公开(公告)号：US07027501B1

公开(公告)日：2006-04-11

申请号：US10084580

申请日：2002-02-27

申请人： Adya S. Tripathi ,  Delon Hanson ,  Kar Shing Chiu ,  Ming-Tak Leung ,  Raman Dakshinamurthy ,  Ki Chun Fu

发明人： Adya S. Tripathi ,  Delon Hanson ,  Kar Shing Chiu ,  Ming-Tak Leung ,  Raman Dakshinamurthy ,  Ki Chun Fu

IPC分类号： H03H7/30 ,  H03H7/40 ,  H03K5/159

CPC分类号： H04L25/03057 ,  H03H21/0012 ,  H03H2021/0092 ,  H04L2025/03471 ,  H04L2025/03617 ,  H04L2025/037

摘要： According to the present invention, methods and apparatus are provided for improving the signal quality of received transmission. An adaptive equalizer includes multiple output delay lines. One output delay line is configured to provide gradient elements. Another output delay line is configured with coefficient multipliers calculated using the gradient elements. The coefficient multipliers are used to alter a received signal to more closely correspond to an expected signal. The adaptive equalizer can be used in systems such as optical transceivers.

摘要翻译： 根据本发明，提供了用于提高接收到的传输的信号质量的方法和装置。 自适应均衡器包括多个输出延迟线。 一个输出延迟线被配置为提供梯度元素。 另一个输出延迟线由使用梯度元素计算的系数乘数配置。 系数乘法器用于改变接收信号以更接近于预期信号。 自适应均衡器可用于诸如光收发器的系统中。

公开(公告)号：US20130308960A1

公开(公告)日：2013-11-21

申请号：US13983233

申请日：2012-02-07

申请人： Kengo Horikoshi ,  Etsushi Yamazaki ,  Takayuki Kobayashi ,  Eiji Yoshida ,  Yutaka Miyamoto

发明人： Kengo Horikoshi ,  Etsushi Yamazaki ,  Takayuki Kobayashi ,  Eiji Yoshida ,  Yutaka Miyamoto

IPC分类号： H04B1/10

CPC分类号： H04B1/10 ,  H03H21/0012 ,  H03H2021/0092 ,  H04B10/616 ,  H04L25/03261

摘要： A parameter of an adaptive filter is optimized so that inter-symbol interference having an amount corresponding to an inserted fixed filter remains. A digital signal processing apparatus which is included in an optical signal receiver and processes a digital signal converted from an optical signal is provided with: a linear adaptive filter which applies a dynamically controllable linear transfer function to the digital signal; a maximum likelihood sequence decoder which applies a transfer function of a transmission-path model to a plurality of signal sequence candidates to generate a plurality of reference signals, and decodes a reception signal using maximum likelihood sequence estimation which evaluates the differences between an output signal of the linear adaptive filter and the reference signals to estimate the most likely transmission time sequence; a signal regenerator which generates a signal corresponding to decoded data from the maximum likelihood sequence decoder; a feedback distortion adding filter which adds distortion that is equivalent to the transmission-path model used in the maximum likelihood sequence decoder to an output signal of the signal regenerator; and an adaptive equalization filter control block which updates a tap coefficient of the linear adaptive filter in accordance with an LMS algorithm using the difference between a target signal that is an output signal of the feedback distortion adding filter and the digital signal as an error signal.

摘要翻译： 对自适应滤波器的参数进行优化，使得具有与插入的固定滤波器相对应的量的符号间干扰保留。 包括在光信号接收机中并处理从光信号转换的数字信号的数字信号处理装置具有：向数字信号施加动态可控线性传递函数的线性自适应滤波器; 最大似然序列解码器，其将传输路径模型的传递函数应用于多个信号序列候选以产生多个参考信号，并且使用最大似然序列估计解码接收信号，所述最大似然序列估计用于评估接收信号的输出信号 线性自适应滤波器和参考信号来估计最可能的传输时间序列; 信号再生器，其生成对应于来自最大似然序列解码器的解码数据的信号; 将与最大似然序列解码器中使用的传输路径模型相当的失真与信号再生器的输出信号相加的反馈失真相加滤波器; 以及自适应均衡滤波器控制块，其使用作为反馈失真相加滤波器的输出信号的目标信号和数字信号之间的差作为误差信号，根据LMS算法来更新线性自适应滤波器的抽头系数。

