Method of synthesizing pronunciation transcriptions for English sentence
patterns/words by a computer
    1.
    发明授权
    Method of synthesizing pronunciation transcriptions for English sentence patterns/words by a computer 失效
    通过计算机合成英文句子/单词的发音转录的方法

    公开(公告)号:US6088666A

    公开(公告)日:2000-07-11

    申请号:US901691

    申请日:1997-07-28

    CPC分类号: G10L13/08

    摘要: A method of synthesizing pronunciation transcriptions for English sentence patterns/words through a computer, including the step of searching out matching rules sign for every individual letter or letter series from a pronunciation rules chart set in the computer subject to the location of every individual letter or letter series in the word and its relationship with the neighbor letters or letter series, the step of searching out the corresponding IPA pronunciation symbols for every individual letter of the word from a pronunciation rules data bank set in the computer, and the step of synthesizing the pronunciation symbols for the individual letters of the word into a pronunciation transcription.

    摘要翻译: 一种通过计算机合成英文句子模式/单词的发音转录的方法,包括从计算机中设置的发音规则图中搜索每个单个字母或字母系列的匹配规则符号的步骤,受每个单独字母的位置的影响,或 字母系列及其与相邻字母或字母系列的关系,从设置在计算机中的发音规则数据库中搜索单词的每个单独字母的相应IPA发音符号的步骤,以及合成 发音符号将单词的单词转化为发音转录。

    System and method for encoding telephone call data using varying codec algorithms
    2.
    发明授权
    System and method for encoding telephone call data using varying codec algorithms 有权
    使用不同的编解码算法对电话呼叫数据进行编码的系统和方法

    公开(公告)号:US09312983B2

    公开(公告)日:2016-04-12

    申请号:US13355803

    申请日:2012-01-23

    摘要: A method and system for transmitting a call over a packet switched network. A gateway server receives a telephone call and converts analog voice signals associated with the telephone call to a stream of digital data. A first part of the digital data is processed using a first codec algorithm and transmitted over the packet switched network. A change in network conditions is detected. A second part of the digital data is processed using a second codec algorithm and transmitted over the packet switched network.

    摘要翻译: 一种用于通过分组交换网络发送呼叫的方法和系统。 网关服务器接收电话呼叫并将与电话呼叫相关联的模拟语音信号转换成数字数据流。 使用第一编解码算法处理数字数据的第一部分并通过分组交换网络传输。 检测到网络状况的变化。 使用第二编解码算法处理数字数据的第二部分,并通过分组交换网络传输。

    System and method for transmitting a telephone call over the internet
    3.
    发明授权
    System and method for transmitting a telephone call over the internet 有权
    通过因特网发送电话的系统和方法

    公开(公告)号:US08238330B2

    公开(公告)日:2012-08-07

    申请号:US13114050

    申请日:2011-05-24

    IPC分类号: H04L12/66 H04L12/28 H04M1/24

    摘要: A method and system for transmitting a call in a client/server architecture. A client device initiates a telephone call and converts first analog voice signals associated with the telephone call to digital signals. The digital signals are then transmitted over the Internet to a first gateway server. The first gateway server processes the digital signals using a codec algorithm and transmits the processed digital signals over the Internet to a second gateway server. The second gateway server converts the processed digital signals to second analog voice signals and transmits the second analog voice signals over a public switched telephone network.

    摘要翻译: 一种用于在客户机/服务器架构中发送呼叫的方法和系统。 客户端设备启动电话呼叫,并将与电话呼叫相关联的第一模拟语音信号转换为数字信号。 数字信号然后通过因特网传输到第一网关服务器。 第一网关服务器使用编解码算法处理数字信号,并通过因特网将处理的数字信号发送到第二网关服务器。 第二网关服务器将经处理的数字信号转换为第二模拟语音信号,并通过公共交换电话网络发送第二模拟语音信号。

    System and method for encoding telephone call data using varying codec algorithms
    4.
    发明申请
    System and method for encoding telephone call data using varying codec algorithms 审中-公开
    使用不同的编解码算法对电话呼叫数据进行编码的系统和方法

    公开(公告)号:US20120127988A1

    公开(公告)日:2012-05-24

    申请号:US13355803

    申请日:2012-01-23

    IPC分类号: H04L12/66

    摘要: A method and system for transmitting a call over a packet switched network. A gateway server receives a telephone call and converts analog voice signals associated with the telephone call to a stream of digital data. A first part of the digital data is processed using a first codec algorithm and transmitted over the packet switched network. A change in network conditions is detected. A second part of the digital data is processed using a second codec algorithm and transmitted over the packet switched network.

