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1. 发明申请

WO2016026263A1 确定自适应滤波器的稳定因子的方法和装置审中-公开
公开(公告)号：WO2016026263A1
公开(公告)日：2016-02-25
申请号：PCT/CN2015/070219
申请日：2015-01-06
申请人：华为技术有限公司
发明人：苏谟特⋅艾雅 , 刘媛媛 , 李海婷
IPC分类号： H03H21/00
CPC分类号： H03H21/0043 , G06F17/16 , H03H2021/0054 , H03H2021/0061
摘要：本发明提供一种确定自适应滤波器的稳定因子的方法和装置，该方法包括：根据输入到自适应滤波器的第一输入信号，确定第一输入信号的参考输入矩阵；根据参考输入矩阵，确定第一输入信号的稳定性参数；根据稳定性参数，确定自适应滤波器的稳定因子。本发明实施例提供的确定自适应滤波器的稳定因子的方法和装置，通过根据自适应滤波器的第一输入信号的参考输入矩阵确定的该第一输入信号的稳定性参数，确定该自适应滤波器的稳定因子，能够实现根据第一输入信号的稳定性特点自适应的获取自适应滤波器的稳定因子，从而能够使得自适应滤波器在收敛速度与稳态误差性能上达到平衡。

2. 发明申请

WO2009045414A1 SYSTEM AND METHOD FOR ADAPTIVE NONLINEAR FILTERING 审中-公开
标题翻译：用于自适应非线性滤波的系统和方法
公开(公告)号：WO2009045414A1
公开(公告)日：2009-04-09
申请号：PCT/US2008/011334
申请日：2008-09-30
申请人： OPTICHRON, INC.
发明人： BATRUNI, Roy, G.
IPC分类号： H03B1/00
CPC分类号： H03H21/0043 , H03H21/0016 , H03H2021/0079
摘要： An adaptive nonlinear filtering system includes an adaptive filter module that is configured to generate relative location information pertaining to a relative location of an input signal within an input range; determine an input dependent filter parameter based at least in part on the relative location information; generate an output signal based at least in part on the input dependent filter parameter; and feed back a feedback signal that is generated based at least in part on the output signal and a target signal.
摘要翻译：自适应非线性滤波系统包括自适应滤波器模块，其被配置为产生与输入范围内的输入信号的相对位置有关的相对位置信息; 至少部分地基于所述相对位置信息来确定输入相关滤波器参数; 至少部分地基于输入依赖滤波器参数来生成输出信号; 并且反馈至少部分地基于输出信号和目标信号产生的反馈信号。

3. 发明申请

WO2005069491A3 METHOD OF FILTERING BINARY DATA 审中-公开
标题翻译：滤除二值数据的方法
公开(公告)号：WO2005069491A3
公开(公告)日：2005-08-25
申请号：PCT/IT2005000022
申请日：2005-01-18
申请人： ATOP INNOVATION SPA , CARABINIERE LUIGI
发明人： CARABINIERE LUIGI
IPC分类号： G06T9/00 , G10L19/04 , H03H21/00 , H03M7/30 , H04N1/417
CPC分类号： H03H21/0043
摘要： The present invention concerns a method of filtering an input binary string W for obtaining an output binary string Y, the method being characterised in that it comprises, for each one of one or more first substrings wj having nj bits at least partially belonging to the input string W, the following steps: A. calculating K prediction substrings ujx, with K>=2, through K operations of multiplication of the first considered substring wj by K functions fk; B. having a second substring wj+1, comprising nj+1 bits at least partially belonging to the input string W; C. calculating, for each one of the K prediction substrings ujk, a corresponding distance Dj+l,k indicative of the deviation of the second substring wj+1 from the prediction substring ujk; D. selecting the function fkDHmin of the K functions fk having the mìnimum distance Dj+1,k from the substring wj+1; and E. inserting a substring yj equal to the prediction substring ujkDHmin generated by the function fkDHmin selected in step D ìnto the output string Y. The present invention further concerns the corresponding defiltering method, and the apparatuses and instruments necessary for performing the method.
摘要翻译：本发明涉及一种对输入二进制串W进行滤波以获得输出二进制串Y的方法，该方法的特征在于，对于一个或多个第一子串wj中的每一个具有至少部分属于输入的nj个比特字符串W执行以下步骤：A.通过将第一个考虑的子串wj乘以K个函数fk的K次运算，计算K个预测子串ujx，其中K> = 2; B.具有第二子串wj + 1，包括至少部分属于输入串W的nj + 1个比特; C.针对K个预测子串ujk中的每一个，计算指示第二子串wj + 1与预测子串ujk的偏差的对应距离Dj + 1，k; D.从子串wj + 1中选择具有最小距离Dj + 1，k的K个函数fk的函数fkDHmin; 以及E.将等于由在步骤D中选择的函数fkDHmin生成的预测子串ujkDHmin的子串yj插入到输出串Y中。本发明还涉及相应的去除过滤方法以及执行该方法所需的设备和仪器。