公开(公告)号：US07881482B2

公开(公告)日：2011-02-01

申请号：US11434496

申请日：2006-05-15

申请人： Markus Christoph

发明人： Markus Christoph

IPC分类号： H03G5/00

CPC分类号： H03H21/0012 ,  H03H21/0014 ,  H03H2021/0041 ,  H03H2021/0092

摘要： An audio enhancement system is provided for compensating for distortions (e.g., linear distortions) of a sound signal reproduced by an audio system in a listening room. The audio enhancement system includes analysis filters that generate a plurality of analysis output signals from an audio signal to be enhanced. The system also includes synthesis filters that generate an enhanced audio signal from a number of synthesis input signals. The number of analysis output signals and the number of synthesis input signals preferably are equal. Signal processing elements between the analysis filters and the synthesis filters generate one of the synthesis input signals from a respective one of the analysis output signals to perform an inverse filtering for linearizing an unknown transfer function indicative of the audio system and the listening room in the respective frequency range.

摘要翻译： 提供了一种音频增强系统，用于补偿由听音室中的音频系统再现的声音信号的失真（例如，线性失真）。 音频增强系统包括从要增强的音频信号产生多个分析输出信号的分析滤波器。 该系统还包括从多个合成输入信号产生增强音频信号的合成滤波器。 分析输出信号的数量和合成输入信号的数量优选地相等。 分析滤波器和合成滤波器之间的信号处理元件产生来自分析输出信号中的相应一个的合成输入信号之一，以执行逆滤波，以将表示音频系统的未知传输函数和收听室线性化 频率范围。

公开(公告)号：US06621863B1

公开(公告)日：2003-09-16

申请号：US09488550

申请日：2000-01-21

申请人： Jun Ido

发明人： Jun Ido

IPC分类号： H03K5159

CPC分类号： H03H21/0012 ,  H03H2021/0092 ,  H04L25/03038 ,  H04L2025/03382 ,  H04L2025/03401 ,  H04L2025/0342 ,  H04L2025/03477 ,  H04L2025/03611

摘要： An equalizer providing a filtered signal to a data decision unit calculates an amplitude error signal by comparing the filtered signal with the data signal output from the data decision unit, and calculates a squared envelope error signal from the filtered signal. These two error signals are separately weighted according to the absolute value of the amplitude error, the weight of the amplitude error signal decreasing and the weight of the squared envelope error signal increasing as the absolute value of the amplitude error increases. The weighted amplitude error signal and weighted squared envelope error signal are added to obtain an error signal used in updating filter coefficients in the equalizer. Rapid convergence of the filter coefficients is obtained, with small residual error.

摘要翻译： 向数据判定单元提供滤波后的信号的均衡器通过将经滤波的信号与从数据判定单元输出的数据信号进行比较来计算振幅误差信号，并根据滤波后的信号计算平方包络误差信号。 这两个误差信号根据振幅误差的绝对值，振幅误差信号的权重减小，平方包络误差信号的权重随振幅误差的绝对值增加而分别加权。 加权幅度误差信号和加权平方包络误差信号被加入以获得用于更新均衡器中的滤波器系数的误差信号。 获得滤波器系数的快速收敛，残差小。

公开(公告)号：US09679260B2

公开(公告)日：2017-06-13

申请号：US14220755

申请日：2014-03-20

申请人： Huawei Technologies Co., Ltd.

发明人： Lan Hu ,  Sai Mohan Kilambi ,  Ming Jia

IPC分类号： G06N99/00 ,  H03H21/00

CPC分类号： G06N99/005 ,  G06N99/00 ,  H03H21/0012 ,  H03H21/0043 ,  H03H2021/0092

摘要： In one embodiment, a method for training an adaptive filter includes receiving, by a processor from a device, an input signal and a training reference signal and determining a correlation matrix in accordance with the input signal, the training reference signal, and a filter type. The method also includes determining a plurality of coefficients in accordance with the correlation matrix and adjusting the adaptive filter in accordance with the plurality of coefficients.

