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1.发明授权
公开(公告)号：US6157637A
公开(公告)日：2000-12-05
申请号：US10004
申请日：1998-01-21
申请人： Claude Galand , Gerald Lebizay , Jean Menez , Gerard Richter
发明人： Claude Galand , Gerald Lebizay , Jean Menez , Gerard Richter
CPC分类号： H04Q11/0478 , H04L2012/5663 , H04L2012/5671 , H04Q2213/13034 , H04Q2213/13103 , H04Q2213/13164 , H04Q2213/13166 , H04Q2213/13196 , H04Q2213/13216 , H04Q2213/1322 , H04Q2213/1329
摘要： Process for transmitting compressed voice circuits over a packet switching network of the type comprising a plurality of switching nodes (SW-1 to SW-7) interconnected by connection lines and including at least an end switching node (SW-1) connected to a source exchange telephone device (PABX A) and another end switching node (SW-3) connected to a destination exchange telephone device (CX), and comprising the steps of receiving from the source exchange telephone device a sequence of uncompressed frames wherein each frame contains n slots containing each J bytes respectively associated to J voice circuits, compressing the data bits of each voice circuit in order to build a packet containing J compressed voice circuits, and providing this packet to the end switching node for transmission to the destination exchange telephone device.
摘要翻译： 用于通过包括通过连接线互连并包括至少连接到源的端交换节点（SW-1）的多个交换节点（SW-1至SW-7）的类型的分组交换网络来发送压缩语音电路的过程 交换电话设备（PABX A）和连接到目的地交换电话设备（CX）的另一终端交换节点（SW-3），并且包括以下步骤：从源交换电话设备接收一系列未压缩帧，其中每个帧包含n 包含分别与J个语音电路相关联的每个J字节的时隙，压缩每个语音电路的数据位，以构建包含J个压缩语音电路的数据包，并将该数据包提供给终端交换节点以传输到目的地交换电话设备。
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2.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
申请人： Claude Galand , Gerald Lebizay , Jean Menez , Bernard Pucci , Gerard Richter , Michele Rosso
发明人： Claude Galand , Gerald Lebizay , Jean Menez , Bernard Pucci , Gerard Richter , Michele Rosso
CPC分类号： G10H1/26 , G10H1/0066 , G10H2230/201 , G10H2240/251
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。
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3.发明授权Method and system for monitoring traffic to optimize the bandwidth reserved to an audio channel connection in a high speed digital network 失效用于监测流量以优化在高速数字网络中为音频信道连接预留的带宽的方法和系统
公开(公告)号：US5912894A
公开(公告)日：1999-06-15
申请号：US786915
申请日：1997-01-22
申请人： Maurice Duault , Claude Galand , Gerard Lebizay , Bernard Pucci , Gerard Richter
发明人： Maurice Duault , Claude Galand , Gerard Lebizay , Bernard Pucci , Gerard Richter
IPC分类号： H04L12/54 , H04L12/801 , H04L12/853 , H04L12/911 , H04L12/913 , H04L12/927 , H04M3/22 , H04M7/12 , H04J3/17
CPC分类号： H04L12/5695 , H04L47/15 , H04L47/2416 , H04L47/724 , H04L47/801 , H04L47/829 , H04M3/2254 , H04M7/12
摘要： This invention deals with a method and system for dynamically adjusting the communication bandwidth assigned to an audio channel connection in a high speed digital network. More particularly, the invention is made to track the activity of a voice assigned connection (e.g. a PBX or PABX entry to the network), define a so-called activity bit for each block of audio channel signal and then dynamically adjust the assigned network communication bandwidth accordingly. Adjustment of the communication bandwidth is accomplished by integrating the audio channel activity bits through a predefined integration function, the result of which is then compared against predefined threshold values in order to determine an appropriate bandwidth setting for the audio channel.
