Iterative channel estimation
    1.
    发明授权
    Iterative channel estimation 有权
    迭代信道估计

    公开(公告)号:US06459728B1

    公开(公告)日:2002-10-01

    申请号:US09302764

    申请日:1999-04-30

    IPC分类号: H04L2701

    摘要: A method and apparatus for estimating channel impulse response and data in a signal transmitted over a channel in a communication system. The channel impulse response is estimated uses correlative channel sounding, and then, using the estimated channel impulse response, the data in the signal is estimated. The output is then fed back to the channel impulse response estimator and the channel impulse estimation is repeated. The data estimation and channel impulse response estimation may be iterated a number of times.

    摘要翻译: 一种用于估计在通信系统中通过信道发送的信号中的信道脉冲响应和数据的方法和装置。 估计信道脉冲响应使用相关信道探测,然后使用估计的信道脉冲响应,估计信号中的数据。 然后将输出反馈到信道脉冲响应估计器,并重复信道脉冲估计。 数据估计和信道脉冲响应估计可以重复多次。

    Method for converting a signal data rate, and a transmitter
    2.
    发明授权
    Method for converting a signal data rate, and a transmitter 失效
    用于转换信号数据速率的方法和发射机

    公开(公告)号:US06332005B1

    公开(公告)日:2001-12-18

    申请号:US09177246

    申请日:1998-10-22

    申请人: Kari Pehkonen

    发明人: Kari Pehkonen

    IPC分类号: H04L2701

    摘要: The invention relates to a method for converting a signal data rate and to a transmitter in a digital data transmission system, the transmitter comprising a number of data sources that generate a signal having a first data rate, and a coder for forming the signal into frames of a given length, which length corresponds to a second data rate, and a coder for grouping the signal into a first block the length of which corresponds to the first data rate. In order to convert the data rate flexibly, the transmitter comprises a coder, which calculates the difference between the frame and the number of symbols in the first block by subtracting the frame length from the number of symbols in the first block, which equalizes the difference by removing or repeating every ith symbol in the first block, i being determined as the nearest bigger or an equal integer when the number of symbols in the first block is divided by the absolute value of the difference that was calculated, which updates the number of the symbols that do not fit in the frame by subtracting therefrom the number of the symbols that have been repeated, and which forms the first block to consist of the symbols that have not been repeated or removed.

    摘要翻译: 本发明涉及一种用于在数字数据传输系统中转换信号数据速率和发射机的方法,所述发射机包括产生具有第一数据速率的信号的多个数据源,以及用于将信号形成帧的编码器 给定长度,其长度对应于第二数据速率;以及编码器,用于将信号分组成第一块,其长度对应于第一数据速率。 为了灵活地转换数据速率,发射机包括编码器,该编码器通过从第一块中的符号数减去帧长度来计算第一块中的帧和符号数之间的差异, 通过去除或重复第一块中的每个第i个符号,当第一个块中的符号数除以计算出的差的绝对值时,i被确定为最接近的较大或相等的整数, 通过从其中减去已经重复的符号的数量并且形成第一块以由未被重复或去除的符号组成的符号,不适合于帧中的符号。

    Method and apparatus for improved channel equalization and level learning in a data communication system

    公开(公告)号:US06459729B1

    公开(公告)日:2002-10-01

    申请号:US09329465

    申请日:1999-06-10

    申请人: Yhean-Sen Lai

    发明人: Yhean-Sen Lai

    IPC分类号: H04L2701

    摘要: A method and apparatus are disclosed for improving channel equalization and level learning in a data communication system. The disclosed equalizer training process separately updates the feed forward filter (FFF) and the level adapter, to gain additional improvements in the training of the feed forward filter (FFF). The multi-step equalizer training process initially trains the feed forward filter (FFF) using a two-level signal EQTR(n) having an ideal value (Step One) to help converge the feed forward filter (FFF) to a certain level. Once the feed forward filter (FFF) reaches a certain level of convergence, the training circuitry is reconfigured during step two of the equalizer training process, to evaluate and update the actual level of the signal EQTR(n), to compensate for the channel. The determined weighting factors are applied to a low pass filter and the actual level of the signal EQTR(n), B(n), is calculated. Once the actual level of the signal EQTR(n), B(n), has been calculated, the actual level of the signal EQTR(n), B(n), is applied to the level adapter, and the level adapter is no longer updated by reconfiguring the training circuitry to remove the error signal, err(n), inputs to the level adapter. During step three, the feed forward filter (FFF) continues to be updated and fine-tuned by the error signal err(n). Since the level of the signal EQTR(n) is the actual value, B(n), the performance of the feed forward filter (FFF) is improved. Once the equalizer training process is complete, the feed forward filter (FFF) is fixed. The improved training of the feed forward filter (FFF) allows the structure of level learning process to be simplified, with the training circuitry removed and the feed forward filter (FFF) fixed, where each level will be divided into six phases and processed individually.

