非均匀噪声背景下混合信号DOA估计算法

针对非均匀噪声背景下非相关信源与相干信源并存时波达方向(DOA)估计问题，提出了基于迭代最小二乘和空间差分平滑的混合信号DOA估计算法。首先，该算法利用迭代最小二乘方法得到噪声协方差矩阵估计，然后对数据协方差矩阵进行“去噪”处理，利用子空间旋转不变技术实现非相关信源DOA估计；其次，基于空间差分法消除非相关信号并构造新矩阵进行前后向空间平滑，利用求根MUSIC算法估计相干信源DOA。相比于传统算法，该算法能估计更多的信源数，在低信噪比情况下DOA估计性能更优越。仿真实验结果验证了该算法的有效性。      

An adaptive Kalman filter for ECG signal enhancement

The ongoing trend of ECG monitoring techniques to become more ambulatory and less obtrusive generally comes at the expense of decreased signal quality. To enhance this quality, consecutive ECG complexes can be averaged triggered on the heartbeat, exploiting the quasi-periodicity of the ECG. However, this averaging constitutes a tradeoff between improvement of the SNR and loss of clinically relevant physiological signal dynamics. Using a bayesian framework, in this paper, a sequential averaging filter is developed that, in essence, adaptively varies the number of complexes included in the averaging based on the characteristics of the ECG signal. The filter has the form of an adaptive Kalman filter. The adaptive estimation of the process and measurement noise covariances is performed by maximizing the bayesian evidence function of the sequential ECG estimation and by exploiting the spatial correlation between several simultaneously recorded ECG signals, respectively. The noise covariance estimates thus obtained render the filter capable of ascribing more weight to newly arriving data when these data contain morphological variability, and of reducing this weight in cases of no morphological variability. The filter is evaluated by applying it to a variety of ECG signals. To gauge the relevance of the adaptive noise-covariance estimation, the performance of the filter is compared to that of a Kalman filter with fixed, (a posteriori) optimized noise covariance. This comparison demonstrates that, without using a priori knowledge on signal characteristics, the filter with adaptive noise estimation performs similar to the filter with optimized fixed noise covariance, favoring the adaptive filter in cases where no a priori information is available or where signal characteristics are expected to fluctuate.     

NAR estimators of spatial covariance matrices for adaptive arraydetection

The theory of noise-alone-reference (NAR) power estimation is extended to the estimation of spatial covariance matrices. A NAR covariance estimator insensitive to signal presence is derived. The SNR (signal-to-noise ratio) loss incurred by using this estimator is independent of the input SNR and is less than that encountered with the maximum likelihood covariance estimator given that the same number of uncorrelated snapshots is available to both estimators. The analysis assumes first a deterministic signal. The results are extended and generalized to signals with unknown parameters or random signals. For the random signal case, generalized and quasi-NAR covariance estimators are presented     

Multiexpert automatic speech recognition using acoustic and myoelectric signals

Classification accuracy of conventional automatic speech recognition (ASR) systems can decrease dramatically under acoustically noisy conditions. To improve classification accuracy and increase system robustness a multiexpert ASR system is implemented. In this system, acoustic speech information is supplemented with information from facial myoelectric signals (MES). A new method of combining experts, known as the plausibility method, is employed to combine an acoustic ASR expert and a MES ASR expert. The plausibility method of combining multiple experts, which is based on the mathematical framework of evidence theory, is compared to the Borda count and score-based methods of combination. Acoustic and facial MES data were collected from 5 subjects, using a 10-word vocabulary across an 18-dB range of acoustic noise. As expected the performance of an acoustic expert decreases with increasing acoustic noise; classification accuracies of the acoustic ASR expert are as low as 11.5%. The effect of noise is significantly reduced with the addition of the MES ASR expert. Classification accuracies remain above 78.8% across the 18-dB range of acoustic noise, when the plausibility method is used to combine the opinions of multiple experts. In addition, the plausibility method produced classification accuracies higher than any individual expert at all noise levels, as well as the highest classification accuracies, except at the 9-dB noise level. Using the Borda count and score-based multiexpert systems, classification accuracies are improved relative to the acoustic ASR expert but are as low as 51.5% and 59.5%, respectively.     