公开(公告)号：US20150333783A1

公开(公告)日：2015-11-19

申请号：US14381223

申请日：2013-01-10

申请人： NEC Corporation

发明人： Junichi ABE

IPC分类号： H04B1/10 ,  H04L27/10 ,  H04B1/12 ,  H03H17/02 ,  H03H21/00

CPC分类号： H04B1/1027 ,  H03H17/0213 ,  H03H21/0027 ,  H03H2021/0092 ,  H04B1/123 ,  H04L27/10 ,  H04L27/265

摘要： A Fourier transform unit (111) performs Fourier transform on a digital signal on a time axis to generate a frequency domain signal which is a signal on a frequency axis. A filter unit (113) equalizes the frequency domain signal in a frequency domain using N first coefficients. An inverse Fourier transform unit (112) performs inverse Fourier transform on the frequency domain signal processed by the filter unit (113) and returns the frequency domain signal to the digital signal on the time axis. That is, the Fourier transform unit (111), the inverse Fourier transform unit (112), and the filter unit (113) compensate for waveform distortion included in the digital signal using an equalization process (that is, frequency-domain equalization (FDE)) in the frequency domain. A first coefficient setting unit (114) sets N first coefficients used by the filter unit (113) using m (provided N>m) second coefficients.

摘要翻译： 傅立叶变换单元（111）对时间轴上的数字信号进行傅立叶变换，生成频域上的信号的频域信号。 滤波器单元（113）使用N个第一系数来均衡频域中的频域信号。 逆傅立叶变换单元（112）对由滤波器单元（113）处理的频域信号进行傅立叶逆变换，并将频域信号在时间轴上返回到数字信号。 也就是说，傅立叶变换单元（111），逆傅里叶变换单元（112）和滤波器单元（113）使用均衡处理（即，频域均衡（FDE））来补偿包括在数字信号中的波形失真 ））。 第一系数设置单元（114）使用m（提供的N> m）个第二系数来设置由滤波器单元（113）使用的N个第一系数。

公开(公告)号：US20150269493A1

公开(公告)日：2015-09-24

申请号：US14220755

申请日：2014-03-20

申请人： FutureWei Technologies, Inc.

发明人： Lan Hu ,  Sai Mohan Kilambi ,  Ming Jia

IPC分类号： G06N99/00 ,  H03H21/00

CPC分类号： G06N99/005 ,  G06N99/00 ,  H03H21/0012 ,  H03H21/0043 ,  H03H2021/0092

摘要： In one embodiment, a method for training an adaptive filter includes receiving, by a processor from a device, an input signal and a training reference signal and determining a correlation matrix in accordance with the input signal, the training reference signal, and a filter type. The method also includes determining a plurality of coefficients in accordance with the correlation matrix and adjusting the adaptive filter in accordance with the plurality of coefficients.

摘要翻译： 在一个实施例中，用于训练自适应滤波器的方法包括由处理器从设备接收输入信号和训练参考信号，并根据输入信号，训练参考信号和滤波器类型来确定相关矩阵 。 该方法还包括根据相关矩阵确定多个系数，并根据多个系数调整自适应滤波器。

公开(公告)号：US20050218973A1

公开(公告)日：2005-10-06

申请号：US10516275

申请日：2003-05-30

申请人： Mirsad Halimic

发明人： Mirsad Halimic

IPC分类号： H03H21/00 ,  H03J3/00 ,  H04L25/03 ,  H03B1/00

CPC分类号： H03J3/00 ,  H03H21/0012 ,  H03H2021/0092 ,  H04L25/03057 ,  H04L25/0307 ,  H04L2025/03592

摘要： In digital communications, a considerable effort has been devoted to neutralise the effect of channels (i.e., the combination of transmit filters, media and receive filters) in transmission systems, so that the available channel bandwidth is utilised efficiently. The objective of channel neutralisation is to design a system that accommodates the highest possible rate of data transmission, subject to a specified reliability, which is usually measured in terms of the error rate or average probability of symbol error. An equaliser normally performs neutralisation of any disturbances the channel may introduce by malting the overall frequency response function T(z) to be flat. Since a channel is time varying, due to variations in a transmission medium, the received signal is nonstationary. Therefore, an adaptive equaliser is utilised to provide control over the time response of a channel. Since an adaptive equaliser is an inverse system of a channel, it amplifies the frequency of noise outside the bandwidth of a channel. In order to reduce the effect of noise, a low pass filter is cascaded with the equaliser. However, the cascaded filter can introduce a negative impact on the speed of adaptation. Therefore, the bandwidth of the cascaded filter is chosen to be very wide at the beginning of the adaptation process. This way, the output reaching the static value will not be delayed. As the output of the adaptive filter is close to the static value, the bandwidth decreases to cancel the effect of noise. The adaptive rule for noise filter can be defined as (I). The constants α and β depend on the level of noise and are chosen by trial and error method. Δ is a variable that is used to change the value of τ and consequently the bandwidth of the filter. Δ acts as an input to the proportional controller. Furthermore, in the same equation, β represents a proportional (P) controller gain (Kp). In order to reduce the disturbance rejection bandwidth, improve speed, resonant frequency and rectify a potential problem, an integral (I) control mode and a differential (D) control mode are proposed to be added to the existing proportional control mode.