4. 发明申请

WO2005069491A2 METHOD OF FILTERING BINARY DATA 审中-公开
标题翻译：滤波二进制数据的方法
公开(公告)号：WO2005069491A2
公开(公告)日：2005-07-28
申请号：PCT/IT2005/000022
申请日：2005-01-18
申请人： ATOP INNOVATION S.P.A. , CARABINIERE, Luigi
发明人： CARABINIERE, Luigi
IPC分类号： H03M7/00
CPC分类号： H03H21/0043
摘要： The present invention concerns a method of filtering an input binary string W for obtaining an output binary string Y, the method being characterised in that it comprises, for each one of one or more first substrings wj having nj bits at least partially belonging to the input string W, the following steps: A. calculating K prediction substrings ujx, with K≥2, through K operations of multiplication of the first considered substring wj by K functions f k ; B. having a second substring wj+1, comprising n j +1 bits at least partially belonging to the input string W; C. calculating, for each one of the K prediction substrings u jk , a corresponding distance D j +l,k indicative of the deviation of the second substring wj+1 from the prediction substring u jk ; D. selecting the function fkDHmin of the K functions f k having the mìnimum distance Dj+1,k from the substring wj+1; and E. inserting a substring y j equal to the prediction substring u jkDHmin generated by the function fkDHmin selected in step D ìnto the output string Y . The present invention further concerns the corresponding defiltering method, and the apparatuses and instruments necessary for performing the method.
摘要翻译：本发明涉及一种过滤输入二进制串W以获得输出二进制串Y的方法，该方法的特征在于，对于具有至少部分属于输入的nj个比特的一个或多个第一子串wj中的每一个，字符串W，以下步骤：A.计算K个预测子串ujx，其中K> = 2，通过K运算将第一个考虑的子串wj乘以K个函数fk; B.具有第二子串wj + 1，包括至少部分属于输入串W的nj + 1位; C.计算K个预测子串ujk中的每一个，表示第二子串wj + 1与预测子串ujk的偏差的相应距离Dj + 1，k; D.从子串wj + 1中选择具有最小距离Dj + 1，k的K个函数fk的函数fkDHmin; 和E.插入等于在步骤D中选择的函数fkDHmin生成的预测子串ujkDHmin的子串yj到输出字符串Y.本发明还涉及相应的去过滤方法以及执行该方法所需的装置和仪器。