    Automatic equalizer capable of surely selecting a suitable sample timing a method for generating sampling clock used for the sample timing and a recording medium usable in control of the automatic equalizer
    4.
    发明授权
    Automatic equalizer capable of surely selecting a suitable sample timing a method for generating sampling clock used for the sample timing and a recording medium usable in control of the automatic equalizer 有权
    自动均衡器,能够可靠地选择合适的采样定时,用于产生用于采样定时的采样时钟的方法和可用于控制自动均衡器的记录介质

    公开(公告)号:US06314133B1

    公开(公告)日:2001-11-06

    申请号:US09243565

    申请日:1999-02-03

    IPC分类号: H04L2701

    摘要: An automatic equalizer has a sampling-clock producing arrangement which is for, before selecting a sample timing, producing a sampling clock at the rate of L times of that after selecting, and after selecting the sample timing, producing a tap-coefficient selection signal according to the sample timing, and a sampling clock at the rate of 1/L times of that before selecting, according to the sample timing. In the sampling-clock producing arrangement, demodulation components are obtained in absolute values of impulse-response signals with respect to L sample timings, respectively. A selecting arrangement selects the sample timing by the use of the demodulation components. The impulse-response signals are produced in response to a sampled received-signal obtained by sampling a received signal with the sampling clock.

    摘要翻译: 自动均衡器具有采样时钟产生装置,用于在选择采样定时之前,以选择后的L倍的速率产生采样时钟,并且在选择采样定时之后,产生抽头系数选择信号,根据 根据采样定时,采样时钟以选择之前的1 / L倍的速率进行采样。 在采样时钟产生装置中,分别获得关于L个采样定时的脉冲响应信号的绝对值的解调分量。 选择装置通过使用解调分量来选择采样定时。 脉冲响应信号响应于通过采样时钟对接收信号进行采样而获得的采样接收信号产生。

    System and method for performing combined multi-rate convolutional coding
    5.
    发明授权
    System and method for performing combined multi-rate convolutional coding 有权
    用于执行组合多速率卷积编码的系统和方法

    公开(公告)号:US06487251B1

    公开(公告)日:2002-11-26

    申请号:US09386053

    申请日:1999-08-30

    IPC分类号: H04L2701

    摘要: A system and method for establishing an integrated forward error correction (FEC) scheme to perform multi-rate encoding on different priority data bits of a channel access message transmitted on a random access channel between devices of a communications network, such as between an access terminal and a base station of a satellite-based communications network. The channel access message includes a first data group representing first information and a second data group representing second information, which is transmitted between an access terminal and a base station in a satellite-based communications network. The system and method encodes the second data group at an encoding rate to provide a second encoded data group, and encodes the first data group at the same encoding rate to provide a first encoded data group. The encoding of the first and second data groups is performed by a single encoder, such as a rate ¼ convolutional encoder. The second encoded data group is transmitted from the access terminal to the base over a random access channel. The second encoded data group further can be punctured during transmission to in effect decrease its coding rate, for example, to rate ½ coding. The first encoded data group is transmitted from the access terminal to the base station, and is then retransmitted from the access terminal to the base station to in effect increase the rate of coding of the first encoded data group to, for example, ⅛ coding. At the base station, a combiner/demodulator combines the transmitted and retransmitted first encoded data group, and the combined first encoded data groups and the second encoded data group are then decoded by a decoder.

    摘要翻译: 一种用于建立综合前向纠错(FEC)方案的系统和方法,用于对通信网络的设备之间在诸如接入终端之间的随机接入信道上发送的信道接入消息的不同优先级数据比特执行多速率编码 以及基于卫星的通信网络的基站。 信道接入消息包括表示第一信息的第一数据组和表示在基于卫星的通信网络中的接入终端和基站之间传输的第二信息的第二数据组。 该系统和方法以编码速率对第二数据组进行编码以提供第二编码数据组,并以相同的编码速率对第一数据组进行编码以提供第一编码数据组。 第一和第二数据组的编码由单个编码器执行,例如速率¼卷积编码器。 第二编码数据组通过随机接入信道从接入终端发送到基站。 第二编码数据组在传输期间还可以被打孔,从而实际上降低了其编码率,例如,对1/2编码进行速率估计。 第一编码数据组从接入终端发送到基站,然后从接入终端重传到基站,实际上将第一编码数据组的编码速率提高到例如1/8 编码。 在基站,组合器/解调器组合发送和重传的第一编码数据组,然后由解码器解码组合的第一编码数据组和第二编码数据组。

