,  Lin Zhong ,  Matthew G. Dyor

IPC: G10L21/00 ,  G06F17/27 ,  G10L15/26

CPC classification number: G06Q10/10 ,  G06F3/165 ,  G06F17/2765 ,  G06F17/289 ,  G10L13/02 ,  G10L15/26 ,  G10L15/265 ,  G10L17/00 ,  H04L12/1831 ,  H04L65/403 ,  H04L67/306 ,  H04M3/42221 ,  H04M3/56 ,  H04M2201/405 ,  H04M2201/41 ,  H04N5/44543

Abstract: Techniques for ability enhancement are described. Some embodiments provide an ability enhancement facilitator system (“AEFS”) configured to enhance voice conferencing among multiple speakers. Some embodiments of the AEFS enhance voice conferencing by recording and presenting voice conference history information based on speaker-related information. The AEFS receives data that represents utterances of multiple speakers who are engaging in a voice conference with one another. The AEFS then determines speaker-related information, such as by identifying a current speaker, locating an information item (e.g., an email message, document) associated with the speaker, or the like. The AEFS records conference history information (e.g., a transcript) based on the determined speaker-related information. The AEFS then informs a user of the conference history information, such as by presenting a transcript of the voice conference and/or related information items on a display of a conferencing device associated with the user.

Abstract translation: 描述了能力提升的技术。 一些实施例提供了一种配置用于增强多个扬声器之间的语音会议的能力增强辅助系统（“AEFS”）。 AEFS的一些实施例通过基于说话者相关信息记录和呈现语音会议历史信息来增强语音会议。 AEFS接收的数据表示多个演讲者正在进行语音会议的演讲。 然后，AEFS确定说话者相关的信息，例如通过识别当前的说话者，定位与说话者相关联的信息项（例如，电子邮件消息，文档）等。 AEFS基于所确定的说话者相关信息记录会议历史信息（例如，抄本）。 然后，AEFS向用户通知会议历史信息，例如通过在与用户相关联的会议设备的显示器上呈现语音会议和/或相关信息项的抄本。

公开(公告)号：US08238525B2

公开(公告)日：2012-08-07

申请号：US12693796

申请日：2010-01-26

Applicant: Zhipeng Zhang ,  Hirotaka Furukawa

Inventor： Zhipeng Zhang ,  Hirotaka Furukawa

IPC: H04M1/64 ,  G10L15/26

CPC classification number: H04M3/42153 ,  G10L15/183 ,  G10L15/30 ,  G10L2015/228 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2250/66

Abstract: A voice recognition server 200 has a voice reception unit 202 which receives a voice from a telephone equipment 100, a model storage unit 208 which stores at least one acoustic model and at least one language model used for converting the voice received by the voice reception unit 202, to character data, a number decision unit 204 which decides a current calling number and a second number of the telephone equipment 100, a model selection unit 206 which selects an acoustic model stored in the model storage unit 208, based on the current calling number and the second number, and which selects a language model stored in the model storage unit 208, based on the current calling number, and a voice recognition unit 210 which converts the voice received by the voice reception unit 202, to character data, based on the acoustic model and the language model selected by the model selection unit 206.

Abstract translation: 语音识别服务器200具有从电话设备100接收语音的语音接收单元202，存储至少一个声学模型的模型存储单元208和用于转换由语音接收单元接收的语音的至少一种语言模型 202，对于字符数据，确定当前主叫号码的号码决定单元204和电话设备100的第二号码，选择存储在模型存储单元208中的声学模型的模型选择单元206，基于当前呼叫 基于当前主叫号码选择存储在模型存储单元208中的语言模型和将由语音接收单元202接收的语音转换为基于字符数据的语音识别单元210 在由模型选择单元206选择的声学模型和语言模型上。

公开(公告)号：US20120155663A1

公开(公告)日：2012-06-21

申请号：US12969622

申请日：2010-12-16

Applicant: Adam WEINBERG ,  Irit Opher ,  Ruth Aloni-Lavi ,  Eyal Kolman ,  Ido Azriel

Inventor： Adam WEINBERG ,  Irit Opher ,  Ruth Aloni-Lavi ,  Eyal Kolman ,  Ido Azriel