公开(公告)号：US09385766B2

公开(公告)日：2016-07-05

申请号：US14381223

申请日：2013-01-10

申请人： NEC Corporation

发明人： Junichi Abe

IPC分类号： H04B15/00 ,  H04B1/10 ,  H04B1/12 ,  H04L27/10 ,  H03H17/02 ,  H03H21/00

CPC分类号： H04B1/1027 ,  H03H17/0213 ,  H03H21/0027 ,  H03H2021/0092 ,  H04B1/123 ,  H04L27/10 ,  H04L27/265

摘要： A Fourier transform unit (111) performs Fourier transform on a digital signal on a time axis to generate a frequency domain signal which is a signal on a frequency axis. A filter unit (113) equalizes the frequency domain signal in a frequency domain using N first coefficients. An inverse Fourier transform unit (112) performs inverse Fourier transform on the frequency domain signal processed by the filter unit (113) and returns the frequency domain signal to the digital signal on the time axis. That is, the Fourier transform unit (111), the inverse Fourier transform unit (112), and the filter unit (113) compensate for waveform distortion included in the digital signal using an equalization process (that is, frequency-domain equalization (FDE)) in the frequency domain. A first coefficient setting unit (114) sets N first coefficients used by the filter unit (113) using m (provided N>m) second coefficients.

摘要翻译： 傅立叶变换单元（111）对时间轴上的数字信号进行傅立叶变换，生成频域上的信号的频域信号。 滤波器单元（113）使用N个第一系数来均衡频域中的频域信号。 逆傅立叶变换单元（112）对由滤波器单元（113）处理的频域信号进行傅立叶逆变换，并将频域信号在时间轴上返回到数字信号。 也就是说，傅立叶变换单元（111），逆傅里叶变换单元（112）和滤波器单元（113）使用均衡处理（即，频域均衡（FDE））来补偿包括在数字信号中的波形失真 ））。 第一系数设置单元（114）使用m（提供的N> m）个第二系数来设置由滤波器单元（113）使用的N个第一系数。

公开(公告)号：US08977666B2

公开(公告)日：2015-03-10

申请号：US13629243

申请日：2012-09-27

申请人： Realtek Semiconductor Corp.

发明人： Cheng-Yi Huang ,  Yi-Lin Li

IPC分类号： G06F17/10 ,  H04L25/03 ,  H03H21/00

CPC分类号： H04L25/03019 ,  H03H2021/0092

摘要： An adaptive filter is disclosed, having a plurality of computation groups, a plurality of computation circuits, a summation circuit, a slicer circuit, an updating circuit, and a control circuit. Each computation group corresponds to an equalization parameter and has a plurality of memory cells. When the corresponding equalization parameter of a computation group is greater than a predetermined value, the control circuit configures the computation group and the computation circuit to collaboratively generate an output of the computation group. The summation circuit sums up the outputs of the computation groups to produce a filter output. The slicer circuit generates a slicer output according to the filter output. The updating circuit updates the equalization parameters according to the filter output and the slicer output.

摘要翻译： 公开了具有多个计算组，多个计算电路，求和电路，限幅电路，更新电路和控制电路的自适应滤波器。 每个计算组对应于均衡参数并具有多个存储单元。 当计算组的相应均衡参数大于预定值时，控制电路将计算组和计算电路配置为协同生成计算组的输出。 求和电路对计算组的输出进行求和，以产生滤波器输出。 限幅电路根据滤波器输出产生限幅器输出。 更新电路根据滤波器输出和限幅器输出更新均衡参数。