Estimation of single event-related potentials utilizing the Pronymethod

This paper deals with estimation of the waveform of a single event-related potential, sERP. An additive noise model is used for the measured signal and the SNR of the disturbed sERP is approximately 0 dB. The sERP is described by a series expansion where the basis functions are damped sinusoids. The fundamental basis function is estimated by the least squares Prony method, derived for colored noise. The performance of the Prony method for different forms of the power density spectrum of the noise is investigated. A white noise approximation can be used at low signal-to-noise (SNR). The basis functions change slowly but the waveform of the sERP may vary from one stimulus to another, thus the authors average a small number of correlation functions in order to increase the SNR. The method is evaluated by using measurements from four subjects and the results confirm the variability of the sERP     

基于分数阶谱相减的语音增强法

该文提出了基于分数阶谱相减的语音增强法(FSS)。该方法通过对带噪语音信号作分数阶傅里叶变换(FRFT),将得到的分数阶语噪混合谱与估计的分数阶噪声谱相减,最后利用分数阶Fourier反变换获得去噪后的语音信号。理论分析表明,所提方法存在一个最佳分数阶阶数,使得语噪混合信号能在分数阶变换域得到最好的分离,从而有效地提高了增强语音的性能。计算机仿真表明,对于混有加性白噪声的男/女声发音信号,所提方法在信噪比提高量和Itakura距离减少量两个方面都优于传统的谱相减法(SS),并且增强语音中的音乐噪声得到了明显抑制。     

低采样率高分辨率压缩功率谱估计方法的仿真研究

低采样率的宽带功率谱估计在很多领域具有应用价值.采用压缩多核采样结构得到信号的压缩测量值, 然后建立测量值相关函数与信号相关函数之间的关系, 用最小二乘法实现相关函数估计, 最后实现功率谱的估计.该压缩采样方法的等效采样率为M/N·f_s, 可在没有任何对时域或频域稀疏性的假设条件下降低采样率.仿真分析表明, 该方法的系统噪声与加性噪声性能比周期图法略有降低, 但只要系统设计合理, 对于一定信噪比的信号, 系统噪声与加性噪声基本可以忽略, 并给出了对应的理论分析.估计分辨率与周期图法相比, 等效长度相同时略有提高; 由于本文方法降低了测量值的数目, 对于一定长度的数据来说, 估计分辨率得到了极大的提高.本文方法适用于低信噪比信号的低采样率高分辨率功率谱估计.     

基于波束域相位稳定性的目标检测方法研究

低信噪比下,为解决常规波束形成等权值累加空间谱各方位谱值,导致目标方位谱值被非目标方位谱值淹没,不能实现对水下辐射噪声信号未知的目标检测问题,本文提出了一种基于波束域相位稳定性的目标检测方法.依据水下目标辐射噪声含有稳定线谱及空间谱各方位对应波束域相位稳定性差异,该方法利用波束域相位方差对各方位谱值进行加权统计,实现了对水下目标方位角的有效估计.数值仿真和实验结果表明：相比常规波束形成,该方法可以进一步增强目标方位能量,抑制非目标方位噪声干扰,改善目标检测信噪比增益.     

Multichannel Linear Prediction and Maximum-Entropy Spectral Analysis Using Least Squares Modeling

Autoregressive data modeling using the least squares linearprediction method is generalized for multichannel time series. A recursive algorithm is obtained for the formation of the system of multichannel normal equations which determine the least squares solution of the multichannel linear-prediction problem. Solution of these multichannel normal equations is accomplished by the Cholesky factorization method. The corresponding multichannel maximum-entropy spectrum derived from these least squares estimates of the autoregressive-model parameters is compared to that obtained using parameters estimated by a multichannel generalization of Burg's algorithm. Numerical experiments have shown that the multichannel spectrum obtained by the least squares method provides for more accurate frequency determination for truncated sinusoids in the presence of additive white noise.     

