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1.发明授权Method and system for transmitting a DTMF signal with compressed voice in a packet switching network 失效用于在分组交换网络中发送压缩语音的DTMF信号的方法和系统公开(公告)号：US06650662B1
公开(公告)日：2003-11-18
申请号：US09149913
申请日：1998-09-09
申请人： Charles Arnaud , Claude Cesaro , Gerard Richter
发明人： Charles Arnaud , Claude Cesaro , Gerard Richter
IPC分类号： H04J312
CPC分类号： H04M7/1295 , H04Q1/4575
摘要： The present invention discloses a system and a method for transmitting DTMF signals over high speed digital networks using voice compression algorithms, and particularly to a system for ensuring the integrity of DTMF signals at the destination node of a network after compression and decompression of data on a voice connection. A DTMF Detector is placed in parallel with a voice compression unit performing the compression algorithm. When a candidate DTMF signal is detected, the signal component at the higher frequency in the candidate is filtered out to eliminate the possibility of double DTMF detection at end user equipment. When the candidate DTMF signal is finally validated by the source node as representing a true DTMF signal, only the data essential for reconstituting the DTMF signal are transferred to the destination node.
摘要翻译： 本发明公开了一种使用语音压缩算法在高速数字网络上发送DTMF信号的系统和方法，特别涉及一种用于在压缩和解压缩数据之后确保网络目的地节点处的DTMF信号完整性的系统 语音连接。 DTMF检测器与执行压缩算法的语音压缩单元并行放置。 当检测到候选DTMF信号时，候选者中较高频率的信号分量被滤除，以消除终端用户设备双DTMF检测的可能性。 当候选DTMF信号最终被源节点验证为表示真正的DTMF信号时，仅将重构DTMF信号所必需的数据传送到目的地节点。
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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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公开(公告)号：US5533118A
公开(公告)日：1996-07-02
申请号：US202615
申请日：1994-02-28
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 (5.2) a dual tone is indicated. If it is greater than the second, the received signal level is compared to the third threshold (-43 dB) and receipt of a voice signal is indicated 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。 如果它大于第一个但小于第二个（5.2），则表示双重音调。 如果大于第二阈值，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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4.发明授权Tone-detection method for detecting at least one tone in a dual-tone multifrequency signal, call-progress method using the same 失效用于检测双音多频信号中的至少一个音调的音调检测方法，使用该音调检测方法的呼叫进行方法
公开(公告)号：US5495526A
公开(公告)日：1996-02-27
申请号：US202601
申请日：1994-02-28
申请人： Claude Cesaro , Gerard Richter
发明人： Claude Cesaro , Gerard Richter
CPC分类号： H04Q1/4575
摘要： A method for detecting the presence of 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 (5.2) a dual tone is indicated. When a single tone is indicated the received signal is subjected to a second order autoregressive process to determine the frequency of the single tone. If a dual tone is indicated the received signal is subjected to a fourth order autoregressive process to determine the frequencies of the dual tones.
摘要翻译： 用于检测单音和双音信号的存在的方法将采样周期内的接收信号的最大值（A max）的平方除以能量的测量E除以三个不同阈值的比率r进行比较。 如果r小于第一阈值（3），则指示单个音调。 如果它大于第一个但小于第二个（5.2），则表示双重音调。 当指示单个音调时，接收的信号经受第二阶自回归过程以确定单音的频率。 如果指示双音，则接收的信号经受第四阶自回归过程以确定双音调的频率。
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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：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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10.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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