5. 发明申请

WO1994025961A1 TRANSMISSION SYSTEM COMPRISING AT LEAST A CODER 审中-公开
标题翻译：传输系统至少包含一个编码器
公开(公告)号：WO1994025961A1
公开(公告)日：1994-11-10
申请号：PCT/IB1994000087
申请日：1994-05-02
申请人： PHILIPS PATENTVERWALTUNG GMBH , PHILIPS NORDEN AB , PHILIPS ELECTRONICS N.V. , HOFMANN, Rudolf
发明人： PHILIPS PATENTVERWALTUNG GMBH , PHILIPS NORDEN AB , PHILIPS ELECTRONICS N.V.
IPC分类号： G10L09/14
CPC分类号： H03H21/0012 , G10L19/06 , G10L25/06 , G10L25/90 , H03H21/0043
摘要： A transmission system is proposed comprising at least a coder (101) for coding a signal (10a). The system furthermore comprises at least a transmitter (102), at least a receiver (103) and at least a decoder (104). At least for a coder (101) is at least provided an adaptive prediction filter (10A, 10B) by means of which speech or audio signals are coded while their bit rates are reduced. The filter coefficients for an adaptive prediction filter (10A, 10B) are recursively determined according to the Levinson-Durbin Recursion in fixed point arithmetic.
摘要翻译：提出了至少包括用于对信号（10a）进行编码的编码器（101）的传输系统。该系统还包括至少一个发射机（102），至少一个接收机（103）和至少一个解码器（104）。至少对于编码器（101）至少设置有自适应预测滤波器（10A，10B），通过该自适应预测滤波器，语音或音频信号被编码，同时它们的比特率被降低。用于自适应预测滤波器（10A，10B）的滤波器系数根据固定点算术中的列文森 - 杜宾迭代递归地确定。

6. 发明申请

WO2006101997A2 CHANNEL ESTIMATION ENHANCED LMS EQUALIZER 审中-公开
标题翻译：信道估计增强LMS均衡器
公开(公告)号：WO2006101997A2
公开(公告)日：2006-09-28
申请号：PCT/US2006/009556
申请日：2006-03-16
申请人： INTERDIGITAL TECHNOLOGY CORPORATION , PIETRASKI, Philip, J.
发明人： PIETRASKI, Philip, J.
IPC分类号： H03H7/40 , H03D1/04
CPC分类号： H03H21/0043 , H03H2021/0056 , H04L25/0226 , H04L25/03019 , H04L2025/03496 , H04L2025/03611
摘要： The present invention is related to an enhanced equalizer using channel estimation. A scaled version of a channel estimate is used as an expected average behavior of the product of a transmitted signal and a received signal to implement Griffith algorithm. The present invention also uses advance or prediction of a channel estimate to overcome the lag problem inherent in a least means square (LMS) algorithm in a time varying channel. Therefore, the present invention enables the use of a small step size while attaining the same tracking capability with a large step size. A channel estimate at some time in the future is used for updating equalizer filter tap coefficients. This may be performed with a prediction filter. Alternatively, a delay may be introduced in the input data to the filter tap coefficient generator, which makes a channel estimate look like a prediction to the filter tap coefficient generator.
摘要翻译：本发明涉及使用信道估计的增强均衡器。信道估计的缩放版本被用作发射信号和接收信号的乘积的预期平均行为以实现Griffith算法。本发明还使用信道估计的提前或预测来克服时变信道中最小均方（LMS）算法中固有的滞后问题。因此，本发明能够在以大的步长获得相同的跟踪能力的同时使用小的步长。将来某个时间的信道估计用于更新均衡器滤波器抽头系数。这可以用预测滤波器来执行。或者，可以将输入数据中的延迟引入到滤波器抽头系数发生器，这使得信道估计看起来像滤波器抽头系数发生器的预测。

