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    • 1. 发明申请
    • Adapter for use with a tandem-free conference bridge
    • 适用于无串联会议桥的适配器
    • US20050114433A1
    • 2005-05-26
    • US10721909
    • 2003-11-26
    • Rafi RabipourPaul Coverdale
    • Rafi RabipourPaul Coverdale
    • G06F15/16H04L29/06
    • H04L65/4038H04L29/06027
    • A conference bridge adapter for processing data carried on a media path between, on the one hand, a conference bridge operative to communicate composite packets carrying media information and auxiliary information pertaining to the media information and, on the other hand, an endpoint characterized by an inability to exchange composite packets with the conference bridge. In one direction, a stream of composite packets is received from the bridge. An output media stream without auxiliary information is generated from the media information in each received composite packet. In the opposite direction, a stream of packets carrying only media information is received from the endpoint. Auxiliary information is derived from the media information and a stream of composite packets is generated by combining the media information and the auxiliary information. Thus, a media conference can be established between seemingly incompatible network elements.
    • 一种用于处理媒体路径上承载的数据的会议桥适配器,一方面是可操作以传送携带媒体信息的复合分组的会议桥和与媒体信息有关的辅助信息,另一方面, 无法与会议桥交换复合数据包。 在一个方向上,从桥接收到复合数据包流。 从每个接收到的复合分组中的媒体信息生成没有辅助信息的输出媒体流。 在相反的方向,从端点接收到仅携带媒体信息的分组流。 从媒体信息导出辅助信息,并通过组合媒体信息和辅助信息来生成复合数据包流。 因此,可以在看似不兼容的网络元素之间建立媒体会议。
    • 2. 发明授权
    • Method and apparatus for asymmetric communication of compressed speech
    • 压缩语音不对称通信的方法和装置
    • US06324515B1
    • 2001-11-27
    • US09088895
    • 1998-06-02
    • Rafi RabipourPaul CoverdaleWilliam Navarro
    • Rafi RabipourPaul CoverdaleWilliam Navarro
    • G10L2104
    • G10L19/00
    • This invention relates to a method and an apparatus for processing digital audio signals that may reduce the signal degradation occurring when the signal is exchanged between two communication terminals equipped with vocoders in a communication network. The solution proposed by this invention is to provide a communication terminal with a vocoder including a decoder section provided with a plurality of decoding units. A switch activates a selected one of the decoding units in dependence of the format of the compressed audio signal data frames received from a remote communication terminal. This system allows the communication terminal to support a number of different speech compression formals. In order to achieve simplicity and low cost, the communication terminal is provided with a single encoding unit. This results in an asymmetric arrangement where the communication terminal has a large number of decoding units than encoding units. The great majority of the speech compression algorithms deployed in wireless and Internet telephony standards have the property that their speech decoders are of far less computational complexity than their respective speech encoder. Therefore, a low-cost terminal can be produced which supports a low complexity speech encoder unit and a variety of speech decoder units.
    • 本发明涉及一种用于处理数字音频信号的方法和装置,其可以减少在通信网络中配备声码器的两个通信终端之间交换信号时发生的信号劣化。 本发明提出的解决方案是提供一种具有声码器的通信终端,该声码器包括设置有多个解码单元的解码器部分。 开关根据从远程通信终端接收的压缩音频信号数据帧的格式来激活所选择的解码单元。 该系统允许通信终端支持多种不同的语音压缩形式。 为了实现简单性和低成本,通信终端设置有单个编码单元。 这导致非对称布置,其中通信终端具有比编码单元大的解码单元数量。 部署在无线和因特网电话标准中的绝大多数语音压缩算法具有这样的性质:它们的语音解码器的计算复杂度远远低于它们各自的语音编码器。 因此,可以制造支持低复杂性语音编码器单元和各种语音解码器单元的低成本终端。
    • 5. 发明授权
    • Method and apparatus for improving the voice quality of tandemed vocoders
    • 提高串联声码器语音质量的方法和装置
    • US5995923A
    • 1999-11-30
    • US883353
    • 1997-06-26
    • Paul MermelsteinRafi RabipourWilliam NavarroPaul Coverdale
    • Paul MermelsteinRafi RabipourWilliam NavarroPaul Coverdale
    • G10L19/02G10L19/14H03M7/30H04W88/18H03M9/00
    • G10L19/16H04W88/181
    • In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.
