随着先进工艺和技术的不断进步,要想保证数据在高速传输中的正确性,均衡器需要有更高的补偿和更低的功耗,才能实现高效通信。基于12 nm互补金属氧化物半导体工艺,设计了一种高增益、低功耗的自适应连续时间线性均衡器(continuous time l...随着先进工艺和技术的不断进步,要想保证数据在高速传输中的正确性,均衡器需要有更高的补偿和更低的功耗,才能实现高效通信。基于12 nm互补金属氧化物半导体工艺,设计了一种高增益、低功耗的自适应连续时间线性均衡器(continuous time linear equalizer,CTLE),该均衡器采用2级级联结构来补偿信道衰减,并提高接收信号的质量。此外,自适应模块通过采用符号-符号最小均方误差(sign-sign least mean square,SS-LMS)算法,使抽头系数加快了收敛速度。仿真结果表明,当传输速率为16 Gbit/s时,均衡器可以补偿-15.53 dB的半波特率通道衰减,均衡器系数在16×10^(4)个单元间隔数据内收敛,并且收敛之后接收误码率低于10^(-12)。展开更多
Random vibration control is aimed at reproducing the power spectral density （PSD） at specified control points. The classical frequency-spectrum equalization algorithm needs to compute the average of the multiple fre...Random vibration control is aimed at reproducing the power spectral density （PSD） at specified control points. The classical frequency-spectrum equalization algorithm needs to compute the average of the multiple frequency response functions （FRFs）, which lengthens the control loop time in the equalization process. Likewise, the feedback control algorithm has a very slow convergence rate due to the small value of the feedback gain parameter to ensure stability of the system. To overcome these limitations, an adaptive inverse control of random vibrations based on the filtered-X least mean-square （LMS） algorithm is proposed. Furthermore, according to the description and iteration characteristics of random vibration tests in the frequency domain, the frequency domain LMS algorithm is adopted to refine the inverse characteristics of the FRF instead of the traditional time domain LMS algorithm. This inverse characteristic, which is called the impedance function of the system under control, is used to update the drive PSD directly. The test results indicated that in addition to successfully avoiding the instability problem that occurs during the iteration process, the adaptive control strategy minimizes the amount of time needed to obtain a short control loop and achieve equalization.展开更多
Underwater acoustic channels are recognized for being one of the most difficult propagation media due to considerable difficulties such as： multipath, ambient noise, time-frequency selective fading. The exploitation ...Underwater acoustic channels are recognized for being one of the most difficult propagation media due to considerable difficulties such as： multipath, ambient noise, time-frequency selective fading. The exploitation of sparsity contained in underwater acoustic channels provides a potential solution to improve the performance of underwater acoustic channel estimation. Compared with the classic 10 and 11 norm constraint LMS algorithms, the p-norm-like （Ip） constraint LMS algorithm proposed in our previous investigation exhibits better sparsity exploitation performance at the presence of channel variations, as it enables the adaptability to the sparseness by tuning of p parameter. However, the decimal exponential calculation associated with the p-norm-like constraint LMS algorithm poses considerable limitations in practical application. In this paper, a simplified variant of the p-norm-like constraint LMS was proposed with the employment of Newton iteration m to approximate the decimal exponential calculation. Num simulations and the experimental results obtained in physical shallow water channels demonstrate the effectiveness of the proposed method compared to traditional norm constraint LMS algorithms.展开更多
The least means squares （LMS） adaptive filter algorithm was used in active suspension system. By adjusting the weight of adaptive filter, the minimum quadratic performance index was obtained. For two-degree-of-freed...The least means squares （LMS） adaptive filter algorithm was used in active suspension system. By adjusting the weight of adaptive filter, the minimum quadratic performance index was obtained. For two-degree-of-freedom vehicle suspension model, LMS adaptive controller was designed. The acceleration of the sprung mass,the dynamic tyre load between wheels and road,and the dynamic deflection between sprung mass and unsprung mass were determined as the evaluation targets of suspension performance. For LMS adaptive control suspension, compared with passive suspension, acceleration power spectral density of sprung mass acceleration under the road input model decreased 8-10 times in high frequency resonance band or low frequency resonance band. The simulation results show that LMS adaptive control is simple and remarkably effective. It further proves that the active control suspension system can improve both the riding comfort and handling safety in various operation conditions, and the method is fit for the active control of the suspension system.展开更多