A new method for low-rank transform domain adaptive filtering

This paper introduces a least squares, matrix-based framework for adaptive filtering that includes normalized least mean squares (NLMS), affine projection (AP) and recursive least squares (RLS) as special cases. We then introduce a method for extracting a low-rank underdetermined solution from an overdetermined or a high-rank underdetermined least squares problem using a part of a unitary transformation. We show how to create optimal, low-rank transformations within this framework. For obtaining computationally competitive versions of our approach, we use the discrete Fourier transform (DFT). We convert the complex-valued DFT-based solution into a real solution. The most significant bottleneck in the optimal version of the algorithm lies in having to calculate the full-length transform domain error vector. We overcome this difficulty by using a statistical approach involving the transform of the signal rather than that of the error to estimate the best low-rank transform at each iteration. We also employ an innovative mixed domain approach, in which we jointly solve time and frequency domain equations. This allows us to achieve very good performance using a transform order that is lower than the length of the filter. Thus, we are able to achieve very fast convergence at low complexity. Using the acoustic echo cancellation problem, we show that our algorithm performs better than NLMS and AP and competes well with FTF-RLS for low SNR conditions. The algorithm lies in between affine projection and FTF-RLS, both in terms of its complexity and its performance     

一种单自旋回波串信号参数估计方法

针对存在加性高斯白噪声多参数变量的单自旋回波串信号参数估计问题,提出一种参数分离化的2-D参数估计方法.利用2-D数据矩阵秩为1的特性,依照迭代加权最小二乘方法,从左、右主奇异值向量中以参数分离的方式分别估计出衰减因子和频率,基于最小二乘方法进一步获得信号幅度估计.该方法在相对高信噪比和/或大数据样本下可达到克拉美罗下界,且计算复杂度较低.仿真数据结果证明了算法的有效性.     

一种空-频联合最优滤波的被动宽带检测方法

常规宽带能量检测在多目标、强干扰环境下输出信噪比(SNR)降低,检测性能大幅度下降。针对此问题,该文提出一种将子阵导向最小方差(STMV)宽带空域自适应波束形成与频域Eckart滤波结合的空-频联合最优滤波宽带检测方法。该方法首先通过子阵导向最小方差波束形成进行空间自适应处理,利用自适应波束形成的干扰抑制能力在空域实现最优滤波;然后通过最大似然估计实时估计信号和噪声的功率谱,构造Eckart滤波对自适应波束形成的输出分配不同权重进行加权滤波,从而实现频域信噪比最大化。所提方法通过空-频联合最优滤波,降低空域旁瓣干扰和频带内噪声的影响,使得输出信噪比最大,从而有效地改善目标宽带检测能力,提高被动声呐的宽带检测性能。仿真和试验数据处理结果验证了该方法的有效性。     

Linear prediction: A tutorial review

This paper gives an exposition of linear prediction in the analysis of discrete signals. The signal is modeled as a linear combination of its past values and present and past values of a hypothetical input to a system whose output is the given signal. In the frequency domain, this is equivalent to modeling the signal spectrum by a pole-zero spectrum. The major part of the paper is devoted to all-pole models. The model parameters are obtained by a least squares analysis in the time domain. Two methods result, depending on whether the signal is assumed to be stationary or nonstationary. The same results are then derived in the frequency domain. The resulting spectral matching formulation allows for the modeling of selected portions of a spectrum, for arbitrary spectral shaping in the frequency domain, and for the modeling of continuous as well as discrete spectra. This also leads to a discussion of the advantages and disadvantages of the least squares error criterion. A spectral interpretation is given to the normalized minimum prediction error. Applications of the normalized error are given, including the determination of an "optimal" number of poles. The use of linear prediction in data compression is reviewed. For purposes of transmission, particular attention is given to the quantization and encoding of the reflection (or partial correlation) coefficients. Finally, a brief introduction to pole-zero modeling is given.     

