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    Network and method for providing a calling name telecommunications service with automatic speech recognition capability
    381.
    再颁专利
    Network and method for providing a calling name telecommunications service with automatic speech recognition capability 有权
    提供具有自动语音识别功能的呼叫名称电信业务的网络和方法

    公开(公告)号：USRE42539E1

    公开(公告)日：2011-07-12

    申请号：US11727638

    申请日：2007-03-27

    Applicant: Zeeman Zhang ,  Joseph Knoerle

    Inventor： Zeeman Zhang ,  Joseph Knoerle

    IPC: H04M1/64

    CPC classification number: H04M3/42204 ,  H04M2201/41 ,  H04M2203/2011 ,  H04Q3/0033

    Abstract: A network for providing a telecommunications service with automatic speech recognition to a telecommunications user, including a switch in communication with a telecommunications device associated with the telecommunications user for detecting a terminating trigger specific to the telecommunications service in response to an incoming communication to the telecommunications device from a calling party, and an intelligent resource server in communication with the switch for receiving the incoming communication from the switch, for placing an outgoing communication to the telecommunications device via the switch, the outgoing communication including an audible message identifying the calling party, and for automatically recognizing a predetermined keyword spoken by the telecommunications user in response to the outgoing communication.

    Abstract translation: 一种用于向电信用户提供具有自动语音识别的电信服务的网络，包括与与电信用户相关联的电信设备通信的交换机，用于响应于到电信设备的进入通信来检测特定于电信服务的终止触发 以及与交换机通信的智能资源服务器，用于从交换机接收进入的通信，用于经由交换机向电信设备发送传出通信，所述传出通信包括识别主叫方的可听消息，以及 用于自动识别由电信用户响应于传出通信所说出的预定关键字。

    Methods and systems for participant sourcing indication in multi-party conferencing and for audio source discrimination
    382.
    发明授权
    Methods and systems for participant sourcing indication in multi-party conferencing and for audio source discrimination 有权
    用于多方会议和音频源歧视的参与者采购指示的方法和系统

    公开(公告)号：US07916848B2

    公开(公告)日：2011-03-29

    申请号：US10677213

    申请日：2003-10-01

    Applicant: Yong Rui ,  Anoop Gupta

    Inventor： Yong Rui ,  Anoop Gupta

    IPC: H04M3/42 ,  H04M1/64 ,  G10L13/00

    CPC classification number: H04L65/4046 ,  H04L29/06027 ,  H04M3/56 ,  H04M2201/38 ,  H04M2201/41 ,  H04Q2213/13034 ,  H04Q2213/13091 ,  H04Q2213/1324 ,  H04Q2213/13248 ,  H04Q2213/13333 ,  H04Q2213/13337 ,  H04Q2213/13389

    Abstract: Indications of which participant is providing information during a multi-party conference. Each participant has equipment to display information being transferred during the conference. A sourcing signaler residing in the participant equipment provides a signal that indicates the identity of its participant when this participant is providing information to the conference. The source indicators of the other participant equipment receive the signal and cause a UI to indicate that the participant identified by the received signal is providing information (e.g. the UI can causes the identifier to change appearance). An audio discriminator is used to distinguish between an acoustic signal that was generated by a person speaking from that generated in a band-limited manner. The audio discriminator analyzes the spectrum of detected audio signals and generates several parameters from the spectrum and from past determinations to determine the source of an audio signal on a frame-by-frame basis.

    Abstract translation: 参与者在多方会议期间提供信息的指示。 每个参与者都有设备在会议期间显示要传送的信息。 驻留在参与者设备中的采购信号器提供当该参与者向会议提供信息时指示其参与者身份的信号。 其他参与者设备的源指示符接收信号并使UI指示由接收到的信号识别的参与者提供信息（例如，UI可以使得标识符改变外观）。 使用音频鉴别器来区分由以频带限制的方式产生的声音所产生的声信号。 音频鉴别器分析检测到的音频信号的频谱，并从频谱和过去的确定中产生几个参数，以逐帧确定音频信号的来源。

    FACILITATION OF A CONFERENCE CALL
    383.
    发明申请
    FACILITATION OF A CONFERENCE CALL 有权
    召开会议召集

    公开(公告)号：US20110069140A1

    公开(公告)日：2011-03-24

    申请号：US12470319

    申请日：2009-05-21

    Applicant: William Charles Gormley Ortel

    Inventor： William Charles Gormley Ortel

    IPC: H04N7/14

    CPC classification number: H04M3/56 ,  H04M3/567 ,  H04M2201/38 ,  H04M2201/41

    Abstract: There is provided a system for facilitating a conference call. The system includes a module to generate a real-time voiceprint from a voice input of a participant in the conference call, and a module to provide information indicative of the participant based on the real-time voiceprint.