The contradiction of variable step size least mean square(LMS)algorithm between fast convergence speed and small steady-state error has always existed.So,a new algorithm based on the combination of logarithmic and sym...The contradiction of variable step size least mean square(LMS)algorithm between fast convergence speed and small steady-state error has always existed.So,a new algorithm based on the combination of logarithmic and symbolic function and step size factor is proposed.It establishes a new updating method of step factor that is related to step factor and error signal.This work makes an analysis from 3 aspects:theoretical analysis,theoretical verification and specific experiments.The experimental results show that the proposed algorithm is superior to other variable step size algorithms in convergence speed and steady-state error.展开更多
针对风洞试验模型系统辨识不准确的问题,利用自适应LMS(least mean square)滤波器模型对跨声速风洞模型进行系统辨识。由于实测信号中存在多模态耦合,为了提高系统辨识精准度,首先对输入输出信号作了FRF(frequency response analysis)...针对风洞试验模型系统辨识不准确的问题,利用自适应LMS(least mean square)滤波器模型对跨声速风洞模型进行系统辨识。由于实测信号中存在多模态耦合,为了提高系统辨识精准度,首先对输入输出信号作了FRF(frequency response analysis)分析得到试验模型俯仰方向前两阶模态,其次利用快速Fourier变换进行模态解耦,接着利用自适应LMS滤波器模型、传递函数模型、多项式模型对俯仰方向单模态进行系统辨识,最后得到了基于自适应LMS滤波器模型的俯仰方向一阶、二阶模态滤波器系数。通过对比不同数学模型的输出与输入之间的相关系数和均方误差及辨识结果,表明自适应LMS滤波器模型具有更高的系统辨识精准度和更简洁的数学模型结构。为后续风洞试验模型振动主动控制计算法的设计提供有力支撑。展开更多
This paper puts forward a new variable step size LMS adaptive algorithm based on variable region. The step size p（k） in the algorithm varies with the variation of the region of deviation e （k） to ensure the optimi...This paper puts forward a new variable step size LMS adaptive algorithm based on variable region. The step size p（k） in the algorithm varies with the variation of the region of deviation e （k） to ensure the optimization of the three performance objectives including initial convergent speed, trace ability of the time-varying system and steady disregulation. The paper demonstrates the convergence of the algorithm accompanied by random noise,展开更多
Based on the good time-frequency localization of the wavelet packet modulation （WPM） system and the analysis of the muhitone （MT） interference, an improved WPM narrow band interference （NBI） rejection system usi...Based on the good time-frequency localization of the wavelet packet modulation （WPM） system and the analysis of the muhitone （MT） interference, an improved WPM narrow band interference （NBI） rejection system using an adaptive least mean square （LMS） algorithm is discussed in this paper. In the WPM interference rejection system, according to the structure of the wavelet packet tree （WFF） and the characteristic of the MT interference, the demodulated WPM signal which contains the interference component is filtered using an adaptive LMS filtering implementation. Theoretical analysis and simulation resuits show that the proposed method improves the bit error rate （BER） performance of the WPM system in the presence of MT interference.展开更多
In this paper,the inter-symbol interference and eliminating method are introduced.After analyzing the principle of adaptive equalization,we designed an adaptive equalizer using the LMS algorithm,and constructed a simu...In this paper,the inter-symbol interference and eliminating method are introduced.After analyzing the principle of adaptive equalization,we designed an adaptive equalizer using the LMS algorithm,and constructed a simulation system using MATLAB.Then we analyzed the convergence speed and mean square error characteristic of the adaptive equalizer by changing the step length factor to test the performance of the algorithm.展开更多
文摘随着先进工艺和技术的不断进步,要想保证数据在高速传输中的正确性,均衡器需要有更高的补偿和更低的功耗,才能实现高效通信。基于12 nm互补金属氧化物半导体工艺,设计了一种高增益、低功耗的自适应连续时间线性均衡器(continuous time linear equalizer,CTLE),该均衡器采用2级级联结构来补偿信道衰减,并提高接收信号的质量。此外,自适应模块通过采用符号-符号最小均方误差(sign-sign least mean square,SS-LMS)算法,使抽头系数加快了收敛速度。仿真结果表明,当传输速率为16 Gbit/s时,均衡器可以补偿-15.53 dB的半波特率通道衰减,均衡器系数在16×10^(4)个单元间隔数据内收敛,并且收敛之后接收误码率低于10^(-12)。
基金Program for New Century Excellent Talents in Universities Under Grant No.NCET-04-0325
文摘Random vibration control is aimed at reproducing the power spectral density （PSD） at specified control points. The classical frequency-spectrum equalization algorithm needs to compute the average of the multiple frequency response functions （FRFs）, which lengthens the control loop time in the equalization process. Likewise, the feedback control algorithm has a very slow convergence rate due to the small value of the feedback gain parameter to ensure stability of the system. To overcome these limitations, an adaptive inverse control of random vibrations based on the filtered-X least mean-square （LMS） algorithm is proposed. Furthermore, according to the description and iteration characteristics of random vibration tests in the frequency domain, the frequency domain LMS algorithm is adopted to refine the inverse characteristics of the FRF instead of the traditional time domain LMS algorithm. This inverse characteristic, which is called the impedance function of the system under control, is used to update the drive PSD directly. The test results indicated that in addition to successfully avoiding the instability problem that occurs during the iteration process, the adaptive control strategy minimizes the amount of time needed to obtain a short control loop and achieve equalization.