    摘要翻译: 一种用于通过分组交换网络发送呼叫的方法和系统。 网关服务器接收电话呼叫并将与电话呼叫相关联的模拟语音信号转换成数字数据流。 使用第一编解码算法处理数字数据的第一部分并通过分组交换网络传输。 检测到网络状况的变化。 使用第二编解码算法处理数字数据的第二部分,并通过分组交换网络传输。

    LASER SYSTEM WITH MULTIPLE OPERATING MODES AND WORK STATION USING SAME
    5.
    发明申请
    LASER SYSTEM WITH MULTIPLE OPERATING MODES AND WORK STATION USING SAME 审中-公开
    具有多个操作模式和工作站的激光系统

    公开(公告)号:US20060289411A1

    公开(公告)日:2006-12-28

    申请号:US11426266

    申请日:2006-06-23

    IPC分类号: B23K26/06 G06F19/00

    CPC分类号: B23K26/0622

    摘要: A method for manufacturing applied to workpieces, such as large flat-panel liquid crystal displays (LCDs) and the like, including identifying and classifying targets on the workpiece, mounting workpiece on a stage, and controlling a laser to generate pulse of light on a single beam line that are adapted to the classification of the target. The laser includes a short pulse mode and a long pulse mode, and provides selectable wavelengths, which are adapted to particular operations on the target. The pulses of light are delivered in both of the first and second modes on the single beam line through an optical system to the targets on the workpiece.

    摘要翻译: 一种应用于工件的制造方法,例如大型平板液晶显示器(LCD)等,其包括对工件上的目标进行识别和分类,将工件安装在台架上,以及控制激光器以产生光的脉冲 适用于目标分类的单光束线。 激光器包括短脉冲模式和长脉冲模式,并提供可选择的波长,其适用于目标上的特定操作。 通过光学系统将单个光束线上的第一和第二模式的光脉冲输送到工件上的目标。

    Dynamic forward error correction algorithm for internet telephone
    6.
    发明授权
    Dynamic forward error correction algorithm for internet telephone 失效
    互联网电话动态前向纠错算法

    公开(公告)号:US06167060A

    公开(公告)日:2000-12-26

    申请号:US907686

    申请日:1997-08-08

    摘要: An Internet telephone system architecture having low latency and permitting voice communication between telephones and computers is disclosed. The architecture permits dynamic packet-to-packet change in various factors to adjust for Internet conditions. A forward error correction algorithm provides a variable level of redundancy from zero to three in the transmission of data packets. The level of error correction redundancy, the codec selection and other factors, are dynamically changeable by a voice port to replace lost packets without interpolation and to thereby maintain the highest voice quality and data compression ratio that is consistent with the quantity of packet loss.

    摘要翻译: 公开了具有低等待时间并且允许电话和计算机之间的语音通信的因特网电话系统架构。 该架构允许在各种因素中动态分组到分组的变化来适应因特网条件。 前向纠错算法在数据包的传输中提供从零到三的可变级别的冗余。 纠错冗余度,编解码器选择等因素,可以通过语音端口动态改变,以便不插值地替换丢失的分组,从而保持与分组丢失量一致的最高语音质量和数据压缩比。

    System architecture for internet telephone
    7.
    发明授权
    System architecture for internet telephone 有权
    互联网电话系统架构

    公开(公告)号:US08032808B2

    公开(公告)日:2011-10-04

    申请号:US10906598

    申请日:2005-02-25

    摘要: The invention is concerned with an Internet telephone system having a client/server architecture and providing voice communication between client stations over the Internet through gateway servers. The system includes a plurality of software modules within each of the gateway servers for performing digital signal processing (DSP), and an account manager placed at an arbitrary location on the Internet for monitoring transactions between client stations to produce billing information. The system is characterized by low latency, full duplex voice communication, and permits telephone to telephone or PC to telephone connections.

