ADPTIVE INTERFERENCE CANCELER BASED ON FOURTH ORDER CUMULANT

An adaptive interference cancler algorithm based on fourth-order cumulant is proposed to overcome the fault of conventional autocorrelation-based Least Mean Square (LMS)algorithm which cannot suppress the correlated Gaussian noises. The recursive formulation of the proposed algorithm is given to reduce its computational complexity. The simulation results are presented to demonstrate the effectiveness of this novel approach.     

An acoustic echo canceler

The authors describe an experimental 1000-tap single-chip adaptive AEC (acoustic echo canceler) occupying 28 mm² of die area in 3-μm, double-metal, p-well, CMOS technology. A floating-point format and power-of-two multiplications are chosen to simplify the hardware. To exploit pipelining and parallelism, interleaved data storage and multibank memory sharing the same addresses are designed. Hardware minimization is considered from both the system and the architecture perspective. In a loudspeaker telephone application, 27 dB of echo reduction is achieved after 1 s of convergence time     

Multiuser soft interference canceler via iterative decoding for DSLapplications

This paper presents a method to mitigate interference on a transmission system by another system with an overlapping frequency band. The systems are uncoordinated so that they cannot be made to transmit in orthogonal coding space. The transmission environment is an interference channel rather than the multiaccess channel often considered in other multiuser detection problems, for instance at the base station of a code division multiple access wireless system. Using the fact that the inputs are all from discrete constellations, it is possible to use a multiuser maximum-likelihood (ML) detector to decode the signals from the system of interest with low probability of error. However, the ML detector often requires impractical computation. A soft linear canceler is instead used to reduce the interference with a small loss of bandwidth in the system of interest. The method has a much lower complexity than a full-blown ML detector, and the soft canceler performance is close to an ML detector. The method is demonstrated on a very-high-speed digital subscriber lines system with a home local area network interference. The system of interest is using a multicarrier modulation transmission scheme such as discrete multitone, while the interfering system transmits using quadrature amplitude modulation     

Adaptive interference canceler for narrowband and widebandinterferences using higher order statistics

A new higher order statistics-based adaptive interference canceler is introduced to mitigate narrowband and wideband interferences in environments where the interference is non-Gaussian and a reference signal, which is highly correlated with the interference, is available. The new scheme uses higher order statistics (HOS) of the primary and reference inputs and employs a gradient-type algorithm for updating the adaptive filter coefficients. The update equation of the HOS-based adaptive filter is independent of uncorrelated Gaussian noises and can mitigate the interference more effectively than adaptive filters based on second-order statistics. The performance of the. HOS-based adaptive filter is much less sensitive to the choice of the step size parameter than the adaptive filters based on the LMS algorithm. It is demonstrated, by means of extensive simulations, that the HOS-based filter can mitigate both narrowband and wideband interferences effectively. Comparisons with adaptive filters based on the LMS algorithm and second-order statistics are also presented in the paper     

Adaptive harmonic jammer canceler

A new adaptive harmonic jammer canceler is proposed. It is based on the use of two sensors that enable an adaptive generation of a reference signal that is uncorrelated with the desired signal. This reference signal is used for the reconstruction of the desired signal by an adaptive subtraction method. This canceler is well suited to radio communications. A theoretical analysis of the convergence of the coupled algorithms is presented with the help of the associated ordinary differential equation introduced by L. Ljung. Numerical simulations illustrate the different proposed algorithms     

Markov-based eigenanalysis method for frequency estimation

This paper proposes an eigenanalysis-based method for estimating the frequencies of complex-valued sine waves. The basic idea behind this method consists of using a set of linearly independent vectors that are orthogonal to the signal subspace spanned by the principal eigenvectors of the data covariance matrix. Exploiting that orthogonality condition gives an overdetermined system of linear equations, the unknown parameters of which are uniquely related to the frequencies. Analytical expressions are derived for the covariances of the equation errors in the sample version of the aforementioned linear system of equations. Based on these expressions a Markov-like estimate of the unknown parameters is introduced, which asymptotically (with respect to either the number of data samples or the signal-to-noise ratio) provides the minimum variance frequency estimates in a fairly large class of consistent estimators. The paper includes Monte-Carlo simulations that support the theoretical analysis results and show that those results may apply to scenarios with rather low values of the number of data samples and the signal-to-noise ratio     

