The degree to which the performance of the adaptive lms transversal filter depends on the ratio of the extreme eigen-values of the input autocorrelation matrix is investigated. Subsequently, performance is related to ...
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The degree to which the performance of the adaptive lms transversal filter depends on the ratio of the extreme eigen-values of the input autocorrelation matrix is investigated. Subsequently, performance is related to the spectral properties of the input signal via a study of the relationship between the eigenvalues and the loci of the poles characterising the input signal.
This paper, based on the original lms algorithm, proposed a novel method with wavelet preprocessing and signal is used as reference input to improve the noise reduction effect of the adaptive noise cancellation system...
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ISBN:
(纸本)9780769551258
This paper, based on the original lms algorithm, proposed a novel method with wavelet preprocessing and signal is used as reference input to improve the noise reduction effect of the adaptive noise cancellation system. The new algorithm, in the presence of impulse noise, low SNR environment, has a higher amount of noise reduction than the original lms algorithm based on adaptive noise cancellation system with better effect in reducing noise. The method first used wavelet packet threshold noise reduction algorithms, including impulse detection and de-noising, to remove the impulse part of the noise. Then the residual noisy signal is eliminated by adaptive noise cancellation system based on lms which has signal as the reference input. Finally, the multi-group simulations are compared. Simulation results show that the improved adaptive noise cancellation system do have better noise reduction and practical value.
To extend the coverage of base stations, the same frequency repeater is widely used in modern communication network However, there exists coupling echo interference between transmitting antenna and receiving antenna o...
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ISBN:
(纸本)9781728124582
To extend the coverage of base stations, the same frequency repeater is widely used in modern communication network However, there exists coupling echo interference between transmitting antenna and receiving antenna of the repeater station, which affects the system stability and the signal quality. The echo interference is in the same frequency with the useful signals of base stations, which cannot be eliminated by traditional frequency-domain filtering. To address this problem, this paper proposes an adaptive echo cancellation algorithm by inducing an additional reference sequences, which is targeted to apply in the hardware platform of Fujian Jingao FDD-TLE same frequency repeater. In this paper, the working principle of the algorithm is described firstly;then the algorithm is analyzed by MATLAB simulation, and finally verified in FPGA. The comparisons between before and after echo cancellation using the proposed algorithm in time domain and frequency domain shows that this algorithm can effectively suppress echo interference, and the output signal spectrum is significantly improved. Compared with the adaptive filtering algorithm based on the transform domain, this algorithm does not need Fourier transform, which has lower computational complexity and takes less resource. Moreover, this algorithm has a wide application scope and does not depend on the specific modulation scheme of the relaying signal.
In this paper, we firstly introduce three conventional digital filtering methods and analyze their qualities of filtering. Then we use a new method based on self-adaptive digital filter and median filter for their def...
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ISBN:
(纸本)1424411351
In this paper, we firstly introduce three conventional digital filtering methods and analyze their qualities of filtering. Then we use a new method based on self-adaptive digital filter and median filter for their defects, and simulate it. It is demonstrated that not only the noises are successfully restrained, but also the useful information is preserved as much as possible, and the performance of the new method is superior to the others.
Combination of two adaptive filters working in parallel for achieving better performance both in term of convergence speed and excess mean square error (EMSE) has been considered by several researchers in recent past....
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ISBN:
(纸本)9781467359528;9781467359504
Combination of two adaptive filters working in parallel for achieving better performance both in term of convergence speed and excess mean square error (EMSE) has been considered by several researchers in recent past. Prominent among these include convex combination (where combinational weight factors are within the range [0 1], while summing up to one), affine combination (where the combinational weight factors are free from any range constraint, while still summing up to one) and unconstrained model combination (where the output of constituent filters are combined using another adaptive algorithm). In this paper, we propose a novel way of using two adaptive filters for achieving better performance, using the cooperative learning approach. For this, we employ one lms based adaptive filter that uses a larger step size and thus has a faster rate of convergence at the expense of higher EMSE. The other filter employed uses a modified version of the lms algorithm, which employs a much lesser step size, but has one extra update term in the weight update relation that helps in learning from the faster filter its filter weight information. The learning takes place during the transient phase, while, in the steady state, two filters become almost independent of each other. Presence of the learning component in the weight update recursion enables the filter to converge much faster while a smaller step size ensures much less steady state EMSE. The claims are supported by theoretical as well as detailed simulation studies.
