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Dive into the research topics where Zhengyou He is active.

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Featured researches published by Zhengyou He.


IEEE Transactions on Audio, Speech, and Language Processing | 2014

Memory proportionate APA with individual activation factors for acoustic echo cancellation

Haiquan Zhao; Yi Yu; Shibin Gao; Xiangping Zeng; Zhengyou He

An individual-activation-factor memory proportionate affine projection algorithm (IAF-MPAPA) is proposed for sparse system identification in acoustic echo cancellation (AEC) scenarios. By utilizing an individual activation factor for each adaptive filter coefficient instead of a global activation factor, as in the standard proportionate affine projection algorithm (PAPA), the adaptation energy over the coefficients of the proposed IAF-MPAPA can achieve a better distribution, which leads to an improvement of the convergence performance. Moreover, benefiting from the memory characteristics of the proportionate coefficients, its computational complexity is less than the PAPA and improved PAPA (IPAPA). In the context of AEC and stereophonic AEC (SAEC) for highly sparse impulse responses, simulation results indicate that the proposed IAF-MPAPA outperforms the PAPA, IPAPA, and memory IPAPA (MIPAPA) in terms of the convergence rate and tracking capability when the unknown impulse response suddenly changes.


IEEE Transactions on Industrial Electronics | 2014

Identification of Nonlinear Dynamic System Using a Novel Recurrent Wavelet Neural Network Based on the Pipelined Architecture

Haiquan Zhao; Shibin Gao; Zhengyou He; Xiangping Zeng; Weidong Jin; Tianrui Li

This paper presents a novel modular recurrent neural network based on the pipelined architecture (PRWNN) to reduce the computational complexity and improve the performance of the recurrent wavelet neural network (RWNN). The PRWNN inherits the modular architectures of the pipelined recurrent neural network proposed by Haykin and Li and is made up of a number of RWNN modules that are interconnected in a chained form. Since those modules of the PRWNN can be simultaneously performed in a pipelined parallelism fashion, this would lead to a crucial improvement of computational efficiency. Furthermore, owing to the cascade interconnection of dynamic modules, the performance of the PRWNN can be further enhanced. An adaptive gradient algorithm based on the real-time recurrent learning is derived to suit for the modular PRWNN. Simulation examples are given to evaluate the effectiveness of the PRWNN model on the identification of nonlinear dynamic systems and analysis of sunspot number time series. According to simulation results, it is clearly shown that the PRWNN provides impressive better performance in comparison with the single RWNN model.


IEEE Transactions on Neural Networks | 2011

Low-Complexity Nonlinear Adaptive Filter Based on a Pipelined Bilinear Recurrent Neural Network

Haiquan Zhao; Xiangping Zeng; Zhengyou He

To reduce the computational complexity of the bilinear recurrent neural network (BLRNN), a novel low-complexity nonlinear adaptive filter with a pipelined bilinear recurrent neural network (PBLRNN) is presented in this paper. The PBLRNN, inheriting the modular architectures of the pipelined RNN proposed by Haykin and Li, comprises a number of BLRNN modules that are cascaded in a chained form. Each module is implemented by a small-scale BLRNN with internal dynamics. Since those modules of the PBLRNN can be performed simultaneously in a pipelined parallelism fashion, it would result in a significant improvement of computational efficiency. Moreover, due to nesting module, the performance of the PBLRNN can be further improved. To suit for the modular architectures, a modified adaptive amplitude real-time recurrent learning algorithm is derived on the gradient descent approach. Extensive simulations are carried out to evaluate the performance of the PBLRNN on nonlinear system identification, nonlinear channel equalization, and chaotic time series prediction. Experimental results show that the PBLRNN provides considerably better performance compared to the single BLRNN and RNN models.