A multirate acoustic echo canceler structure

A new subband echo canceler (SBEC) structure is proposed to reduce the transmission delay introduced by conventional SBEC structures, without distorting the near-end signal. The proposed structure is based on computing two output errors, one for using during single-talk and the other one for using during double-talk periods. With the SBEC structure we propose a double-talk detector with a subband configuration which allows a fast and accurate detection of double-talk periods, enabling the SBEC algorithm to track changes in the echo path impulse response when the near-end signal is absent. Computer simulations using actual speech signals, and subjective evaluation tests are given to show the convergence performance, tracking and double-talk detection ability, of the proposed scheme     

Performance analysis of adaptive eigenanalysis algorithms

We present a rigorous analysis of several popular forms of short memory adaptive eigenanalysis algorithms using a stochastic averaging method. A first-order analysis shows that the algorithms do not have any equilibrium points despite published claims to the contrary. Through averaging analysis, we show that they hover around an appropriate eigenvector. A second-order analysis is also given without the Gaussian noise assumption, and our results greatly outperform an earlier approximation in the literature. The second-order analysis has been of much interest in the offline study but, in the dynamic adaptive case, is uncommon     

Compressing image-based relighting data using eigenanalysis and wavelets

In image-based relighting (IBL) a tremendous number of reference images are needed to synthesise a high-quality novel image. This collection of reference images is referred as an IBL data set. An effective compression method for IBL data makes the IBL technique more practical. Within an IBL data set, there is a strong correlation among different reference images. In conventional eigen-based image compression methods, the principal component analysis (PCA) process is used for exploiting the correlation within a single image. Such an approach is not suitable for handling IBL data. The authors present an eigenimage-based method for compressing IBL data. The method exploits the correlation among reference images. Since there is a huge number of images and pixel values, the cascade recursive least square (CRLS) network based PCA is used to extract eigenimages. Afterwards, the wavelet approach is used for compressing those eigenimages. Simulation results demonstrate that this approach is much superior to that of compressing each reference image with JPEG and JPEG2000.     

Echo cancellation in DMT-receivers: circulant decomposition canceler

Asymmetric digital subscriber lines (ADSLs) employ discrete multitone modulation (DMT) as transmission format, where subcarriers are assigned to the up- and/or downstream transmission direction. To separate up- and downstream signals, the ADSL standard allows the use of echo cancellation resulting in improved bit rates, reach, and/or noise margins. In DMT-based modems, typically, the mixed time/frequency (MTF) domain echo canceling scheme, as proposed by Ho et al., is implemented. This technique estimates the echo filter in the frequency domain using the least mean square (LMS) algorithm with the transmitted echo symbols as update directions. Since not every tone of the transmitted echo signal will carry data, i.e., will be excited, the MTF adaptation process does not lead to a good estimate for the echo channel, unless extra power on unused echo tones is transmitted. However, transmitting extra power on such tones is often undesired. In this paper, we present an alternative echo canceling scheme referred to as the circulant decomposition canceler (CDC), which works without extra power requirements and with comparable complexity as the method of Ho et al. Similar to MTF echo canceling, the CDC scheme can easily be incorporated into a multirate environment with different transmit and receive rates and can also cheaply be combined with per-tone equalization and double talk cancellation to allow fast tracking and/or convergence in the presence of a far-end signal.     

Improved tracking adaptive noise canceler for nonstationaryenvironments

The performance of multiple reference adaptive noise cancelers is investigated and a new filter structure is proposed that provides better tracking in the multipath, multisource, nonstationary automobile noise environment studied. The filter uses the least mean square (LMS) algorithm with multiple filtering stages and subbanded sections to improve the overall tracking performance while maintaining filter stability     

A new IIR adaptive echo canceler: GIVE

A novel IIR adaptive gradient instrumental variable echo canceler (GIVE) is presented. Its features include adaptive controllability during double-talk periods in acoustic conference systems; guarantee of global convergence; low computational cost (the same order as the IIR LMS algorithm of the equation error method); and flexible structures (parallel or series-parallel structures). We also show a convergence analysis for gradient adaptive algorithms including GIVE. Based on this analysis, the optimum stepsize for GIVE and three suboptimum algorithms are proposed to accelerate convergence and reduce misadjustment. In addition, a simple method that guarantees the stability of IIR filters and a configuration of GIVE applicable to closed loop systems are presented. These proposals are extensively studied by computer simulations     

A stable and distortion-free IIR echo and howling canceler

The authors propose a configuration for an infinite impulse response (IIR) adaptive echo and howling canceller, which consists of a two-channel maximum entropy lattice filter and its inverse filter. The echo is canceled by the adaptive lattice filter, while the signal distortion is eliminated by the inverse lattice. With stability guaranteed without the necessity of testing, the structure costs O (N) multiplications per sampling period. The algorithm can also be greatly simplified for white input cases     