7. 发明申请

WO2006017934A1 ADAPTIVE FILTERING USING FAST AFFINE PROJECTION ADAPTATION 审中-公开
标题翻译：自适应滤波使用快速AFFINE投影适配
公开(公告)号：WO2006017934A1
公开(公告)日：2006-02-23
申请号：PCT/CA2005/001261
申请日：2005-08-17
申请人： NATIONAL RESEARCH COUNCIL OF CANADA , DING, Heping
发明人： DING, Heping
IPC分类号： H03H21/00
CPC分类号： H03H21/0043 , H03H2021/0054 , H04B3/23 , H04B3/238
摘要： A method of adaptive filtering using fast affine projection (FAP) that allows for direct solution of the projected error vector from the autocorrelation matrix using a backward and forward recursion technique or LDL T factorization of the autocorrelation matrix followed by forward substitution, scaling and backward substitution. The method results in less computational complexity in the implementation, while still having good stability and fast convergence.
摘要翻译：使用快速仿射投影（FAP）的自适应滤波的方法，其允许使用向后和向前递归技术或LDL

8. 发明申请

WO2005027446A1 ADAPTIVE EQUALIZATION USING A CONDITIONAL UPDATE SIGN-SIGN LEAST MEAN SQUARE ALGORITHM 审中-公开
标题翻译：使用条件更新的自适应均衡签名最小均方算法
公开(公告)号：WO2005027446A1
公开(公告)日：2005-03-24
申请号：PCT/US2004/029175
申请日：2004-09-08
申请人： INTEL CORPORATION , BALAMURUGAN, Ganesh , CASPER, Bryan , JAUSSI, James , MOONEY, Stephen
发明人： BALAMURUGAN, Ganesh , CASPER, Bryan , JAUSSI, James , MOONEY, Stephen
IPC分类号： H04L25/03
CPC分类号： H03H21/0043 , H03H21/0001 , H03H2021/0065 , H04L25/03038 , H04L25/03885 , H04L2025/03477 , H04L2025/03617
摘要： An adaptive equalizer finite impulse response (FIR) filter for high-speed communication channels with modest complexity, where the filter is iteratively updated during a training sequence by a circuit performing the update: (I), where (II) is the filter vector representing the filter taps of the FIR filter, (III) is the data vector representing present and past samples of the received data x (t), d(t) is the desired data used for training, z(t) is the output of the FIR filter, µ determines the memory or window size of the adaptation, and K is a scale factor taking into account practical limitations of the communication channel, receiver, and equalizer. Furthermore, a procedure and circuit structure is provided for calibrating the scale factor K .
摘要翻译：用于具有适度复杂度的高速通信信道的自适应均衡器有限脉冲响应（FIR）滤波器，其中通过执行更新的电路在训练序列期间迭代地更新滤波器：（I），其中（II）是表示 FIR滤波器的滤波器抽头（III）是表示接收数据x（t）的当前和过去样本的数据向量，d（t）是用于训练的期望数据，z（t）是 FIR滤波器，μ确定适配的存储器或窗口大小，K是考虑通信信道，接收机和均衡器的实际限制的比例因子。此外，提供了用于校准比例因子K的过程和电路结构。