IEEE Transactions on Audio, Speech, and Language Processing | 2012

Adaptive Extended Pipelined Second-Order Volterra Filter for Nonlinear Active Noise Controller

Haiquan Zhao; Xiangping Zeng; Xiaoqiang Zhang; Zhengyou He; Tianrui Li; Weidong Jin

This correspondence presents an extended pipelined second-order Volterra (EPSOV) filter for active control of nonlinear noise processes. The corresponding nonlinear filtered-x algorithms using the filter bank implementation are also suggested. Compared to the standard SOV filter using the filtered-x least mean square (SOVFXLMS), those modules of the EPSOV filter can be performed simultaneously in a pipelined parallelism fashion, and this would lead to a significant improvement in its total computational efficiency. Results obtained from computer simulations for nonlinear noise processes demonstrate that the proposed method outperforms the SOV.


Signal Processing | 2016

A robust band-dependent variable step size NSAF algorithm against impulsive noises

Yi Yu; Haiquan Zhao; Zhengyou He; Badong Chen

Proposed is a new subband adaptive filter (SAF) algorithm by minimizing Hubers cost function that is robust to impulsive noises. Generally, this algorithm works in the mode of the normalized SAF (NSAF) algorithm, while it behaves like the sign SAF (SSAF) algorithm only when the impulsive noises appear. To further improve the robustness of this algorithm against impulsive noises, the subband cutoff parameters are updated in a recursive way. Moreover, the proposed algorithm can be interpreted as a variable step size NSAF algorithm, thus it exhibits faster convergence rate and lower steady-state error than the NSAF. Simulation results, using different colored input signals in both impulsive and free-impulsive noise environments, show that the proposed algorithm works better than some existing algorithms.


Signal Processing | 2014

A new normalized LMAT algorithm and its performance analysis

Haiquan Zhao; Yi Yu; Shibin Gao; Xiangping Zeng; Zhengyou He

Abstract As one of adaptive filtering algorithms based on the high order error power (HOEP) criterion, the least mean absolute third (LMAT) algorithm outperforms the least mean square (LMS) algorithm in terms of the convergence performance. However, the choice range of its step-size is dependent on the power of the input signal. To overcome this shortcoming, a new normalized LMAT (NLMAT) algorithm is presented in this paper. The proposed algorithm has a good anti-jamming capability against the impulsive noise via assigning a upper-bound to the square of the feedback error in the weight update rule. Moreover, the range of the step-size is derived in detail to guarantee the stability of the proposed algorithm in the mean and mean-square senses. Furthermore, the performance of the proposed algorithm is analyzed in terms of the steady-state mean square deviation (MSD) and mean square error (MSE) as well as computational complexity. Simulation results in the context of system identifications illustrate that the proposed algorithm performs much better than the existing algorithms in various noise environments, with a fast convergence rate, low steady-state error and good tacking capability.


Iet Communications | 2012

Complex-valued pipelined decision feedback recurrent neural network for non-linear channel equalisation

Haiquan Zhao; Xiangping Zeng; Zhengyou He; Weidong Jin; Tianrui Li

A novel complex-valued non-linear equaliser-based pipelined decision feedback recurrent neural network (CPDFRNN) is proposed in this study for non-linear channel equalisation in wireless communication systems. The CPDFRNN with low computational complexity, a modular structure comprising a number of modules that are interconnected in a chained form, is an extension of the recently proposed real-valued pipelined decision feedback recurrent neural equalisers. Each module is implemented by a small-scale complex-valued decision feedback recurrent neural network (CDFRNN). Moreover, a decision feedback part in each module can overcome the unstable characteristic of the complex-valued recurrent neural network (CRNN). To suit the modularity of the CPDFRNN, an adaptive amplitude complex-valued real-time recurrent learning (CRTRL) algorithm is presented. Simulations demonstrate that the CPDFRNN equaliser using the amplitude CRTRL algorithm with less computational complexity not only eliminates the adverse effects of the nesting architecture, but also provides a superior performance over the CRNN and CDFRNN equalisers for non-linear channels in wireless communication systems.


