Repetitive pattern removal in a voice channel of a communication network
    2.
    发明授权
    Repetitive pattern removal in a voice channel of a communication network 失效
    在通信网络的语音信道中重复模式删除

    公开(公告)号:US6144658A

    公开(公告)日:2000-11-07

    申请号:US989585

    申请日:1997-12-12

    摘要: Repetitive packets in a voice/data stream, are detected and suppressed in the transmitting side of a network, after a predefined number of consecutive repetitive packets have been transmitted. Then, at the receiving side of the network, suppressed repetitive packets are reconstituted by filling the resulting gap in the voice/data stream with the repetitive pattern contained in the last received repetitive packet. When the input voice/data stream is compressed, only non-repetitive packets are compressed so that repetitive patterns are not corrupted by compression.

    摘要翻译: 在发送预定数量的连续重复分组之后,在网络的发送侧检测并抑制语音/数据流中的重复分组。 然后,在网络的接收侧,通过用包含在最后接收的重复分组中的重复模式填充语音/数据流中的所得到的间隙来重构抑制的重复分组。 当输入的语音/数据流被压缩时,只有非重复的数据包被压缩,使得重复的模式不被压缩破坏。

    Method and apparatus to reduce jitter and end-to-end delay for
multimedia data signalling
    3.
    发明授权
    Method and apparatus to reduce jitter and end-to-end delay for multimedia data signalling 失效
    用于减少多媒体数据信令的抖动和端到端延迟的方法和装置

    公开(公告)号:US5996018A

    公开(公告)日:1999-11-30

    申请号:US758071

    申请日:1996-11-27

    IPC分类号: G06F13/38 H04J3/06 G06F13/42

    CPC分类号: H04J3/0632 G06F13/387

    摘要: A method and an apparatus for reducing the jitter and end-to-end delay on lines of a packet switching network conveying voice or video digitalized data for one or more connections between a local source and a remote source at a constant bit rate.The method and apparatus of the invention are for use in a voice or video processor of a voice or video processing server of a network node; the method and the apparatus provide a way of controlling the remote traffic rate from the remote source before accessing the processor without using an external clocking such as the network clock.The solution proposed by the invention consists in buffering the remote and local traffics to adapt the remote traffic rate to the local traffic rate, which is supposed having a limited jitter, while sequencing of the access of the buffered data to the processor.

    摘要翻译: 一种用于减少分组交换网络的线路上的抖动和端到端延迟的方法和装置,其以恒定的比特率传送本地源和远程源之间的一个或多个连接的语音或视频数字化数据。 本发明的方法和装置用于网络节点的语音或视频处理服务器的语音或视频处理器; 该方法和装置提供了在访问处理器之前控制来自远程源的远程业务速率的方式,而不使用诸如网络时钟的外部时钟。 本发明提出的解决方案在于缓存远程和本地业务,以使远程业务速率适应本地业务速率,其被认为具有有限的抖动,同时缓冲数据到处理器的访问排序。

    Data processing method for efficiently transporting multimedia packets
over a conventional digital packet switching network
    4.
    发明授权
    Data processing method for efficiently transporting multimedia packets over a conventional digital packet switching network 失效
    用于通过传统数字分组交换网络高效传输多媒体分组的数据处理方法

    公开(公告)号:US5930265A

    公开(公告)日:1999-07-27

    申请号:US782694

    申请日:1997-01-16

    CPC分类号: H04Q11/0478 H04L2012/5652

    摘要: A data processing method for efficiently transporting multimedia data packets of fixed and/or variable length over an Assynchronous Transfer Mode (ATM) network made to transport fixed length ATM cells including a fixed length user data payload and a fixed length ATM header. The data processing method includes concatenating said fixed and/or variable length user data and appending said concatenated data with a so-called trailer defining the various concatenated user data lengths and identifications, for being further split into ATM cells payloads before being transmitted over said ATM network.

    摘要翻译: 一种数据处理方法,用于通过异步传输模式(ATM)网络高效地传输固定和/或可变长度的多媒体数据分组,用于传输固定长度的ATM信元,包括固定长度用户数据有效载荷和固定长度的ATM报头。 数据处理方法包括连接所述固定和/或可变长度的用户数据,并将所述连接的数据与定义各种级联用户数据长度和标识的所谓拖尾相连,以便在通过所述ATM发送之前进一步分割成ATM信元有效载荷 网络。

    Adaptive playout buffer and method for improved data communication
    5.
    发明授权
    Adaptive playout buffer and method for improved data communication 失效
    自适应播出缓冲器和改进数据通信的方法

    公开(公告)号:US06912224B1

    公开(公告)日:2005-06-28

    申请号:US09021707

    申请日:1998-02-10

    IPC分类号: H04L12/28 H04L12/56 H04L29/06

    CPC分类号: H04L49/90

    摘要: An adaptive apparatus and method for managing a playout buffer (POB) in an edge node of a packet-based data communications network, such as an ATM network, in order to reduce the end-to-end communication delay introduced thereby and to allow the POB filling level to be corrected in the event of clock speed differences. A monitoring mechanism determines if the minimum filling level of the playout buffer over a time period whereby the average filling level of the playout buffer is reduced according to the minimum filling level of the playout buffer over the time period.