摘要翻译： 本发明涉及用于动态调整分配给高速数字网络中的音频信道连接的通信带宽的方法和系统。 更具体地说，本发明用于追踪语音分配的连接（例如，到网络的PBX或PABX条目）的活动，为每个音频信道信号块定义所谓的活动比特，然后动态地调整分配的网络通信 带宽相应。 通过将音频信道活动比特整合到预定义的积分函数来实现通信带宽的调整，然后将其结果与预定义的阈值进行比较，以便为音频信道确定适当的带宽设置。
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公开(公告)号：US5619565A
公开(公告)日：1997-04-08
申请号：US598294
申请日：1996-02-08
申请人： Claude Cesaro , Gerard Richter
发明人： Claude Cesaro , Gerard Richter
摘要： A method for detecting the presence of voice, single tone and dual tone signals compares a ratio r which equals the square of the maximum value of the received signal (A max) during a sampling period divided by a measure E of the energy to three different thresholds. If r is less than the first threshold (3), a single tone is indicated. If it is greater than the first but less than the second, the received signal level is compared to the third threshold (-43 dB) and receipt of a voice signal is indicted if the level of the received signal exceeds this threshold.
摘要翻译： 用于检测语音，单音和双音信号的存在的方法将在采样周期期间等于接收信号的最大值（A max）的平方的比率r除以能量的测量E除以三个不同的 阈值。 如果r小于第一阈值（3），则指示单个音调。 如果它大于第一但小于秒，则将接收信号电平与第三阈值（-43dB）进行比较，并且如果接收信号的电平超过该阈值则指示语音信号的接收。
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5.发明授权Decimation filter for a sigma-delta converter and A/D converter using the same 失效用于Σ-Δ转换器和使用其的A / D转换器的抽取滤波器
公开(公告)号：US5461641A
公开(公告)日：1995-10-24
申请号：US981157
申请日：1992-11-23
CPC分类号： H03H17/0664
摘要： A Decimation filter for converting a train of sigma-delta pulses S(i) in synchronism with a sigma-delta clock (fs) into a train of Pulse Coded Modulation (PCM) samples in accordance with the formula ##EQU1## where Cn is the sequence of the coefficients of the decimation filter which corresponds to a determined decimation factor, and the PCM samples being processed by a Digital Signal Processor (DSP). The decimation filter includes a device for storing a digital value representative of the DC component introduced during the sigma-delta coding process, with the digital value being computing by the DSP processor during an initialization phase. The decimation filter further includes a device operating after the latter initialization phase for subtracting the stored digital value from each of the PCM samples so that the resulting sequence of PCM samples appears free of any DC component introduced during the sigma-delta coding. This accurate DC component suppression is achieved without necessitating the use of additional digital signal processor resources from the processor. Preferably, the decimation filter comprises a device for detecting a saturation occurring in the computing of the PCM sample, and responsive to the saturation detection, for transmitting a predetermined PCM sample to the DSP processor.
摘要翻译： 一种抽取滤波器，用于根据与之对应的抽取滤波器的公式来将与Σ-Δ时钟（fs）同步的Σ-Δ脉冲序列转换成脉冲编码调制（PCM）采样序列 到确定的抽取因子，并且PCM采样由数字信号处理器（DSP）处理。 抽取滤波器包括用于存储代表在Σ-Δ编码处理期间引入的DC分量的数字值的装置，数字值由DSP处理器在初始化阶段期间计算。 抽取滤波器还包括在后一初始化阶段之后操作的装置，用于从每个PCM样本中减去所存储的数字值，使得所得到的PCM样本序列在Σ-Δ编码期间不会出现任何DC分量。 实现这种精确的DC分量抑制，而不需要使用来自处理器的附加数字信号处理器资源。 优选地，抽取滤波器包括用于检测在PCM采样的计算中出现的饱和度并且响应饱和检测用于将预定的PCM采样发送到DSP处理器的装置。
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公开(公告)号：US06912224B1
公开(公告)日：2005-06-28
申请号：US09021707
申请日：1998-02-10
申请人： Maurice Duault , Bernard Pucci , Serge Leroy , Gerard Richter , Francois Kermarec , Aline Fichou
发明人： Maurice Duault , Bernard Pucci , Serge Leroy , Gerard Richter , Francois Kermarec , Aline Fichou
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。
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公开(公告)号：US5822609A
公开(公告)日：1998-10-13
申请号：US666767
申请日：1996-06-19
申请人： Gerard Richter
发明人： Gerard Richter
CPC分类号： G06F17/15 , G06F17/10 , G06F9/3885
摘要： A processing unit for performing convolution computation according to the HARVARD architecture which includes a first and second input register for receiving a first and second operand, a multiplier for multiplying the operand and a Arithmetic and Logic Unit (ALU) circuit. The unit further includes a coefficient storage memory which is used for loading at least one set of coefficients allowing the convolution computation. The memory storage is addressed either from an internal address generator or directly from the internal data bus thereby allowing the possibility to store either coefficients or data into the memory. The flexibility is still increased by the use of a particular set of multiplexing circuits allowing multiple configurations. An internal address generation circuit is used for performing a partial addressing of the set of coefficients thereby providing decimation capability.