Applied Soft Computing | 2016

Improved functional link artificial neural network via convex combination for nonlinear active noise control

Haiquan Zhao; Xiangping Zeng; Zhengyou He; Shujian Yu; Badong Chen

The combination scheme based on adaptive FLANN filter is designed for nonlinear ANC systems.The CNFSLMS algorithm in the filter bank form is derived to obtain an improved convergence behavior.To reduce the computational complexity, the modified CNFSLMS algorithm is proposed by replacing the sigmoid function with a Versorial function.The analysis of the computational complexity is discussed.The steady state performance is analyzed. A method relying on the convex combination of two normalized filtered-s least mean square algorithms (CNFSLMS) is presented for nonlinear active noise control (ANC) systems with a linear secondary path (LSP) and nonlinear secondary path (NSP) in this paper. The proposed CNFSLMS algorithm-based functional link artificial neural network (FLANN) filter, aiming to overcome the compromise between convergence speed and steady state mean square error of the NFSLMS algorithm, offers both fast convergence rate and low steady state error. Furthermore, by replacing the sigmoid function with the modified Versorial function, the modified CNFSLMS (MCNFSLMS) algorithm with low computational complexity is also presented. Experimental results illustrate that the combination scheme can behave as well as the best component and even better. Moreover, the MCNFSLMS algorithm requires less computational complexity than the CNFSLMS while keeping the same filtering performance.


Digital Signal Processing | 2015

Adaptive recursive algorithm with logarithmic transformation for nonlinear system identification in α-stable noise

Haiquan Zhao; Lu Lu; Zhengyou He; Badong Chen

Although the least mean pth power (LMP) and normalized LMP (NLMP) algorithms of adaptive Volterra filters outperform the conventional least mean square (LMS) algorithm in the presence of α-stable noise, they still exhibit slow convergence and high steady-state kernel error in nonlinear system identification. To overcome these limitations, an enhanced recursive least mean pth power algorithm with logarithmic transformation (RLogLMP) is proposed in this paper. The proposed algorithm is adjusted to minimize the new cost function with the p-norm logarithmic transformation of the error signal. The logarithmic transformation, which can diminish the significance of outliers under α-stable noise environment, increases the robustness of the proposed algorithm and reduces the steady-state kernel error. Moreover, the proposed method improves the convergence rate by the enhanced recursive scheme. Finally, simulation results demonstrate that the proposed algorithm is superior to the LMP, NLMP, normalized least mean absolute deviation (NLMAD), recursive least squares (RLS) and nonlinear iteratively reweighted least squares (NIRLS) algorithms in terms of convergence rate and steady-state kernel error. An enhanced recursive least mean pth power algorithm with logarithmic transformation is proposed.The new cost function with the p-norm logarithmic transformation of the error signal is presented.The convexity of the proposed cost function is demonstrated.The convergence performance of the proposed RLogLMP algorithm is analyzed.


Circuits Systems and Signal Processing | 2016

Two Improved Normalized Subband Adaptive Filter Algorithms with Good Robustness Against Impulsive Interferences

Yi Yu; Haiquan Zhao; Badong Chen; Zhengyou He

To improve the robustness of subband adaptive filter (SAF) against impulsive interferences, we propose two modified SAF algorithms with an individual scale function for each subband, which are derived by maximizing correntropy-based cost function and minimizing logarithm-based cost function, respectively, called MCC-SAF and LC-SAF. Whenever the impulsive interference happens, the subband scale functions can sharply drop the step size, which eliminate the influence of outliers on the tap-weight vector update. Therefore, the proposed algorithms are robust against impulsive interferences and exhibit the faster convergence rate and better tracking capability than the sign SAF (SSAF) algorithm. Besides, in impulse-free interference environments, the proposed algorithms achieve similar convergence performance as the normalized SAF (NSAF) algorithm. Simulation results have demonstrated the performance of our proposed algorithms.

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Haiquan Zhao

Southwest Jiaotong University

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Xiangping Zeng

Southwest Jiaotong University

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Tianrui Li

Southwest Jiaotong University

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Badong Chen

Xi'an Jiaotong University

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Weidong Jin

Southwest Jiaotong University

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Yi Yu

Southwest Jiaotong University

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Shibin Gao

Southwest Jiaotong University

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Lu Lu

Southwest Jiaotong University

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Xiaoqiang Zhang

Southwest Jiaotong University

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