Fetal Electrocardiogram Enhancement by Time-Sequenced Adaptive Filtering

An adaptive method for performing optimal time-varying filtering of nonstationary signals having a recurring statistical character, e.g., recurring pulses in noise, has been proposed. This method, called time-sequenced adaptive filtering, is applied to the enhancement of abdominally derived fetal electrocardiograms against background muscle noise. It is shown that substantial improvement in terms of signal distortion is obtained when time-sequenced filtering, rather than conventional time-invariant filtering, is employed. The method requires two or more abdominal channels containing correlated signal components, but uncorrelated muscle noise components. The location of the fetal pulses in time must be estimated in order to synchronize the filter's time-varying impulse response to the fetal cardiac cycle.     

Superefficiency in blind source separation

Blind source separation is the problem of extracting independent signals from their mixtures without knowing the mixing coefficients nor the probability distributions of source signals and may be applied to EEG and MEG imaging of the brain. It is already known that certain algorithms work well for the extraction of independent components. The present paper is concerned with superefficiency of these based on the statistical and dynamical analysis. In a statistical estimation using t examples, the covariance of any two extracted independent signals converges to 0 of the order of 1/t. On-line dynamics shows that the covariance is of the order of η when the learning rate η is fixed to a small constant. In contrast with the above general properties, a surprising superefficiency holds in blind source separation under certain conditions where superefficiency implies that covariance decreases in the order of 1/t² or of η² . The paper uses the natural gradient learning algorithm and method of estimating functions to obtain superefficient procedures for both batch estimation and on-line learning. A standardized estimating function is introduced to this end. Superefficiency does not imply that the error variances of the extracted signals decrease in the order of 1/t² or η² but implies that their covariances (and independencies) do     

Limitations and Difficulties in Signal Processing by Means of the Principal-Components Analysis

Principal-components analysis is sometimes used in analysis of biophysical signals. Because the expansion coefficients obtained in this analysis have the property that they are uncorrelated, several authors made the extrapolation that the eigenfunctions of the covariance matrix, which also result from the principal- components analysis, correspond to independent physiological generators. By means of a theoretical example it is proved that this interpretation is unjustified because the eigenvectors are necessarily orthogonal due to the symmetry of the covariance matrix. Signals resulting from independent physiological generators do not necessarily have this property.     

Adaptive reduction of heart sounds from lung sounds usingfourth-order statistics

When recording lung sounds, an incessant noise source occurs due to heart sounds. This noise source severely contaminates the breath sound signal and interferes in the analysis of lung sounds. In this paper, an adaptive heart-noise reduction method, based on fourth-order statistics (FOS) of the recorded signal, without requiring recorded “noise-only” reference signal, is presented. This algorithm uses adaptive filtering to preserve the entire spectrum. Furthermore, the proposed filter is independent of Gaussian uncorrelated noise and insensitive to the step-size parameter. It converges fast with small excess errors and, due to the narrowband nature of heart noise (HN), it requires a very small number of taps. Results from experiments with healthy subjects indicate a local HN reduction equal to or greater than 90%     

Maximum likelihood array processing in spatially correlated noisefields using parameterized signals

This paper deals with the problem of estimating signal parameters using an array of sensors. This problem is of interest in a variety of applications, such as radar and sonar source localization. A vast number of estimation techniques have been proposed in the literature during the past two decades. Most of these can deliver consistent estimates only if the covariance matrix of the background noise is known. In many applications, the aforementioned assumption is unrealistic. Recently, a number of contributions have addressed the problem of signal parameter estimation in unknown noise environments based on various assumptions on the noise. Herein, a different approach is taken. We assume instead that the signals are partially known. The received signals are modeled as linear combinations of certain known basis functions. The exact maximum likelihood (ML) estimator for the problem at hand is derived, as well as computationally more attractive approximation. The Cramer-Rao lower bound (CRB) on the estimation error variance is also derived and found to coincide with the CRB, assuming an arbitrary deterministic model and known noise covariance     

