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    • 11. 发明授权
    • Methods and apparatus for data communication
    • 数据通信的方法和装置
    • US08194649B2
    • 2012-06-05
    • US12458883
    • 2009-07-27
    • Rafi RabipourChung Cheung C. ChuPierre GendronWilliam NavarroPaul Boudreaux
    • Rafi RabipourChung Cheung C. ChuPierre GendronWilliam NavarroPaul Boudreaux
    • H04L12/66
    • H04M7/0027H04L12/6418H04L69/08H04L69/24H04L2012/6486H04M3/08
    • Data communication apparatus including a port and a control entity. The control entity is operative to establish a connection with a remote entity over a first path and negotiate with the remote entity using in-band signaling over the first path establishment of a second path allowing the exchange of data between the data communication apparatus and the remote entity. The invention presents advantages from the standpoint of ease of implementation and bandwidth and resource savings. The use of an in-band messaging protocol to negotiate a establishment of the second path can be implemented generally in a straight forward manner. At the same time, the ability to transfer at least part of the connection to the second path avoids the drawbacks that would arise if that part of the connection were constrained to the first path. This feature allows the operator to take advantage of benefits provided by the second path but not available to the first path.
    • 包括端口和控制实体的数据通信装置。 控制实体可操作以通过第一路径与远程实体建立连接,并使用允许在数据通信设备和远端之间的数据交换的第二路径的第一路径建立的带内信令与远程实体进行协商 实体。 从实现的方便性和带宽和资源节省的观点出发,本发明具有优点。 通常使用带内消息协议来协商第二路径的建立通常以直接方式实现。 同时,将连接的至少一部分传送到第二路径的能力避免了如果连接的那部分被限制到第一路径时会出现的缺点。 该特征允许操作者利用由第二路径提供的优点,但不能用于第一路径。
    • 12. 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。
    • 13. 发明授权
    • Speech coding method using synthesis analysis using iterative
calculation of excitation weights
    • 使用迭代计算激励权重的语音编码方法
    • US5899968A
    • 1999-05-04
    • US860799
    • 1997-10-14
    • William NavarroMichel Mauc
    • William NavarroMichel Mauc
    • G10L19/10G10L9/14
    • G10L19/10
    • A linear prediction analysis is performed for each frame of a speech signal to determine the coefficients of a short-Term synthesis filter. For each sub-frame, an excitations sequence which, when applied to the short-term synthesis filter generates a synthetic signal representative of the speech signal, is determined by means of an iterative process in which a symmetrical matrix B.sub.n, is gradually built up with each iteration. The matrix B.sub.n is reversed with each iteration by decomposing the pattern B.sub.n =L.sub.n .multidot.R.sub.n.sup.T with L.sub.n =R.sub.n .multidot.K.sub.n where L.sub.n and R.sub.n are triangular matrices and K.sub.n is a diagonal matrix, and matrix L.sub.n has only 1s on its main diagonal.
    • PCT No.PCT / FR96 / 00005 Sec。 371日期1997年10月14日第 102(e)日期1997年10月14日PCT 1996年1月3日PCT PCT。 出版物WO96 / 21219 日期1996年7月11日对语音信号的每个帧执行线性预测分析,以确定短期合成滤波器的系数。 对于每个子帧,当施加到短期合成滤波器产生代表语音信号的合成信号时,通过迭代过程来确定兴奋序列,其中对称矩阵Bn逐渐与 每次迭代。 通过用Ln = RnxKn分解图形Bn = LnxRnT,其中Ln和Rn是三角矩阵,Kn是对角矩阵,并且矩阵Ln在其主对角线上只有1s,矩阵Bn被反转。