IPC: H04R29/00

CPC classification number: G10L17/00 ,  G10L17/26 ,  H04L63/304 ,  H04M3/2281 ,  H04M2201/405 ,  H04M2201/41 ,  H04M2203/301 ,  H04R27/00 ,  H04S2400/15

Abstract: A method for spotting an interaction in which a target speaker associated with a current index or current interaction speaks, the method comprising: receiving an interaction and an index associated with the interaction, the index associated with additional data; receiving the current interaction or current index associated with the target speaker; obtaining current data associated with the current interaction or current index; filtering the index using the additional data, in accordance with the current data associated with the current interaction or current index, and obtaining a matching index; and comparing the current index or a representation of the current interaction with the matching index to obtain a target speaker index.

Abstract translation: 用于发现与当前索引或当前交互相关联的目标说话者的交互的方法，所述方法包括：接收与所述交互相关联的交互和与附加数据相关联的索引; 接收与目标扬声器相关联的当前交互或当前索引; 获取与当前交互或当前索引相关联的当前数据; 根据与当前交互或当前索引相关联的当前数据，使用附加数据过滤索引，并获得匹配索引; 并且将当前索引或当前交互的表示与匹配索引进行比较以获得目标说话者索引。

公开(公告)号：US20100195806A1

公开(公告)日：2010-08-05

申请号：US12693796

申请日：2010-01-26

Applicant: Zhipeng ZHANG ,  Hirotaka Furukawa

Inventor： Zhipeng ZHANG ,  Hirotaka Furukawa

IPC: H04M1/64

CPC classification number: H04M3/42153 ,  G10L15/183 ,  G10L15/30 ,  G10L2015/228 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2250/66

Abstract: A voice recognition server 200 has a voice reception unit 202 which receives a voice from a telephone equipment 100, a model storage unit 208 which stores at least one acoustic model and at least one language model used for converting the voice received by the voice reception unit 202, to character data, a number decision unit 204 which decides a current calling number and a second number of the telephone equipment 100, a model selection unit 206 which selects an acoustic model stored in the model storage unit 208, based on the current calling number and the second number, and which selects a language model stored in the model storage unit 208, based on the current calling number, and a voice recognition unit 210 which converts the voice received by the voice reception unit 202, to character data, based on the acoustic model and the language model selected by the model selection unit 206.

Abstract translation: 语音识别服务器200具有从电话设备100接收语音的语音接收单元202，存储至少一个声学模型的模型存储单元208和用于转换由语音接收单元接收的语音的至少一种语言模型 202，对于字符数据，确定当前主叫号码的号码决定单元204和电话设备100的第二号码，选择存储在模型存储单元208中的声学模型的模型选择单元206，基于当前呼叫 基于当前主叫号码选择存储在模型存储单元208中的语言模型和将由语音接收单元202接收的语音转换为基于字符数据的语音识别单元210 在由模型选择单元206选择的声学模型和语言模型上。

公开(公告)号：US07693267B2

公开(公告)日：2010-04-06

申请号：US11324208

申请日：2005-12-30

Applicant: David Andrew Howell ,  Sridhar Sundararaman ,  David T. Fong ,  Frank Torsten Bernd Seide

Inventor： David Andrew Howell ,  Sridhar Sundararaman ,  David T. Fong ,  Frank Torsten Bernd Seide

IPC: H04M11/00

CPC classification number: H04M3/5307 ,  H04M2201/405 ,  H04M2201/60 ,  H04M2203/253 ,  H04M2203/4509

Abstract: Improved systems and methods are provided for transcribing audio files of voice mails sent over a unified messaging system. Customized grammars specific to a voice mail recipient are created and utilized to transcribe a received voice mail by comparing the audio file to commonly utilized words, names, acronyms, and phrases used by the recipient. Key elements are identified from the resulting text transcription to aid the recipient in processing received voice mails based on the significant content contained in the voice mail.

Abstract translation: 提供了改进的系统和方法，用于转录通过统一消息系统发送的语音邮件的音频文件。 通过将音频文件与收件人使用的常用词，名称，首字母缩略词和短语进行比较，创建和利用特定于语音邮件收件人的定制语法来转录接收到的语音邮件。 从所得到的文本转录中识别关键要素，以帮助接收者基于语音邮件中包含的重要内容来处理接收的语音邮件。