Analysis of the performance and sensitivity of an eigenspace-basedinterference canceler

Eigenspace-based interference cancelers (EICs) possess the advantages of providing maximal suppression of interference with fast convergence over conventional adaptive beamformers. However, the performance and sensitivity to steering angle error of EICs have not been analyzed due to the use of a signal blocking matrix. We first present a technique to construct a positive definite matrix based on the signal blocking matrix and then use this matrix to compensate the effect of the signal blocking matrix on the sensor noise received by an EIC. Therefore, the interference subspace required for finding the optimal weight vector can be obtained using conventional eigenvalue decomposition (EVD). Moreover, the performance and sensitivity to the steering angle error of the EIC can be analyzed. Simulation examples are provided for confirming the theoretical results     

MIMO干扰信道下的协作干扰对齐优化算法

干扰对齐是将来自其他发射机的干扰信号对准到同一信号子空间,从而减少对期望信号的干扰的一种传输技术,近年来得到了广泛的关注.本文为K用户多输入多输出(MIMO)干扰信道系统提出了一种协作干扰对齐优化算法.通过对信道的奇异值分解,基于矩阵弦距离,不仅选出一组最好的特征子信道来传输信号,而且还选取一组有利于有用信号远离干扰信号的预编码矩阵,这样,可以最大化减少干扰对有用信号的影响.另外,本文还根据得出的预编码矩阵来计算和优化干扰抑制矩阵.本文算法在没有用迭代的基础上来计算预编码矩阵和干扰抑制矩阵,这样不仅使整个系统的复杂度大大降低,进一步,仿真结果显示容量和能量效率较其他算法有明显改进,这也正是最近受到广泛瞩目的绿色通信所要求的.     

Adaptive multichannel lattice-escalator filter structure: anapplication to generalized sidelobe canceler

An adaptive digital filter structure which can be used in multichannel noise-canceling applications is described. The proposed structure is obtained by creating a hybrid from the lattice and the escalator (Gram-Schmidt) structures, with the coefficients being updated using the least mean square (LMS) algorithm. As an application, the proposed adaptive multichannel digital filter is applied to implement an adaptive generalized sidelobe canceler     

An adaptive array for interference rejection

An adaptive array that rejects undesired or interfering signals is presented. The array pattern is controlled by an adaptive feedback system based on a steepest descent minimization of mean-square error. Error is defined as the difference between the array output and a locally generated reference signal. Minimization of mean-square error is closely related to maximization of signal-to-noise ratio (SNR). A two-element adaptive array has been built, and its experimental performance is discussed. Typical patterns for various desired and interfering signals are shown, as well as measured transient response. Finally, some experiments showing the array behavior with modulated signals are described. The results show that such an antenna system is capable of automatically rejecting interfering signals, subject only to certain basic constraints. No a priori information about the angles of arrival of the signals is required, Detailed knowledge of the waveforms of the desired and interfering signals is also not needed, although the spectral characteristics of the desired signal must be known.     

集总干扰的一种分析方法

0前言集总干扰这个说法并不时髦。顾名思义,集总干扰就是干扰集合起来,对另外一个系统产生更大的干扰。单个干扰源往往不足以对接收机产生干扰,但是     

Pre-DFT Processing using eigenanalysis for coded OFDM with multiple receive antennas

In broadband wireless communications, coded orthogonal frequency-division multiplexing (OFDM) can be used with multiple receive antennas to achieve both frequency diversity and space diversity. In this scenario, the optimal approach is subcarrier-based space combining. However, such an approach is quite complex, because multiple discrete Fourier transform (DFT) blocks, each per receive antenna, are used. We propose a pre-DFT processing scheme based upon eigenanalysis. In the proposed scheme, the received signals are weighted and combined both before and after the DFT processing. As a result, the required number of DFT blocks can be significantly reduced. With perfect weighting coefficients, the margin of the performance improvement decreases along with the increase of the number of DFT blocks, thus enabling effective performance and complexity tradeoff. To achieve a maximum average pairwise codeword distance, it will be shown that the maximum number of DFT blocks required is equal to the minimum of the number of receive antennas and the number of distinct paths in the channel. When the number of distinct paths is larger than the number of receive antennas and with a smaller number of DFT blocks, extensive simulation results will also show that near-optimal performance can still be achieved for most channels. Finally, in an OFDM system with differential modulation, we use a signal covariance matrix to obtain the weighting coefficients before the DFT processing. In this case, simulation results will demonstrate that the performance of the proposed scheme can be better than subcarrier-based space combining, but with much lower complexity.