    摘要翻译: 一种用于管理诸如ATM网络的基于分组的数据通信网络的边缘节点中的播出缓冲器(POB)的自适应装置和方法,以便减少由此引入的端到端通信延迟并允许 在时钟速度差异的情况下要纠正POB填充级别。 监视机构确定在一段时间内播放缓冲器的最小填充水平是否根据播放缓冲器在该时间段内的最小填充水平而减小播放缓冲器的平均填充水平。

    Method and a system for silence removal in a voice signal transported
through a communication network
    6.
    发明授权
    Method and a system for silence removal in a voice signal transported through a communication network 失效
    通过通信网络传输的语音信号中的静音消除方法和系统

    公开(公告)号:US5870397A

    公开(公告)日:1999-02-09

    申请号:US695280

    申请日:1996-08-06

    摘要: A method and an apparatus for removing the silence from the digitalized voice signals conveyed through packets or cells switching networks. The silence samples are neither packetized nor sent over the network but are regenerated at the output of the network. The silence samples generated are white noise samples, where the level is adapted to the background noise of the silence samples received at the input node of the network. For long periods of silence, the white noise level is periodically refreshed to be adapted to the last silence samples received at the input node of the network. The method provides also a control of packet or cell loss. The method uses are not control packets; in the later case, it can be used for ATM networks with AAL1. The method is implemented as a program executed in a Digital Signal Processor located on adapter cards dedicated to voice processing in the network access nodes.

    摘要翻译: 一种用于从通过分组或小区交换网络传送的数字化语音信号中去除静音的方法和装置。 沉默样本既不打包也不通过网络发送,而是在网络的输出端重新生成。 产生的静音样本是白噪声样本,其中该电平适应于在网络的输入节点接收的静音样本的背景噪声。 对于长时间的静音,白噪声电平被周期性刷新以适应于在网络的输入节点处接收到的最后沉默样本。 该方法还提供了分组或信元丢失的控制。 该方法使用的不是控制包; 在后一种情况下,它可以用于具有AAL1的ATM网络。 该方法被实现为位于专用于网络接入节点中的语音处理的适配卡上的数字信号处理器中执行的程序。

    System for coding voice signals to optimize bandwidth occupation in high
speed packet switching networks
    7.
    发明授权
    System for coding voice signals to optimize bandwidth occupation in high speed packet switching networks 失效
    用于编码语音信号的系统,以优化高速分组交换网络中的带宽占用

    公开(公告)号:US6104998A

    公开(公告)日:2000-08-15

    申请号:US213505

    申请日:1998-12-17

    摘要: A system for coding voice signal to optimize bandwidth occupation in a High Speed Packet Switching network while ensuring best voice transmission quality.The voice signal is first encoded using a conventional GSM like RPE/LTP coder providing first sub-frames of coded signal and tagging these first sub-frames as being non-discardable. In addition, a convenient difference between an RPE/LTP provided signal and a corresponding synthesized image is performed (see 36) and is also block encoded into second sub-frames which second sub-frames are tagged as being discardable sub-frames. Said second sub-frames when concatenated to corresponding first sub-frames provide so-called multirate frames. Then, when transmitting said multirate frames over the High Speed packet switching network, dropping discardable tagged data enables solution network congestion situations in any network node and at random with no significant disturbing effect over the voice communication operation.

    摘要翻译: 一种用于编码语音信号的系统,以优化高速分组交换网络中的带宽占用,同时确保最佳语音传输质量。 语音信号首先使用传统的GSM,如RPE / LTP编码器进行编码,提供编码信号的第一子帧,并将这些第一子帧标记为不可丢弃。 此外,执行RPE / LTP提供的信号和对应的合成图像之间的方便的差异(参见36),并且也被块编码成第二子帧,其中第二子帧被标记为可丢弃的子帧。 当连接到对应的第一子帧时,所述第二子帧提供所谓的多速率帧。 然后,当通过高速分组交换网络发送所述多速率帧时,丢弃可丢弃的标记数据使任何网络节点中的解决方案网络拥塞情况随机地在语音通信操作上无明显的干扰效应。

    File management method and system
    8.
    发明授权
    File management method and system 失效
    文件管理方法和系统

    公开(公告)号:US08412731B2

    公开(公告)日:2013-04-02

    申请号:US13164888

    申请日:2011-06-21

    IPC分类号: G06F17/30

    CPC分类号: G06F17/30106 G06F17/30115

    摘要: The invention provides a file management method and system for managing file retrieval and access. The method operates at the operating system level within a file system of a computer device and allows creating file move links upon detection of a file move request. The file move link associates the file source location with the file target location and is stored within a file move link table of the file system for subsequent file access request. File path to target location is automatically retrieved and file reached transparently for the user.

    摘要翻译: 本发明提供一种用于管理文件检索和访问的文件管理方法和系统。 该方法在计算机设备的文件系统内的操作系统级别操作,并且允许在检测到文件移动请求时创建文件移动链接。 文件移动链接将文件源位置与文件目标位置相关联,并存储在文件系统的文件移动链接表中以供后续文件访问请求。 自动检索到目标位置的文件路径,并为用户透明地显示文件。

    METHOD FOR DYNAMIC LEARNING OF INDIVIDUAL VOICE PATTERNS
    9.
    发明申请
    METHOD FOR DYNAMIC LEARNING OF INDIVIDUAL VOICE PATTERNS 失效
    动态语音模式的动态学习方法

    公开(公告)号:US20100153108A1

    公开(公告)日:2010-06-17

    申请号:US12368349

    申请日:2009-02-10

    IPC分类号: G10L15/06

    CPC分类号: G10L17/04 G10L15/065

    摘要: The present invention is a system and method for generating a personal voice font including, monitoring voice segments automatically from phone conversations of a user by a voice learning processor to generate a personalized voice font and delivering the personalized voice font (PVF) to the a server.

    摘要翻译: 本发明是一种用于生成个人语音字体的系统和方法,包括通过语音学习处理器自动从用户的电话对话中监控语音片段,以产生个性化语音字体,并将个性化语音字体(PVF)传送到服务器 。