Active bearing estimation from a mobile two-dimensional array of sensors

A new approach for active-sonar target detection and bearing estimation from a mobile two-dimensional array of sensors operating in a predominantly noisy environment is presented. Sensor-level adaptive noise cancellation featuring an unconventional method for reference-noise estimation is the key preprocessing step in the proposed approach. A signal-subspace algorithm resulting from two-stage optimisation based on a generalised eigendecomposition of the signal plus (residual) noise covariance matrix is employed to estimate the bearing of the detected target. Simulation results conclusively demonstrate that the proposed scheme is capable of performing target detection and the subsequent two-dimensional bearing estimation with a high degree of reliability at signal-to-noise power ratios as low as -70 and -40-dB, respectively.     

An experimental SMI adaptive antenna array simulator for weakinterfering signals

An experimental sample matrix inversion (SMI) adaptive antenna array for suppressing weak interfering signals is described. The experimental adaptive array uses a modified SMI algorithm to increase the interference suppression. In the modified SMI algorithm, the sample covariance matrix is redefined to reduce the effect of thermal noise on the weights of an adaptive array. This is accomplished by subtracting a fraction of the smallest eigenvalue of the original covariance matrix from its diagonal entries. The test results obtained using the experimental system are compared with theoretical results. The two show a good agreement     

A functional GaAs FET noise model

It is the purpose of this paper to develop a theory upon which the design of low noise FET amplifiers can be based. This is not a fundamenta model of the noise mechanisms in GaAs FET's, but rather, an endeavor to relate physically measurable device capacitances and resistances to the device noise figure and optimum noise source impedance. I will be shown that the noise performance of an FET can be adequately described by two uncorrelated noise sources. One, at the input of the FET, is the thermal noise generated in the various resis, tances in the gate-source loop. This noise source is frequency dependent and it can be calculated from the equivalent circuit of the FET. The second noise source, in the Output of the FET, is frequency independent, and not recognizably related to any measured parameters. This output nise is a function of drain current and voltage. The decomposition of the FET noise into two uncorrelated sources simplifies the design of broad-band low noise amplifiers. Once the equivalent circuit of a device and its noise figure at one frequency are known, the optimum noise source impedance and noise figure over a broad range of frequencies may be calculated. For the device designer this model also may be helpful in balancing input-output noise tradeoffs.     

一种信源个数与波达方向联合估计的新算法

针对多级维纳滤波器(MSWF)用于子空间估计时信号特征矢量泄漏到噪声子空间的问题,提出了一种新的信号子空间估计算法,该算法不需要训练信号和信源个数的先验知识.随后,给出了一种信源个数的后判断方法,最终完成信源个数及方向的同时估计.整个算法不需要协方差矩阵的计算和特征值的分解,具有较低的计算复杂度.在均匀线阵且信号互不相关情况下,改进后的算法用于波达方向估计时拥有与基于特征分解方法近似的性能.仿真结果验证了该方法的有效性.     

Information theoretic criteria for the determination of the numberof signals in spatially correlated noise

The problem of determining the number of signals in high-resolution array processing when the noise is spatially correlated (having an unknown covariance matrix) is examined. By considering a model in which two sensor arrays are well separated such that their noise outputs are uncorrelated, the authors develop a likelihood function whose maximum can be expressed in a very simple form involving the canonical correlation coefficients. This likelihood function and a choice of penalty functions constitute a number of new information theoretic criteria suitable for the determination of the number of signals in an unknown correlated noise environment. Furthermore, it is demonstrated that the new criteria are applicable in the case when only one sensor array is available     

SMI adaptive antenna arrays for weak interfering signals

The performance of adaptive antenna arrays in the presence of weak interfering signals (below thermal noise) is studied. It is shown that conventional adaptive antenna arrays sample matrix inversion (SMI) algorithm is used are unable to suppress such interfering signals. To overcome this problem, the SMI algorithm is modified. In the modified algorithm, the covariance matrix is redefined such that the effect of thermal noise on the weights of adaptive arrays is reduced. Thus, the weights are dictated by relatively weak signals. It is shown that the modified algorithm provides the desired interference protection.     