    Abstract translation: 提供了一种便于电话会议的系统。 该系统包括用于从会议呼叫中的参与者的语音输入生成实时声纹的模块，以及用于基于实时声纹提供指示参与者的信息的模块。

    Apparatus and methods for implementing voice enabling applications in a converged voice and data network environment
    386.
    发明授权
    Apparatus and methods for implementing voice enabling applications in a converged voice and data network environment 有权
    在融合语音和数据网络环境中实现语音使能应用的装置和方法

    公开(公告)号：US07805310B2

    公开(公告)日：2010-09-28

    申请号：US10469110

    申请日：2002-02-26

    Applicant: Elizabeth A. Rohwer

    Inventor： Elizabeth A. Rohwer

    IPC: G10L21/00

    CPC classification number: H04M7/006 ,  G10L15/22 ,  G10L15/30 ,  H04M2201/41 ,  H04M2201/42

    Abstract: Human speech is transported through a voice and data converged Internet network to recognize its content, verify the identity of the speaker, or to verify the content of a spoken phrase by utilizing the Internet protocol to transmit voice packets. The voice data (4) entered is processed and transmitted in the same way as Internet data packets over converged voice and data IP networks. A voice-enabled application sends a message (5), which is decoded by the speech API (2) and the appropriate control and synchronization information is issued (7) to the data preparation module (9) and to the speech engine (3). Standard voice over IP includes a speech compression algorithm and the use of RTP (Real Time Protocol), enabling additional processing of the human voice anywhere in the network to perform speaker verification, with or without the knowledge of the speaker.

    Abstract translation: 人类语音通过语音和数据融合的互联网传输，以识别其内容，验证说话者的身份，或通过利用因特网协议来传输语音数据包来验证口头短语的内容。 输入的语音数据（4）以与互联网数据包相同的方式通过融合的语音和数据IP网络进行处理和传输。 语音应用发送由语音API（2）解码的消息（5），并且向数据准备模块（9）和语音引擎（3）发出适当的控制和同步信息（7） 。 标准IP语音包括语音压缩算法和RTP（实时协议）的使用，能够在或不用说话者的知识的情况下在网络中的任何地方进行人声的附加处理来执行说话者验证。

    Method and apparatus for providing voice control for accessing teleconference services
    388.
    发明授权
    Method and apparatus for providing voice control for accessing teleconference services 有权
    用于提供语音控制以访问电话会议服务的方法和装置

    公开(公告)号：US07593520B1

    公开(公告)日：2009-09-22

    申请号：US11294322

    申请日：2005-12-05

    Applicant: Marian Croak ,  Hossein Eslambolchi

    Inventor： Marian Croak ,  Hossein Eslambolchi

    IPC: H04M3/42

    CPC classification number: H04M3/56 ,  H04M3/385 ,  H04M2201/41 ,  H04M2207/20

    Abstract: A method and apparatus for providing access to teleconference services using voice recognition technology to receive information on packet networks such as Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) networks are disclosed. In one embodiment, the service provider enables a caller to enter access information for accessing a conference service using at least one natural language response.

    Abstract translation: 公开了一种用于使用语音识别技术访问电话会议服务以接收诸如因特网协议语音（VoIP）和因特网协议（SoIP）网络之类的分组网络上的信息的方法和装置。 在一个实施例中，服务提供商使呼叫者能够使用至少一个自然语言响应来输入访问会议服务的访问信息。

    Method and apparatus for active speaker selection using microphone arrays and speaker recognition
    389.
    发明申请
    Method and apparatus for active speaker selection using microphone arrays and speaker recognition 有权
    用于使用麦克风阵列和扬声器识别的主动扬声器选择的方法和装置

    公开(公告)号：US20090220065A1

    公开(公告)日：2009-09-03

    申请号：US12074276

    申请日：2008-03-03

    Applicant: Sudhir Raman Ahuja ,  Jingdong Chen ,  Yiteng Arden Huang ,  Dong Liu ,  Qiru Zhou

    Inventor： Sudhir Raman Ahuja ,  Jingdong Chen ,  Yiteng Arden Huang ,  Dong Liu ,  Qiru Zhou

    IPC: H04M3/56

    CPC classification number: H04M3/569 ,  G10L17/00 ,  G10L2021/02166 ,  H04M2201/41 ,  H04M2203/5072

    Abstract: A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface. The remote participants may then select one of the identified speakers, and the speech signal selection module then selects for transmission the speech signal associated with the selected identified speaker, thereby enabling the participants at the remote location to listen to the selected speaker and neglect the speech from other active speakers.

    Abstract translation: 用于在电话会议应用中执行主动扬声器选择的方法和装置示例性地包括麦克风阵列模块，扬声器识别系统，用户界面和语音信号选择模块。 麦克风阵列模块将来自每个有源扬声器的语音信号与其他有源扬声器的语音信号分离，从而提供多个单独的扬声器的语音信号。 扬声器识别系统使用常规扬声器识别/识别技术识别每个当前有效的扬声器。 然后将这些身份发送到远程电话会议位置，以通过用户界面向远程参与者显示。 然后，远程参与者可以选择所识别的扬声器中的一个，并且语音信号选择模块然后选择用于传输与所选择的所识别的扬声器相关联的语音信号，从而使远程位置的参与者能够听取所选择的说话者并忽略语音 从其他有源音箱

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