Robust FM demodulation of discrete-time signals using least squares differential ratio

A technique for the robust demodulation of discrete FM signals in the presence of additive noise is presented. Based upon least squares, the proposed technique is shown to improve the accuracy of instantaneous frequency estimation by 17 to 79% compared with the popular discrete energy separation (DESA), smooth DESA and Hilbert transform-based algorithms, e.g. noisy signals at low SNRs.     

A joint 2D AM?CFM estimation based on higher order Teager?CKaiser energy operators

In this work, an image demodulation algorithm based on two-dimensional higher order Teager–Kaiser (TK) operators is presented. We show quantitatively and qualitatively that the introduction of higher orders in TK operator improves amplitude modulation (AM) and frequency modulation (FM) estimation results, compared to classical approaches such as the Discrete Energy Separation Algorithm (DESA) or the Analytic Signal (AS) method. Indeed, for a wide class of images, obtained demodulation errors for both the amplitude and frequency are numerically lower than the obtained ones with the DESA and AS method. The proposed method is illustrated on both synthetic and real images. Moreover, it turns out for some real images that the algorithm is very efficient in the sense that it tracks the most significant part in images and segments regions of interests, particularly, the AM counterpart. Finally, an application of our approach to the segmentation of mines’ shadows in Sonar images is presented. This is very important for both civil and military applications.     

Nonparametric BDS detector of chaotic signals against the background of white noise

This paper includes an algorithm proposed for detecting a chaotic signal, the functional block diagram for implementing the algorithm proposed, and also the ATS algorithm for generation of surrogate signals that are used in a detector for empirical estimation of the likelihood ratio.Acomparative analysis of the detection characteristics obtained by using a conventional (energy) approach and the proposed approach based on using topological properties of signals and noises.     

基于Radon-Ambiguity变换的LFM信号时/频差快速联合估计

提出了一种基于Radon-Ambiguity变换(Radon-Ambiguity Transform, RAT)的线性调频(Linear Frequency Modulated, LFM)信号时/频差快速联合估计的算法.根据LFM信号在多个不同角度上的RAT峰值位置建立一组以信号间时差和频差为未知量的方程组,求解方程组即可得到时/频差的估计值.对于存在噪声的信号,RAT误差会导致方程组不能直接求解,为了抑制噪声干扰,采用最小二乘法估计时/频差.本文算法无需计算二维平面上各点的模糊函数值,并且由于离散RAT可以通过快速傅里叶变换快速实现,具有所需运算量低的优点.仿真实验表明,相比于常见的基于模糊函数峰值搜索的时/频差估计算法,本文算法在保证时/频差估计精度的同时能够显著提高运算效率.     

基于改进互耦系数估计算法的自适应波束形成

阵列信号在实际应用中,常常会面临噪声的干扰,低信噪比(signal noise ratio, SNR)时波束形成器性能会急剧下降. 针对该问题,提出一种基于改进互耦系数估计的抗互耦算法的自适应波束形成,即研究信号在不同输入SNR以及不同干扰波达方向(direction of arrival,DOA)估计误差下的互耦系数估计误差;并通过误差分析修正原算法不同SNR下参与估计信号的数目. 改进算法改进了导向矢量和期望信号,提高了低SNR下互耦系数估计的准确度,对比分析验证了其在低SNR情况下有更强的抗干扰能力,对互耦现象有着更好的稳健性.     

Parameter Blind Estimation of Frequency-Hopping Signal Based on Time–Frequency Diagram Modification

To effectively estimate the parameters of the multiple frequency-hopping signals, a blind parameter estimation method based on time–frequency diagram modification is proposed. Firstly, the observed signal is transformed to the time–frequency domain, using short time Fourier transform with overlapping windows. Then an energy detection method based on adaptive threshold is used to modify the time–frequency diagram, and the parameters of the frequency-hopping signals are finally obtained from the modified spectrogram. Theoretical analysis and simulation results show that the method proposed can get a clear time–frequency diagram at low signal-to-noise ratio (SNR), and its accuracy of parameter estimated is higher than that of previous methods. When SNR is ?10 dB, estimation errors of frequency, hopping time and hop duration is 0.0002, 0.0008 and 0.0013, respectively, which are about 1–2 orders of magnitude lower over the previous method.

     

基于独立分量分析的PCMA信号盲分离算法

针对非协作通信中成对载波多址(Paired Carrier Multiple Acess,PCMA)信号的盲分离问题,提出了一种基于独立分量分析(Independent component analysis,ICA)的单通道盲分离算法。首先对接收到的单路PCMA信号进行参数估计得到其残余载波频率,再对其处理得到两路基带混合信号,最后利用ICA算法分离出源基带信号。该算法在未知两个卫星地面站发送信号的情况下,从接收到的PCMA信号中恢复出两路源基带信号。仿真实验表明,本文算法在信噪比为-10dB时仍具有良好的分离效果,两路基带信号的波形相似系数可分别达到0.94与0.86以上。     