    Stepwise adaptive finite impulse response filter for spread spectrum radio
    6.
    发明授权
    Stepwise adaptive finite impulse response filter for spread spectrum radio 有权
    用于扩频无线电的逐步自适应有限脉冲响应滤波器

    公开(公告)号:US06438156B1

    公开(公告)日:2002-08-20

    申请号:US09248664

    申请日:1999-02-11

    IPC分类号: H04L2701

    摘要: A distortion compensation filtering mechanism for a direct spread-spectrum radio receiver comprises an iteratively adaptive FIR filter installed in the received signal processing path of the radio just upstream of the despreading function. The filter may be implemented as a relatively small numbered tap filter, having its precursor tap fixed at a maximum value. The remaining filter tap values are individually adaptively adjusted by the radio's control processor, which executes a tap adjustment routine to iteratively increment or decrement each variable tap value of the FIR filter to an ‘optimized’ value, that effectively minimizes the total (I and Q) power in the error in the data decisions performed by data signal analyzer.

    摘要翻译: 用于直接扩频无线电接收机的失真补偿滤波机构包括安装在解扩功能上游的无线电接收信号处理路径中的迭代自适应FIR滤波器。 滤波器可以被实现为相对小的编号的抽头滤波器,其前体抽头固定在最大值。 剩余的滤波器抽头值通过无线电的控制处理器单独地自适应地调整,无线电的控制处理器执行抽头调整例程,以将FIR滤波器的每个可变抽头值迭加地递减或递减到“优化”值,其有效地最小化总计(I和Q )功率由数据信号分析仪执行的数据决策中的误差。

    Finite impulse response filter for wave-shaping digital quadrature amplitude modulation symbols
    7.
    发明授权
    Finite impulse response filter for wave-shaping digital quadrature amplitude modulation symbols 失效
    用于波形整形数字正交幅度调制符号的有限脉冲响应滤波器

    公开(公告)号:US06188723B1

    公开(公告)日:2001-02-13

    申请号:US09112059

    申请日:1998-07-09

    IPC分类号: H04L2701

    CPC分类号: H04L27/34 H04L25/03834

    摘要: A finite impulse response (FIR) filter for wave-shaping digital quadrature amplitude modulation (QAM) symbols is disclosed, in which multipliers are replaced with multiplexers, the replaced multiplexers are utilized to receive the symbols directly from a symbol encoder without zero (0) interpolations, and the critical path is reduced by shifting the position of a delay device. The filter includes a first FIR means for delaying the externally inputted symbol data, and for utilizing the delayed symbol data as selection signals to sum up the selected multiplication product (selected from among products obtained by multiplying the symbol values by a pre-set filter tab coefficient) and the selected value selected by a first multiplexing means. A second FIR means delays again the delayed symbol data of the first FIR means, and utilizes the delayed symbol data as selection signals to sum up the selected multiplication product and the output value of the first FIR means.

    摘要翻译: 公开了一种用于波形整形数字正交幅度调制(QAM)符号的有限脉冲响应(FIR)滤波器,其中乘法器被多路复用器替代,替换的多路复用器被用于直接从符号编码器接收符号,而不用零(0) 内插,并且通过移动延迟装置的位置来减小关键路径。 滤波器包括用于延迟外部输入的符号数据的第一FIR装置,并且用于利用延迟的符号数据作为选择信号来对所选乘法乘积进行求和(从通过将符号值乘以预设的滤波器选项卡获得的乘积中选择 系数)和由第一多路复用装置选择的选择值。 第二FIR装置再次延迟第一FIR装置的延迟符号数据,并且利用延迟的符号数据作为选择信号来对所选乘法乘积和第一FIR装置的输出值求和。

    Method and apparatus for fault recovery in a decision feedback equalizer
    8.
    发明授权
    Method and apparatus for fault recovery in a decision feedback equalizer 失效
    决策反馈均衡器中故障恢复的方法和装置

    公开(公告)号:US06608864B1

    公开(公告)日:2003-08-19

    申请号:US09320258

    申请日:1999-05-26

    申请人: Jeffrey C. Strait

    发明人: Jeffrey C. Strait

    IPC分类号: H04L2701

    摘要: A method and apparatus for frequency domain equalization. The method and apparatus are particularly well suited for use in a receiver in a multicarrier transmission system that has predetermined symbols periodically embedded in the transmissions, such as in an ADSL system. The equalizer apparatus includes a digital filter having a plurality of single-tap filters. The filters operate on the symbols in the frequency domain, accepting frequency domain representations of the received symbols. The equalizer also includes a reference symbol generator that provides reference symbols, a coefficient generator that accepts the equalized frequency domain symbols from the digital filter and the reference symbols and updates the filter taps using a received predetermined symbol and the reference symbol. The coefficient generator includes a generator filter having a first and a second adaptation increment, an error generator, and a threshold detector that controls the coefficient generator in response to said error signal. Preferably, the threshold detector determines whether the equalizer output would have been decoded improperly during a synchronization symbol, and if so, enables the use of the second update increment in the generator filter.