一种新颖的DOA估计算法

子空间类波达方向(Direction Of Arrival, DOA)估计算法的关键在于得到高质量的信号子空间估计。该文利用矩阵伪逆的双正交性,针对源信号不相关而其本身是色信号的情况,给出了一种新颖的DOA估计算法,它不需要知道噪声统计特性。该算法利用一组空时相关矩阵的结构化信息,能稳健而精确地估计出信号子空间,从而得到DOA的精确估计。仿真实验证实了所给算法的有效性。     

Spectral self-coherence restoral: a new approach to blind adaptivesignal extraction using antenna arrays

A new approach to blind adaptive signal extraction using narrowband antenna arrays is presented. The approach has the capability to extract communication signals from cochannel interference environments using only known spectral correlation properties of those signals, i.e. without using knowledge of the content or direction of arrival of the transmitted signal, or the array manifold or background noise covariance of the receiver, to train the antenna array. The class of spectral self-coherence restoral (SCORE) objective functions is introduced, and algorithms for adapting antenna arrays to optimize these objective functions are developed. Using the theory of spectral correlation, it is shown by analysis and simulation that these algorithms maximize the signal-to-interference-and-noise ratio at the output of the narrowband antenna array when a single communication signal with spectral self-coherence at a known value of frequency separation, along with an arbitrary number of interferers without spectral self-coherence at that frequency separation, are impinging on the array     

Covariance analysis of time delay estimates for strained signals

Estimates of time-varying delays from the ultrasonic echo signals of compressed biological media are the basis of a new type of medical imaging known as elastography. This paper is focused on predicting the covariance between time delays estimated from sequential Gaussian-weighted echo segments that overlap. The accuracy of the analysis was tested and improved by comparisons with measurements involving ultrasonic waveforms simulated from independent band-limited Gaussian signal and noise spectra. Data were generated to explore the dependence of time delay covariance on ultrasonic signal-to-noise ratio, time-bandwidth product W, fractional bandwidth, window separation, and the amount of strain (a is the time rate of change in delay). The relationship between a and other experimental parameters was crucial for understanding how signal decorrelation affects time delay error and, ultimately, elastographic noise. For echo waveforms without strain (a=0), delay variance was found to decrease with W. However, when waveform segments were strained (a≠0) a minimum was found in the plot of time delay variance versus W, where delay errors from additive noise equaled those from signal decorrelation caused by strain. Delay covariance decreased monotonically with increasing window separation (less overlap) when a=0. When a≠0, however, the covariance became negative for large separations. Properties of strain image noise were predicted from knowledge of the experimental parameters and time delay errors     

水声通信系统中基于迭代自适应的脉冲噪声抑制方法

针对水声通信系统中脉冲噪声抑制问题,提出了一种迭代自适应的脉冲噪声抑制方法。基于 OFDM子载波之间的正交性,该方法首先利用空子载波矩阵从接收信号中提取出背景噪声和脉冲噪声。然后,利用空子载波矩阵构造导频矩阵,得到脉冲噪声的干扰协方差矩阵,并在加权最小二乘准则下通过对代价函数的求解得到脉冲噪声的闭式解。最后,在接收信号中减去脉冲噪声的估计值,完成对脉冲噪声的抑制。仿真结果表明,本文方法有效降低了水声通信系统的误码率,且在高信干噪比下性能提升更加明显。     

A novel approach for DOA estimation in unknown correlated noise fields

The key of the subspace-based Direction Of Arrival （DOA） estimation lies in the estimation of signal subspace with high quality. In the case of uncorrelated signals while the signals are temporally correlated, a novel approach for the estimation of DOA in unknown correlated noise fields is proposed in this paper. The approach is based on the biorthogonality between a matrix and its Moore-Penrose pseudo inverse, and made no assumption on the spatial covariance matrix of the noise. The approach exploits the structural information of a set of spatio-temporal correlation matrices, and it can give a robust and precise estimation of signal subspace, so a precise estimation of DOA is obtained. Its performances are confirmed by computer simulation results.