    • 近年来,电讯业目睹了各种数字声码器的扩散,以满足不同有线和无线通讯系统的带宽需求。 网络的多样性的快速增长和这种网络的用户数量正在增加两个声码器串联放置以服务于单个连接的情况的数量。 低比特率编解码器的这种布置可能会降低发送语音的质量。 为了克服这个问题,本发明提供了一种用于在无线通信环境中发送数字化语音信号的新颖方法和装置。 该装置能够通过中间公共格式将压缩的语音信号从一种格式转换成另一格式,从而避免必须将语音数据连续解压缩到PCM型数字化,然后再重新压缩语音数据。
    • 7. 发明申请
    • MECHANISM FOR DYNAMIC SIGNALING OF ENCODER CAPABILITIES
    • 编码器功能动态信号的机制
    • US20130046534A1
    • 2013-02-21
    • US13588445
    • 2012-08-17
    • Rafi RabipourChung-Cheung ChuDaniel Cohn
    • Rafi RabipourChung-Cheung ChuDaniel Cohn
    • G10L19/12
    • H04M7/0072G10L19/00H04L65/1059H04L65/608H04W28/18H04W88/181
    • The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node.
    • 本公开提供用于动态地信令相应通信节点的声码器的编码器能力的系统和方法。 在一个实施例中,在第一通信节点和第二通信节点之间的呼叫期间,用于第一通信节点的控制节点(例如,基站控制器或移动交换中心)发送用于第一通信节点的声码器的语音编码器的能力信息 通信节点到第二通信节点的控制节点。 结果,第二通信节点能够基于第一通信节点的声码器的语音编码器的能力来选择并请求第一通信节点的声码器的语音编码器的优选编码器模式。
    • 9. 发明申请
    • METHOD AND APPARATUS FOR TIME ALIGNMENT ALONG A MULTI-NODE COMMUNICATION LINK
    • 多节点通信链路上时间对齐的方法与装置
    • US20090046698A1
    • 2009-02-19
    • US11839861
    • 2007-08-16
    • Chung Cheung CHURafi Rabipour
    • Chung Cheung CHURafi Rabipour
    • H04J3/06
    • H04L12/66H04N21/2335H04N21/23805H04N21/6181H04N21/64307H04N21/64322
    • A network entity, which comprises an input configured to receive from an upstream network entity a stream of first media data elements; an output configured to release towards a downstream network entity a stream of second media data elements; a processing engine configured to effect processing tasks on the first media data elements, thereby to generate the second media data elements, the processing tasks being effected in a set of processing intervals; and a control entity. The control entity is configured for receiving a request for a first phase adjustment from the downstream network entity; modifying the set of processing intervals in which are effected the processing tasks in an attempt to accommodate the first phase adjustment; determining a second phase adjustment based on arrival characteristics of the first media data elements and the modified set of processing intervals; and releasing towards the upstream network entity a request for the second phase adjustment.
    • 网络实体,其包括被配置为从上游网络实体接收第一媒体数据元素流的输入; 被配置为向下游网络实体释放第二媒体数据元素流的输出; 处理引擎被配置为对所述第一媒体数据元素执行处理任务,从而生成所述第二媒体数据元素,所述处理任务在一组处理间隔中进行; 和控制实体。 控制实体被配置为从下游网络实体接收对第一阶段调整的请求; 修改处理间隔的集合,以试图适应第一阶段调整; 基于所述第一媒体数据元素的到达特性和所述经修改的处理间隔集确定第二相位调整; 向上游网络实体发布第二阶段调整请求。