摘要翻译： 一种处理单元，用于根据HARVARD架构执行卷积计算，该结构包括用于接收第一和第二操作数的第一和第二输入寄存器，用于乘法运算的乘法器和算术逻辑单元（ALU）电路。 该单元还包括系数存储存储器，其用于加载允许卷积计算的至少一组系数。 存储器存储器可以从内部地址生成器或直接从内部数据总线寻址，从而允许将系数或数据存储到存储器中的可能性。 通过使用允许多种配置的特定多路复用电路集，灵活性仍然增加。 内部地址生成电路用于执行该组系数的部分寻址，从而提供抽取能力。
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8.发明授权
公开(公告)号：US5220327A
公开(公告)日：1993-06-15
申请号：US878106
申请日：1992-05-04
CPC分类号： H03H17/0614 , H03H17/0664
摘要： A decimation filter for converting a train of sigma-delta pulses S(i) in synchronism with a sigma-delta clock (fs) into a train of PCM samples which includes counters (321, 331, 341) driven by the sigma-delta clock (fs) and which is continuously incremented by one during N sigma-delta clock pulses, then decremented by two during N following sigma-delta clock pulses and then incremented again by one during N following sigma-delta clock pulses in order to provide a sequence of incrementation parameter DELTA(n). The decimation filter further includes storages (320, 330, 340) for storing the value of the coefficient C(n) corresponding to the decimation filter transfer function, and incrementers (327, 337, 347) driven by the sigma-delta clock fs for incrementing the storages with the incrementation parameter DELTA(n). Finally, the decimation filter includes computers (323, 333, 343, 327, 337, 347) for deriving from the contents C(n) of said storages and from the train of input sigma-delta samples S(i+n) one Pulse Code Modulation (PCM) sample every 3.times.N input sigma-delta samples according to the formula: ##EQU1##
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9.发明授权
公开(公告)号：US6144658A
公开(公告)日：2000-11-07
申请号：US989585
申请日：1997-12-12
申请人： Gerard Lebizay , Maurice Duault , Bernard Pucci , Gerard Richter
发明人： Gerard Lebizay , Maurice Duault , Bernard Pucci , Gerard Richter
CPC分类号： H04L12/6418 , H04J3/17 , H04L2012/6467 , H04L2012/6481
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。
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公开(公告)号：US5418817A
公开(公告)日：1995-05-23
申请号：US26319
申请日：1993-03-03
申请人： Gerard Richter
发明人： Gerard Richter
CPC分类号： H04L25/0305
摘要： Adaptive equalization system for allowing the equalization of a base-band line of a DCE, includes an adaptive filter for adjusting its coefficients in accordance with an adaptive algorithm. The equalizer includes controls to work during a preliminary period, in an adaptive or non-adaptive mode according to the convergence of the recursive algorithm process. The algorithm is triggered depending on whether the equalized signal belongs to one of several predefined intervals of convergence. After this preliminary period, when the mean square of the residual error is considered small enough to ensure the convergence of the adaptive process, the algorithm is computed on a continuous mode.
摘要翻译： 用于允许DCE的基带线路的均衡的自适应均衡系统包括用于根据自适应算法来调整其系数的自适应滤波器。 均衡器包括根据递归算法进程的收敛在自适应或非自适应模式下在初步期间工作的控制。 根据均衡信号是否属于几个预定义的收敛间隔之一来触发该算法。 在这个初步阶段之后，当剩余误差的均方根被认为足够小以确保自适应过程的收敛时，算法在连续模式下计算。
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