基金Supported by the National Natural Science Foundation of China （No.11274259） and the Specialized Research Foundation for the Doctoral Program of Higher Education of China （No.20120121110030）.
文摘Underwater acoustic channels are recognized for being one of the most difficult propagation media due to considerable difficulties such as： multipath, ambient noise, time-frequency selective fading. The exploitation of sparsity contained in underwater acoustic channels provides a potential solution to improve the performance of underwater acoustic channel estimation. Compared with the classic 10 and 11 norm constraint LMS algorithms, the p-norm-like （Ip） constraint LMS algorithm proposed in our previous investigation exhibits better sparsity exploitation performance at the presence of channel variations, as it enables the adaptability to the sparseness by tuning of p parameter. However, the decimal exponential calculation associated with the p-norm-like constraint LMS algorithm poses considerable limitations in practical application. In this paper, a simplified variant of the p-norm-like constraint LMS was proposed with the employment of Newton iteration m to approximate the decimal exponential calculation. Num simulations and the experimental results obtained in physical shallow water channels demonstrate the effectiveness of the proposed method compared to traditional norm constraint LMS algorithms.
文摘The least means squares （LMS） adaptive filter algorithm was used in active suspension system. By adjusting the weight of adaptive filter, the minimum quadratic performance index was obtained. For two-degree-of-freedom vehicle suspension model, LMS adaptive controller was designed. The acceleration of the sprung mass,the dynamic tyre load between wheels and road,and the dynamic deflection between sprung mass and unsprung mass were determined as the evaluation targets of suspension performance. For LMS adaptive control suspension, compared with passive suspension, acceleration power spectral density of sprung mass acceleration under the road input model decreased 8-10 times in high frequency resonance band or low frequency resonance band. The simulation results show that LMS adaptive control is simple and remarkably effective. It further proves that the active control suspension system can improve both the riding comfort and handling safety in various operation conditions, and the method is fit for the active control of the suspension system.
基金the National Natural Science Foundation of China(No.51575328,61503232).
文摘The contradiction of variable step size least mean square(LMS)algorithm between fast convergence speed and small steady-state error has always existed.So,a new algorithm based on the combination of logarithmic and symbolic function and step size factor is proposed.It establishes a new updating method of step factor that is related to step factor and error signal.This work makes an analysis from 3 aspects:theoretical analysis,theoretical verification and specific experiments.The experimental results show that the proposed algorithm is superior to other variable step size algorithms in convergence speed and steady-state error.
文摘针对风洞试验模型系统辨识不准确的问题,利用自适应LMS(least mean square)滤波器模型对跨声速风洞模型进行系统辨识。由于实测信号中存在多模态耦合,为了提高系统辨识精准度,首先对输入输出信号作了FRF(frequency response analysis)分析得到试验模型俯仰方向前两阶模态,其次利用快速Fourier变换进行模态解耦,接着利用自适应LMS滤波器模型、传递函数模型、多项式模型对俯仰方向单模态进行系统辨识,最后得到了基于自适应LMS滤波器模型的俯仰方向一阶、二阶模态滤波器系数。通过对比不同数学模型的输出与输入之间的相关系数和均方误差及辨识结果,表明自适应LMS滤波器模型具有更高的系统辨识精准度和更简洁的数学模型结构。为后续风洞试验模型振动主动控制计算法的设计提供有力支撑。
基金Supported by Natural Science Foundation of Beijing of China (No.2005AA501140)
文摘This paper puts forward a new variable step size LMS adaptive algorithm based on variable region. The step size p（k） in the algorithm varies with the variation of the region of deviation e （k） to ensure the optimization of the three performance objectives including initial convergent speed, trace ability of the time-varying system and steady disregulation. The paper demonstrates the convergence of the algorithm accompanied by random noise,
基金Supported by the National Natural Science Foundation of China （No. 60532030）.
文摘Based on the good time-frequency localization of the wavelet packet modulation （WPM） system and the analysis of the muhitone （MT） interference, an improved WPM narrow band interference （NBI） rejection system using an adaptive least mean square （LMS） algorithm is discussed in this paper. In the WPM interference rejection system, according to the structure of the wavelet packet tree （WFF） and the characteristic of the MT interference, the demodulated WPM signal which contains the interference component is filtered using an adaptive LMS filtering implementation. Theoretical analysis and simulation resuits show that the proposed method improves the bit error rate （BER） performance of the WPM system in the presence of MT interference.
文摘In this paper,the inter-symbol interference and eliminating method are introduced.After analyzing the principle of adaptive equalization,we designed an adaptive equalizer using the LMS algorithm,and constructed a simulation system using MATLAB.Then we analyzed the convergence speed and mean square error characteristic of the adaptive equalizer by changing the step length factor to test the performance of the algorithm.