基于自相关和时频分析的跳频参数估计

为解决低信噪比条件下跳频参数估计算法性能低的问题,提出了一种基于自相关和时频分析的跳频参数估计算法。首先,采用基于能量检测的分段自相关算法对接收端信号进行预处理;然后,进行时频变换,得到信号的时频矩阵,通过二值化和形态学滤波完成对信号的降噪提取;最后,通过聚类算法完成参数估计。仿真实验表明,该算法具有较高的估计精度和良好的抗噪声性能,在信噪比最低为-11 dB时估计误差数量级仍为10^-7,同时自相关运算对参数估计算法的抗噪声性能具有明显的提高作用。     

Estimation of Nonlinear Wireless Channel Based on Incremental Spline Adaptive Filtering

Channel estimation in a wireless sensor network is imperative to error-free information dissemination and data collection. The estimation procedure is challenging if there exists a nonlinear distortion to the communication signal due to the radio-frequency components in the transmitting or receiving entity. It has drawn attention to nonlinear system modeling for channel estimation, where lately, one of the most important methods has been spline adaptive filter (SAF). The necessity of updating both linear filter coefficients and nonlinear control points makes the adaptation process slow. Hence, we propose an incremental spline adaptive filter using the least mean square algorithm (ISAF-LMS), which acquires faster convergence while estimating non-linearity along with linear filter coefficients. The steady-state performance of the proposed method is carried out by following the energy conservation approach. The simulation result shows faster convergence in the distributed case than in non-cooperative estimation. Further, the performance is compared with diffusion SAF and incremental version of conventional Volterra adaptive filter-based nonlinear channel estimation (IVLMS). The proposed algorithm performance is better than IVLMS.

     

未知信源数目的DOA估计方法

针对信源数目未知情况下的DOA估计问题,该文提出了两种基于稀疏表示的DOA估计方法。一种是基于阵列协方差矩阵特征向量稀疏表示的DOA估计方法,首先证明了阵列协方差矩阵的最大特征向量是所有信号导向矢量的线性组合,然后利用阵列协方差矩阵的最大特征向量建立稀疏模型进行DOA估计;另一种是基于阵列协方差矩阵高阶幂稀疏表示的DOA估计方法,根据信号特征值大于噪声特征值的特性,通过对协方差矩阵的高阶幂逼近信号子空间,利用协方差矩阵的高阶幂的列向量建立DOA估计的稀疏模型进行DOA估计。理论分析和仿真实验验证,两种方法都不需要进行信号源数目的估计,具有较高的精度、较好的分辨力,对相干信号也具有优越的适应能力。     

Analysis of Acoustic Events in Speech Signals Using Bessel Series Expansion

In this paper, we propose an approach for the analysis and detection of acoustic events in speech signals using the Bessel series expansion. The acoustic events analyzed are the voice onset time (VOT) and the glottal closure instants (GCIs). The hypothesis is that the Bessel functions with their damped sinusoid-like basis functions are better suited for representing the speech signals than the sinusoidal basis functions used in the conventional Fourier representation. The speech signal is band-pass filtered by choosing the appropriate range of Bessel coefficients to obtain a narrow-band signal, which is decomposed further into amplitude modulated (AM) and frequency modulated (FM) components. The discrete energy separation algorithm (DESA) is used to compute the amplitude envelope (AE) of the narrow-band AM-FM signal. Events such as the consonant and vowel beginnings in an unvoiced stop consonant vowel (SCV) and the GCIs are derived by processing the AE of the signal. The proposed approach for the detection of the VOT using the Bessel expansion is shown to perform better than the conventional Fourier representation. The performance of the proposed GCI detection method using the Bessel series expansion is compared against some of the existing methods for various noise environments and signal-to-noise ratios.     

实现单通道MPSK信号盲分离的MCMC新算法

本文根据单通道接收两路MPSK混合信号在过采样下的基本模型,针对粒子滤波算法在单通道信号盲分离中的性能瓶颈以及高复杂度问题,提出了基于MCMC方法的新算法。该算法对接收信号进行过采样处理,能够利用更多的波形信息,从而有效抑制噪声的影响。新算法利用Gibbs采样估计MPSK调制符号的后验概率,近似实现了贝叶斯最优估计,并利用最小二乘法实现参数的迭代估计。理论分析与仿真实验表明,相对粒子滤波算法,本文提出的新算法在误码率性能以及复杂度方面具有良好的表现。      

基于整体最小二乘的联合信道估计及OFDM信号检测算法

针对现有的联合信道估计及OFDM信号检测算法性能不高的问题,该文提出一种基于整体最小二乘的联合信道估计及OFDM信号检测算法。该算法首先通过导频估计初始的信道信息,在此基础上不断地采用整体最小二乘进行OFDM信号检测及信道估计,有效缓解迭代模型误差的影响,加快了算法迭代的收敛速度,提高了信道估计的精度,从而降低了OFDM系统的误码率。该文推导的信道估计克拉美罗界及仿真结果均表明所提出的算法在时变信道环境下优于现有的联合信道估计及OFDM信号检测算法。     