    摘要翻译: 一种用于频域均衡的方法和装置。 该方法和装置特别适用于具有周期性地嵌入在传输中的预定符号的多载波传输系统中的接收机中,例如在ADSL系统中。 均衡器装置包括具有多个单抽头滤波器的数字滤波器。 滤波器对频域中的符号进行操作,接收接收符号的频域表示。 均衡器还包括提供参考符号的参考符号发生器,接收来自数字滤波器和参考符号的均衡频域符号的系数发生器,并且使用接收到的预定符号和参考符号来更新滤波器抽头。 系数发生器包括具有第一和第二自适应增量的发生器滤波器,误差发生器和响应于所述误差信号控制系数发生器的阈值检测器。 优选地,阈值检测器确定均衡器输出是否在同步符号期间被不正确地解码,并且如果是,则使能在生成器滤波器中使用第二更新增量。

    Apparatus for, and method of, processing signals transmitted over a local area network
    9.
    发明授权
    Apparatus for, and method of, processing signals transmitted over a local area network 有权
    用于处理通过局域网传输的信号的装置和方法

    公开(公告)号:US06459730B1

    公开(公告)日:2002-10-01

    申请号:US09482699

    申请日:2000-01-13

    IPC分类号: H04L2701

    摘要: Digital signals provided by a repeater connected to a plurality of clients by unshielded twisted wire pairs, are converted to analog signals which become degraded during transmission through the wires. Clients convert the degraded analog signals to digital signals. Digital signal phases are coarsely adjusted to have assumed zero crossing times coincide in-time with a clock signal zero crossing. Signal polarity, and the polarity of any change, is determined at the assumed zero crossing times of the digital signals. Pre-cursor and post-cursor responses, resulting from signal degradation, are respectively inhibited by a feed forward and a decision feedback equalizer. The time duration of post-cursor response is further inhibited by a high pass filter and a tail canceller. Phase adjustments are made, after response inhibition, by determining the polarity, and the polarity of any change, at the assumed zero crossing times. Before phase adjustments are made, a phase offset is provided in order to compensate for phase degradations introduced by the unshielded twisted wire pairs.

    摘要翻译: 由通过非屏蔽双绞线连接到多个客户端的中继器提供的数字信号被转换为在通过电线传输期间降级的模拟信号。 客户端将降级的模拟信号转换为数字信号。 数字信号相位粗略地调整为假定零交叉时间与时钟信号过零时间一致。 信号极性和任何变化的极性在假定的数字信号的零交叉时间确定。 分别由前馈和判决反馈均衡器阻止由信号劣化引起的前标和后视标响应。 后置光标响应的持续时间被高通滤波器和尾部消除器进一步抑制。 通过在假设的零交叉时间确定极性和任何变化的极性,在响应抑制之后进行相位调整。 在进行相位调整之前,提供相位偏移以补偿由未屏蔽的双绞线引入的相位劣化。

    Blind start-up of a dual mode CAP-QAM receiver
    10.
    发明授权
    Blind start-up of a dual mode CAP-QAM receiver 失效
    双模式CAP-QAM接收机的盲启动

    公开(公告)号:US06252903B1

    公开(公告)日:2001-06-26

    申请号:US09109364

    申请日:1998-07-02

    IPC分类号: H04L2701

    摘要: A receiver has a dual mode of operation—a carrierless amplitude modulation/phase modulation (CAP) mode and a quadrature amplitude modulation (QAM) mode-yet only requires a single equalizer structure for both the CAP mode of operation and the QAM mode of operation during blind start-up. The receiver uses the same blind equalization updating algorithm independent of the type of received signal for converging the equalizer structure. The blind equalization updating algorithm incorporates a constant R, whose value is a function of the type of received signal, e.g., a QAM signal or a CAP signal. The type of received signal is determined as a function of the in-phase component of the mean-squared error, E[e2n.

    摘要翻译: 接收机具有双重操作模式 - 无载波幅度调制/相位调制(CAP)模式和正交幅度调制(QAM)模式 - 但是仅需要CAP操作模式和QAM操作模式的单个均衡器结构 在盲人启动期间。 接收机使用与接收信号类型无关的盲均衡更新算法,用于收敛均衡器结构。 盲均衡更新算法包含常数R,其值是接收信号类型的函数,例如QAM信号或CAP信号。 接收信号的类型被确定为均方误差E [e2n]的同相分量的函数。