9. 发明申请

WO01082474A2 INITIALIZATION/PREWINDOWING REMOVAL POSTPROCESSING FOR FAST RLS FILTER ADAPTATION 审中-公开
标题翻译：用于快速RLS滤波器适配的初始化/去除移除POSPOROCESS
公开(公告)号：WO01082474A2
公开(公告)日：2001-11-01
申请号：PCT/US2001/011724
申请日：2001-04-19
IPC分类号： H03H21/00 , H03H
CPC分类号： H03H21/0043 , H03H2021/0049
摘要： The present invention, generally speaking, accelerates convergence of a fast RLS adaptation algorithm by, following processing of a burst of data, performing postprocessing to remove the effects of prewindowing, fictitious data initialization, or both. This postprocessing is part of a burst mode adaptation strategy in which data (signals) get processed in chunks (bursts). Such a burst mode processing approach is applicable whenever the continuous adaptation of the filter is not possible (algorithmic complexity too high to run in real time) or not required (optimal filter setting varies only slowly with time). Postprocessing consists of a series of "downdating" operations (as opposed to updating) that in effect advance the beginning point of the data window. The beginning point is advanced beyond fictitious data used for initialization and beyond a prewindowing region. In other variations, downdating is applied to data within a prewindowing region only. The forgetting factor of conventional algorithms can be eliminated entirely. Performance equivalent to that of GWC RLS algorithms is achieved at substantially lower computational cost. In particular, a postprocessing Fast Kalman Algorithm in effect transforms an initialized/prewindowed least squares estimate into a Covariance Window least squares estimate. Various further refinements are possible. Initialization may be cancelled completely or only partially. For example, in order to reduce the dynamic range of algorithmic quantities, it may be advantageous to, in a subsequent initialization, add an increment to a forward error energy quantity calculated during a previous burst. Postprocessing may then be performed to cancel only the added increment. Also, to reduce the usual large startup error transient, the desired response data can be modified in a way that dampens the error transient. The modified desired response data are saved for use in later postprocessing. Furthermore, to allow for more rapid adaptation without the use of an exponential forgetting factor, a weighting factor less than one may be applied to the forward error energy quantity during initialization from one burst to the next. This allows for the most efficient use of data but limited adaptation within a burst, but more rapid adaptation from one burst to the next.
摘要翻译：一般而言，本发明通过在数据突发的处理之后加速快速RLS自适应算法的收敛，执行后处理以去除预窗口化，虚拟数据初始化或两者的影响。该后处理是突发模式适应策略的一部分，其中数据（信号）以块（突发）进行处理。只要滤波器的连续自适应不可能（算法复杂度太高而不能实时运行），或者不需要（优化滤波器设置随时间变化缓慢），这种突发模式处理方法是适用的。后处理包括一系列“缩减”操作（而不是更新），其实际上提高了数据窗口的起点。起始点超出用于初始化的虚拟数据，超出了预画面区域。在其他变体中，只有在预开窗区域内的数据才应用缩减时间。传统算法的遗忘因素可以完全消除。性能相当于GWC RLS算法的性能实现在较低的计算成本。特别地，后处理的快速卡尔曼算法实际上将初始化/预先窗口化的最小二乘估计变换为协方差窗口最小二乘估计。各种进一步的改进是可能的。初始化可能会被完全取消或仅部分取消。例如，为了减少算法量的动态范围，在随后的初始化中可能有利的是将增量添加到在先前突发期间计算出的前向误差能量。然后可以执行后处理以仅取消添加的增量。此外，为了减少通常的大的启动错误瞬态，可以以抑制误差瞬变的方式修改所需的响应数据。修改的所需响应数据被保存以用于稍后的后处理。此外，为了允许在不使用指数遗忘因子的情况下进行更快速的适应，可以在从一个突发到下一个突发的初始化期间将小于一的加权因子应用于前向误差能量。这允许最有效地使用数据，但在突发中有限的自适应，但是从一个突发到下一个突发的更快速的适应。

10. 发明申请

WO1993000741A1 A METHOD FOR OPTIMIZING AN ADAPTIVE FILTER UPDATE COEFFICIENT 审中-公开
标题翻译：一种优化自适应滤波器更新系数的方法
公开(公告)号：WO1993000741A1
公开(公告)日：1993-01-07
申请号：PCT/US1992003767
申请日：1992-05-08
申请人： MOTOROLA, INC.
发明人： MOTOROLA, INC. , BAUM, Kevin, L.
IPC分类号： H03H07/30
CPC分类号： H03H21/0043
摘要： The method of the present invention generates an optimal adaptive filter update coefficient by first generating three error signals using a signal input and three update coefficients (301-303). Mean Square Error (MSE) values are estimated (307 and 308) for the first and third error signals. The first MSE value (307) is subtracted (309) from the third MSE value (308) to generate a difference signal. The difference signal is used to generate (310) an update signal that is used to modify the update coefficients. The process of the present invention is repeated until the difference signal is substantially zero, thus optimizing the second update coefficient. This process enables an update coefficient for an adaptive filter to quickly adapt to a changing environment.

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