Analytic behavior of the LMS adaptive line enhancer for sinusoidscorrupted by multiplicative and additive noise

The least mean squares adaptive line enhancer (LMS ALE) has been widely used for the enhancement of coherent sinusoids in additive wideband noise. This paper studies the behavior of the LMS ALE when applied to the enhancement of sinusoids that have been corrupted by both colored multiplicative and white additive noise. The multiplicative noise decorrelates the sinusoid, spreads its power spectrum, and acts as an additional corrupting noise. Closed-form expressions are derived for the optimum (Wiener filter) ALE output SNR as a function of the residual coherent sine wave power, the noncoherent sine wave power spectrum, and the background additive white noise. When the coherent to noncoherent sine wave power ratio is sufficiently small, it is shown that a nonlinear (e.g., square law) transformation of the ALE input results in a larger optimum ALE output SNR     

Blind NLLS Carrier Frequency-Offset Estimation for QAM, PSK, and PAM Modulations: Performance at Low SNR

We address the problem of blind carrier frequency-offset (CFO) estimation in quadrature amplitude modulation, phase-shift keying, and pulse amplitude modulation communications systems. We study the performance of a standard CFO estimate, which consists of first raising the received signal to the$M$th power, where$M$is an integer depending on the type and size of the symbol constellation, and then applying the nonlinear least squares (NLLS) estimation approach. At low signal-to noise ratio (SNR), the NLLS method fails to provide an accurate CFO estimate because of the presence of outliers. In this letter, we derive an approximate closed-form expression for the outlier probability. This enables us to predict the mean-square error (MSE) on CFO estimation for all SNR values. For a given SNR, the new results also give insight into the minimum number of samples required in the CFO estimation procedure, in order to ensure that the MSE on estimation is not significantly affected by the outliers.     

Two-branch wavelet denoising for accurate spectrum sensing in cognitive radios

In order to detect the unused spectrum bands (the spectrum holes) efficiently in cognitive radios with low signal-to-noise radio (SNR), we propose to adopt two independent branches of wavelet to detect the singularities of the received signals’ power spectrum density (PSD). The sensing structure is flexible such that we can use one or two branches to cope with different SNRs. Under low SNR condition, each branch uses distinct characteristics between noise and signals in the wavelet transform to eliminate the singularities generated by the noise. By using bandpass filter to calculate PSD values of the subbands which are distinguished by the signal’s singularities, the subband with the minimum PSD value among all of the subbands could be found. Then, the results of the two branches are merged and analyzed in order to make the final decision. Finally, we use signal reconstruction to further remove the noise and then accurately detect the spectrum holes. When the SNR is high, only one branch through the denoising procedure is needed to get accurate sensing result. Our simulation results show that the two-branch wavelet method is more accurate than conventional approaches under given SNRs.     

基于STFD&SCMUSIC的多跳频信号DOA估计与网台分选

为了在欠定条件下利用跳频信号的空域特征参数进行网台分选,该文提出一种基于STFD&SCMUSIC的跳频信号DOA估计算法。首先在时频域提取跳频信号的有效跳(hop),并建立该hop的空时频矩阵(STFD);然后在MUSIC算法基础上,利用噪声子空间降维思想构造SCMUSIC空间谱;最终通过半谱搜索实现DOA快速估计,进而利用DOA信息完成信号的分选;同时为了提高低信噪比算法的性能,采用形态学滤波的方法对时频图进行修正,在修正的时频图上完成跳频信号有效hop的提取。理论分析和仿真实验表明了该算法的有效性和良好的估计性能。      

A Novel Technique for the Identification of ARMA Systems Under Very Low Levels of SNR

In this paper, a novel technique for the identification of minimum-phase autoregressive moving average (ARMA) systems from the output observations in the presence of heavy noise is presented. First, starting from the conventional correlation estimator, a simple and accurate ARMA correlation (ARMAC) model in terms of the poles of the ARMA system is presented in a unified manner for white noise and impulse-train excitations. The AR parameters of the ARMA system are then obtained from the noisy observations by developing and using a residue-based least-squares correlation-fitting optimization technique that employs the proposed ARMAC model. As for the estimation of the MA parameters, it is preceded by the application of a new technique intended to reduce the noise present in the residual signal that is obtained by filtering the noisy ARMA signal via the estimated AR parameters. A scheme is then devised whereby the task of MA parameter estimation is transformed into a problem of correlation-fitting of the inverse autocorrelation function corresponding to the noise-compensated residual signal. In order to demonstrate the effectiveness of the proposed method, extensive simulations are performed by considering synthetic ARMA systems of different orders in the presence of additive white noise and the results are compared with those of some of the existing methods. It is shown that the proposed method is capable of estimating the ARMA parameters accurately and consistently with guaranteed stability for signal-to-noise ratio (SNR) levels as low as $-{5}~{hbox {dB}}$ . Simulation results are also provided for the identification of a human vocal-tract system using natural speech signals showing a superior performance of the proposed technique in terms of the power spectral density of the synthesized speech signal.      