This paper presents a comparative study of beamforming techniques using least mean square (lms) algorithm and its variants, like, normalized least mean square (Nlms) algorithm and sign least mean square (Slms) algorit...
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ISBN:
(纸本)9781479975495
This paper presents a comparative study of beamforming techniques using least mean square (lms) algorithm and its variants, like, normalized least mean square (Nlms) algorithm and sign least mean square (Slms) algorithm. The accuracy of beam generation toward the direction of arrival (DoA) and null generation toward the interferer, depends on the value of step size parameter used in the algorithm. Beamwidth and side lobe levels (SLL) are also compared for these three algorithms.
This paper presents Echo Cancellers based on the Wavelet-lms algorithm. The performance of the Least Mean Square algorithm in Wavelet transform domain is observed and its application in Echo cancellation is analyzed. ...
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ISBN:
(纸本)0819445584
This paper presents Echo Cancellers based on the Wavelet-lms algorithm. The performance of the Least Mean Square algorithm in Wavelet transform domain is observed and its application in Echo cancellation is analyzed. The Widrow-Hoff Least Mean Square algorithm is most widely used algorithm for Adaptive filters that function as Echo Cancellers. The present day communication signals are widely non-stationary in nature and some errors crop up when Least Mean Square algorithm is used for the Echo Cancellers handling such signals. The analysis of non-stationary signals often involves a compromise between how well transitions or discontinuities can be located. The multi-scale or multi-resolution of signal analysis, which is the essence of wavelet transform, makes Wavelets popular in non-stationary signal analysis. In this paper, we present a Wavelet-lms algorithm wherein the wavelet coefficients of a signal are modified adaptively using the Least Mean Square algorithm and then reconstructed to give an Echo-free signal. The Echo Canceller based on this algorithm is found to have a better convergence and a comparatively lesser MSE (Mean Square error).
With the advancement in the field of design of adaptive filter it is expected that the convergence will improve correctness in the estimates of output. Adaptive filter system focus on integrity in the existing lms alg...
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ISBN:
(纸本)9781509006656
With the advancement in the field of design of adaptive filter it is expected that the convergence will improve correctness in the estimates of output. Adaptive filter system focus on integrity in the existing lms algorithm including FIR filter, with the computation using different data formats. Adaptive filter system support various applications with the objective to provide stable system performance. Current adaptive filter system needs in depth investigation in the computing domain to enhance the quality of the estimates. Hence interval arithmetic domain is used to increase the precision of computation. Proposed solution to this research task in this aspect is examined for a sine signal.
This paper proposes the least mean square (lms) algorithm based versatile, vector, and fault tolerant adaptive finite impulse response (FIR) filter designs. Here, the M-taps versatile design is to perform the filter o...
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ISBN:
(纸本)9789813297661;9789813297678
This paper proposes the least mean square (lms) algorithm based versatile, vector, and fault tolerant adaptive finite impulse response (FIR) filter designs. Here, the M-taps versatile design is to perform the filter operation with the number of filter co-efficients varied from 2 to M. The M-taps vector design is to perform left perpendicular left perpendicular M/L right perpendicular numbers of L-taps filter operations in parallel, where M >= L. The fault tolerantM-taps filter is to perform the (M - N)-taps fault free filter operation under the N numbers of faulty filter kernels, where (M - N) >= 2. All the existing and proposed designs are implemented with 45 nm CMOS technology. The proposed 16-taps vector adaptive filter design achieves 93% of improvement in throughput as compared with the distributed arithmetic based design.
lms algorithm based on the S-function has a small amount of calculation, faster convergence rate and good tracking properties for time-varying systems. But when the signal's error is small, the step factor changes...
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ISBN:
(纸本)9781510600294
lms algorithm based on the S-function has a small amount of calculation, faster convergence rate and good tracking properties for time-varying systems. But when the signal's error is small, the step factor changes too fast, system identification is not quick enough and the controllable variables are few. To solve the shortcomings, an improved S-function algorithm has been proposed. Simulation results show that the convergence rate of the algorithm is superior to other improved algorithms, and the tracking property for the time-varying system is better than the improved normalized lms algorithms. The algorithm proposed in this paper not only overcomes the discrepancy between the signal's error and step factor, but also makes the algorithm more flexible by introducing a new controllable variable.
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