基于高阶累积量的近场通信波达方向估计算法

通过对近场通信波达方向准确估计,提高目标信源的定位能力.传统方法中对近场源通信信源的波达方向估计采用多普勒估计方法,由于近场通信的空间信源为窄带信号,多普勒估计会导致DOA估计频谱失真.提出一种基于高阶累积量的近场通信波达方向估计算法.采用均匀间隔线列阵构建近场通信的信号模型,进行近场源目标特征构建,提取近场源通信信号的斜度和峰度等特征,采用高阶累积量特征提取方法,分别求得对应近场通信信源的方位角、频率和距离三维参数,使得每个信源的参数自动配对,提高了近场通信DOA波达方向估计的效率和精度,实现近场源通信信号的波达方向估计算法改进.仿真实验结果表明,采用该方法进行近场方法波达方向估计的精度较高,对信源方位的定位准确,性能优越于传统方法,在近场通信中具有较好的应用价值.     

DOA estimation of LFM signals based on STFT and multiple invariance ESPRIT

In order to improve the angle measurement precision of LFM signals with a low computational complexity, a direction of arrival (DOA) estimation algorithm STFT-MI-ESPRIT is proposed in this paper. The algorithm is based on short time Fourier transform (STFT) and multiple invariance estimation of signal parameters via rotational invariance techniques (MI-ESPRIT). Firstly, the STFT of the array element’s output is calculated and the signals are transformed to the time-frequency domain. Then the spatial time-frequency distribution matrix can be obtained through selecting multiple single-source time-frequency points in the time-frequency plane and the signal subspace can also be obtained using Eigen decomposition. Finally, the multiple rotational invariant equation of the array based on STFT is obtained and the closed-form solution is obtained using the multi-least-squares (MLS) criterion. The simulation results show that the proposed algorithm can improve the estimation precision greatly compared with the traditional ESPRIT-like algorithms and its computational complexity remains the same in general. This paper also proposes that the STFT-MI-ESPRIT algorithm can use partial rotational invariances of the array instead of all the rotational invariances, which can reduce the computational complexity on the basis of ensuring the estimation precision basically. The simulation results verify the effectiveness of the conclusion.     

基于Fast-ICA的CDMA信号扩频序列盲估计及性能分析

DS-CDMA分为长码和短码扩频2种扩频方式,已有的扩频序列盲估计算法只适用于短码扩频信号,不能完成长码扩频信号的扩频序列估计。对长码扩频CDMA信号的数学模型进行分析,提出了一种的新的基于Fast-ICA的多用户扩频序列盲估计方法并推导了算法的理论性能。该方法不仅能适用于长码扩频CDMA信号,而且对同步短码扩频CDMA信号也同样适用。理论分析和仿真结果验证了该方法的有效性。     

Effects of model errors on waveform estimation using the MUSICalgorithm

Sensor arrays are frequently used to separate and reconstruct superimposed signals arriving from different directions. The paper studies the effect of model errors, i.e., differences between the assumed and actual array response, on the quality of the reconstructed signals. Model errors are the limiting factor of array performance when the observation time is sufficiently long. The authors analyze a signal estimation technique which is based on the MUSIC algorithm. Formulas are derived for the signal-to-interference and signal-to-noise ratios as function of the model errors. By evaluating these formulas for selected test cases they gain some insights into the sensitivity of the signal estimation problem to model uncertainty     

The effects of array calibration errors on DF-based signal copyperformance

This paper studies the effect of array calibration errors on the performance of various direction finding (DF) based signal copy algorithms. Unlike blind copy methods, this class of algorithms requires an estimate of the directions of arrival (DOAs) of the signals in order to compute the copy weight vectors. Under the assumption that the observation time is sufficiently long, the following algorithms are studied: classical beamforming, least squares, total least squares, linearly constrained minimum variance beamforming, and structured stochastic estimation. Expressions for the mean-square error of the signal estimates are derived as a function of the calibration errors for both the case where the DOAs are known precisely and for the case where the DOAs must be estimated     

An efficient coding algorithm for the compression of ECG signals using the wavelet transform

A wavelet-based electrocardiogram (ECG) data compression algorithm is proposed in this paper. The ECG signal is first preprocessed, the discrete wavelet transform (DWT) is then applied to the preprocessed signal. Preprocessing guarantees that the magnitudes of the wavelet coefficients be less than one, and reduces the reconstruction errors near both ends of the compressed signal. The DWT coefficients are divided into three groups, each group is thresholded using a threshold based on a desired energy packing efficiency. A binary significance map is then generated by scanning the wavelet decomposition coefficients and outputting a binary one if the scanned coefficient is significant, and a binary zero if it is insignificant. Compression is achieved by 1) using a variable length code based on run length encoding to compress the significance map and 2) using direct binary representation for representing the significant coefficients. The ability of the coding algorithm to compress ECG signals is investigated, the results were obtained by compressing and decompressing the test signals. The proposed algorithm is compared with direct-based and wavelet-based compression algorithms and showed superior performance. A compression ratio of 24:1 was achieved for MIT-BIH record 117 with a percent root mean square difference as low as 1.08%.