LTE中一种基于探测参考信号的信噪比估计算法

为了在上行链路支持频率选择性调度,长期演进(LTE)系统定义了探测参考信号(SRS)用于信道质量估计。该文主要研究SRS的信噪比估计方法,针对Boumard方法和传统DFT方法的缺点,提出一种改进的基于DFT的估计方法。该方法通过在时域修正噪声的估计区间,减小高信噪比时有用信号能量泄露对噪声估计的影响,从而获得更准确的信噪比估计。仿真结果表明,所提方法的估计性能优于Boumard方法和传统的DFT方法,提高了高信噪比时的估计精度,在高信噪比区域,平均估计性能提高了约6 dB以上。     

时域反相对称与扩频技术的混合技术研究

根据噪声在时域上的相关性以及反相对称法的抗噪原理,将反相对称法与扩频通信相结合,提出了一种新的扩频通信方法。通过仿真和实验验证表明,时域反相对称扩频技术能使系统获得较高的输出信噪比,且在相同的传输条件下,反相对称扩频技术的性能优于一般的扩频技术。     

一种基于序列相关的信噪比估计算法

针对多径信道,提出了一种基于序列相关的信噪比估计算法,利用本地序列与接收信号相关,并采用最小二乘估计法,精确地估计了接收信号幅度和噪声方差,得到了两径信道下信噪比的估计值。仿真结果表明该算法整体估计性能较好,特别适合于低信噪比条件下。在信噪比为-1dB时,与现有的频域和二阶矩四阶矩（M2M4,2-order and 4-order Moments）估计算法相比,该算法的归一化均方误差分别降低了0．09和0．2。     

基于自适应滤波的噪声抵消法

语音降噪就是从带噪语音信号中提取尽可能纯净的原始语音。文中介绍了一种基于自适应滤波的噪声抵消法，采用归一化最小均方误差算法，采集实际噪声环境下各种不同信噪比的带噪语音样本进行降噪处理，实验结果表明，处理后信号的信噪比得到了较大程度的提高，大大改善了听音效果，具有很高的可懂度，且语音自然度好，没有失真；并与谱减法进行了比较，自适应噪声抵消法的降噪幅度比谱减法有一定提高，在听音效果上，用自适应噪声抵消法处理后的语音在清晰度、自然度方面优于谱减法。     

Speech enhancement using constrained spectral amplitude subtraction based on noncausal a priori SNR

Two gain forms of spectral amplitude subtraction are derived theoretically without neglecting the correlation of speech and noise spectrum during the period of a fralne. In the implementation, the constrained gain is expressed as a function of noncausal a priori SNR （Signal-to-Noise Ratio）. Noise and noncausal a priori SNR are estimated from the multitaper spectrum of the noisy signal with algorithms modified to be suitable for the multitaper spectruln. Objective evaluations show that in case of white Gaussian noise the proposed method outperforms some methods based on LSA （Log Spectral Amplitude） in terms of MBSD （Modified Bark Spectral Distortion）, segmental SNR and overall SNR, and informal listening tests show that speech reconstructed in this way has little speech distortion and musical noise is nearly inaudible even at low SNR.     

频域内的卡尔曼滤波语音增强算法

提出了在频域内实现的卡尔曼滤波算法,该算法利用语音和噪声幅度谱的时变特性,先对语音幅度谱进行初步修正,提取较为准确的LPC系数,然后在每一频率点下对语音幅度用卡尔曼滤波进行递推估计,最终得到效果更好的增强语音.实验结果表明,本文算法有效地提高了增强语音的SNR,尤其是在高信噪比的情况下,效果更加明显.