共查询到20条相似文献，搜索用时 953 毫秒
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针对矿井中的回声消除的问题,采用自适应滤波的LMS算法进行解决,首先分析了自适应滤波算法的优缺点,其次针对LMS算法存在的问题进行改进,并从算法收敛性、抗噪性和稳定性进行了理论上分析,在仿真实验中,采用矿井的声音样本进行分析,从收敛性、计算量,输出结果进行分析,实验说明本文算法在回声消除方面具有一定的效果。 相似文献
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本文以蓝牙耳机为研究对象,旨在解决用户通话过程中的回音问题。本设计利用自适应回消除音技术,采用DSP芯片,运用LMS自适应算法进行软硬件设计,通过在麦克风的近端信号中减去回音估值,使近端传出的信号减小回音干扰,实现回音消除。最后,通过仿真信号实验测试,验证了设计结果的正确性。 相似文献
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传统的基于LMS算法的自适应陷波器,由于极易受到步长及其它参数的影响,学习曲线并不理想。文章在分析步长对基于LMS算法自适应陷波器的影响的基础上,将粒子群(PSO)算法应用到自适应陷波器的设计中。通过仿真结果显示,基于PSO算法的自适应陷波器收敛速度快、具有鲁棒性,优于传统的基于LMS的自适应陷波器,从而证明其有效性、可行性及工程价值。 相似文献
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Kenneth Abend 《Journal of The Franklin Institute》2002,339(3):283-294
The Franklin Institute, Philadelphia, Pennsylvania, awarded the Benjamin Franklin Medal in Engineering to Bernard Widrow, Professor of Electrical Engineering at Stanford University, for pioneering work in adaptive signal processing. Adaptive systems have the ability to learn and improve their behavior through interaction with their environments. Dr. Widrow developed the least mean squared (LMS) algorithm, which is a computationally facile means of finding the optimal weight vector for suppressing unknown noise. For example, every high-speed modem contains an adaptive filter or automatic equalizer based on the Widrow-Hoff LMS algorithm. Such a telephone channel equalizer makes it possible for computers to communicate at high speed (such as for the internet) over regular telephone lines, which were never intended for this purpose. Dr. Widrow was amongst the first to publish a general theory of adaptive antennas, including space-time processing. His adaptive learning algorithms made artificial neural networks possible. His latest invention is a directional hearing aid. 相似文献
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An LMS adaptive algorithm with a new step-size control equation 总被引：2，自引：0，他引：2
In this paper, we introduce a new variable step-size LMS (VSSLMS) adaptive algorithm. The algorithm step-size equations estimate an optimal derived step-size and are controlled by only one parameter. Mean-square performance analysis is provided for zero-mean stationary Gaussian input signal, and a simple expression that predicts the algorithm steady state misadjustment is derived for small step-size fluctuations. The algorithm is compared with other well-known VSSLMS algorithms through simulation experiments, which demonstrate the performance advantages of the proposed algorithm over these algorithms. 相似文献
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在海洋信道进行无线数字通信中,多径传播效应和频率选择性衰落会导致传输信号失真,信道均衡是降低这种失真的重要手段。本文将详细论述根据LMS算法的基本原理来构建自适应均衡器的过程,并通过仿真来检验该算法的效果。 相似文献
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本文基于最小均方误差(least mean square,LMS)算法自适应滤波器的基本原理,介绍了一种在Xilinx公司System Generator开发环境中采用MATLAB语言建立算法模型并在FPGA实现的设计方法。整个设计在Xilinx Virtex-5sx50tf1 136型芯片下验证。相比使用传统硬件描述语言的设计方法,MATLAB语言具有编写灵活简单易调试、设计效率高等优点。该方法不但可以很好的完成设计指标,还有效地提高了FPGA系统级设计的效率,同时降低了设计人员对硬件底层结构知识的要求。 相似文献
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In this paper, we propose an adapt-then-combine (ATC) diffusion normalized Huber adaptive filtering (ATC-DNHuber) algorithm for distributed estimation in impulsive interference environments. Firstly, a normalized Huber adaptive filter (NHuber) is developed to reduce the effect of the eigenvalue spread of the input signal. Then we extend the NHuber to develop an ATC diffusion algorithm by applying the NHuber algorithm at all agents. In addition, the mean stability performance and computational complexity are analyzed theoretically. In addition, Furthermore, simulation results demonstrate that the ATC-DNHuber algorithm can perform better in identifying the unknown coefficients under the complex and changeable impulsive interference environments. 相似文献
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本文提出一种新的基于α稳定分布噪声环境下的自适应滤波算法,这种算法针对变步长自适应滤波算法收敛速度和稳态误差相矛盾的不足,建立了步长μ(n)与误差信号e(n)之间的新的非线性函数关系。该函数能够削弱输入端不相关α稳定分布噪声对步长调整的影响,更好地解决稳态误差与收敛时间之间的矛盾。通过系统辨识仿真结果表明,新的算法α对稳定分布下的尖峰脉冲噪声有较强的韧性,比传统的NLMP算法有更快的参数辨识速度和更小的稳态误差,同时还具有很好地跟踪多时变系统的能力。 相似文献
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提出一种基于小波包与自适应预测器的音频隐写分析方法,主要用于检测加性噪声模型.利用加性噪声对音频高频部分比低频部分影响显著的特点,对音频信号进行小波包分解;然后利用最小均方(LMS)自适应预测器对高频小波包系数进行预测,选取预测误差的统计量作为统计特征;最后采用支持向量机分类.实验证明,对于常用的加性噪声隐写方法,即使在嵌入强度或嵌入率较低的情况下,也能达到较高的分类准确率. 相似文献
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多用户检测主要应用于CDMA接收机设计中以消除多址干扰（MAI）。盲自适应多用户检测不需要训练序列,只需要知道期望用户的特征波形。LSCMA是一种常用的恒模算法,具有收敛速度快的优点。论文将LSCMA应用于DS—CDMA系统中,对期望用户的信号进行检测,并通过仿真验证了算法的可行性。 相似文献
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《Journal of The Franklin Institute》2023,360(12):7645-7675
Recently, the augmented complex-valued normalized subband adaptive filtering (ACNSAF) algorithm has been proposed to process colored non-circular signals. However, its performance will deteriorate severely under impulsive noise interference. To overcome this issue, a robust augmented complex-valued normalized M-estimate subband adaptive filtering (ACNMSAF) algorithm is proposed, which is obtained by modifying the subband constraints of the ACNSAF algorithm using the complex-valued modified Huber (MH) function and is derived based on CR calculus and Lagrange multipliers. In order to improve both the convergence speed and steady-state accuracy of the fixed step size ACNMSAF algorithm, a variable step size (VSS) strategy based on the minimum mean squared deviation (MSD) criterion is devised, which allocates individual adaptive step size to each subband, fully exploiting the structural advantages of SAF and significantly improving the convergence performance of the ACNMSAF algorithm as well as its tracking capability in non-stationary environment. Then, the stability, transient and steady-state MSD performance of the ACNMSAF algorithm in the presence of colored non-circular inputs and impulsive noise are analyzed, and the stability conditions, transient and steady-state MSD formulas are also derived. Computer simulations in impulsive noise environments verify the accuracy of theoretical analysis results and the effectiveness of the proposed algorithms compared to other existing complex-valued adaptive algorithms. 相似文献