    摘要翻译: 本发明涉及具有客户端/服务器架构的因特网电话系统,并且通过网关服务器在因特网上的客户站之间提供语音通信。 该系统包括用于执行数字信号处理(DSP)的每个网关服务器内的多个软件模块,以及放置在因特网上的任意位置的客户管理器,用于监视客户站之间的事务以产生计费信息。 该系统的特点是低延迟,全双工语音通信,并允许电话到电话或PC到电话连接。

    System and method for transmitting a telephone call over the Internet
    8.
    发明申请
    System and method for transmitting a telephone call over the Internet 有权
    通过互联网发送电话的系统和方法

    公开(公告)号:US20110222530A1

    公开(公告)日:2011-09-15

    申请号:US13114050

    申请日:2011-05-24

    IPC分类号: H04L12/66

    摘要: A method and system for transmitting a call in a client/server architecture. A client device initiates a telephone call and converts first analog voice signals associated with the telephone call to digital signals. The digital signals are then transmitted over the Internet to a first gateway server. The first gateway server processes the digital signals using a codec algorithm and transmits the processed digital signals over the Internet to a second gateway server. The second gateway server converts the processed digital signals to second analog voice signals and transmits the second analog voice signals over a public switched telephone network.

    摘要翻译: 一种用于在客户机/服务器架构中发送呼叫的方法和系统。 客户端设备启动电话呼叫,并将与电话呼叫相关联的第一模拟语音信号转换为数字信号。 数字信号然后通过因特网传输到第一网关服务器。 第一网关服务器使用编解码算法处理数字信号,并通过因特网将处理的数字信号发送到第二网关服务器。 第二网关服务器将经处理的数字信号转换为第二模拟语音信号,并通过公共交换电话网络发送第二模拟语音信号。

    Receiving box
    9.
    外观设计
    Receiving box 失效
    接收箱

    公开(公告)号:USD468099S1

    公开(公告)日:2003-01-07

    申请号:US29141841

    申请日:2001-05-15

    申请人: Jerry Chang

    设计人: Jerry Chang

    System and method for dynamically changing error algorithm redundancy levels
    10.
    发明授权
    System and method for dynamically changing error algorithm redundancy levels 有权
    用于动态改变错误算法冗余级别的系统和方法

    公开(公告)号:US07286562B1

    公开(公告)日:2007-10-23

    申请号:US09993104

    申请日:2001-11-06

    IPC分类号: H04J3/22 G10L21/00

    摘要: The invention is concerned with improvements in full duplex Internet telephone systems with a system architecture having low latency and permitting voice communication with telephone to telephone or PC to telephone connections. The architecture permits dynamic packet-to-packet change in codec to adjust for Internet conditions. The voice port creates self-describing packet conditions so that the higher-level software of the system is independent of codec selection. In addition to adjusting the codec, the voice port has the capability of dynamically and concurrently selecting other factors such as the level of error correction redundancy, the packet size and packet bundling on a packet-to-packet basis. The invention further includes a technique to eliminate dead air spaces in the voice data transmission stream by speeding up or slowing down the data rate in the buffer while maintaining a constant pitch of speech.

    摘要翻译: 本发明涉及全双工互联网电话系统的改进,具有低延迟的系统架构,并允许与电话到电话或PC到电话连接的语音通信。 该架构允许编解码器中的动态数据包到数据包更改以适应Internet条件。 语音端口创建自描述分组条件,使得系统的较高级软件与编解码器选择无关。 除了调整编解码器之外,语音端口还具有动态并同时选择其他因素的能力,如纠错冗余级别,数据包大小和数据包到包的分组捆绑。 本发明还包括一种通过加速或减慢缓冲器中的数据速率来消除话音数据传输流中的死空间的技术,同时保持语音的恒定音调。