EMGdi signal enhancement based on ICA decomposition and wavelet transform

Diaphragmatic electromyogram (EMGdi) signal plays an important role in the diagnosis and analysis of respiratory diseases. However, EMGdi recordings are often contaminated by electrocardiographic (ECG) interference, which posing serious obstacle to traditional denoising approaches due to overlapped spectra of these signals. In this paper, a novel method based on wavelet transform and independent component analysis (ICA) is proposed to remove the ECG interference from noisy EMGdi signals. With the proposed method, the original independent components of contaminated EMGdi signal were first obtained with ICA. Then the ECG components contained were removed by a specially designed wavelet domain filter. After that, the purified independent components were reconstructed back to the original signal space by ICA to obtain clean EMGdi signals. Experimental results achieved on practical clinical data show that the proposed approach is better than several traditional methods include wavelet transform (WT), ICA, digital filter and adaptive filter in ECG interference removing.     

从肌电信号中减少工频干扰方法的性能比较分析

如何从肌电信号中有效地减少工频干扰一直是肌电信号检测与应用中的突出问题。本文总结数字陷波、LMS自适应滤波、卡尔曼(Kalman)滤波和S变换等几种适合进行实时工频干扰去除的方法,研究和分析它们在去除肌电信号中工频干扰的性能。初步结果表明:Kalman滤波方法在从肌电信号中减少工频干扰方面表现出了较好的整体性能,而S变换方法对具有严重工频干扰的肌电信号具有较好的噪声抑制效果。     

Superior foetal electrocardiogram signal elicitation using a novel artificial intelligent Bayesian methodology

Innumerable casualties due to intrauterine hypoxia are a major worry during prenatal phase besides advanced patient monitoring with latest science and technology. Hence, the analysis of foetal electrocardiogram (fECG) signals is very vital in order to evaluate the foetal heart status for timely recognition of cardiac abnormalities. Regrettably, the latest technology in the cutting edge field of biomedical signal processing does not seem to yield the desired quality of fECG signals required by physicians, which is the major cause for the pathetic condition. The focus of this work is to extort non-invasive fECG signal with highest possible quality with a motive to support physicians in utilizing the methodology for the latest intrapartum monitoring technique called STAN (ST analysis) for forecasting intrapartum foetal hypoxia. However, the critical quandary is that the non-invasive fECG signals recorded from the maternal abdomen are affected by several interferences like power line interference, baseline drift interference, electrode motion interference, muscle movement interference and the maternal electrocardiogram (mECG) being the dominant interference. A novel hybrid methodology called BANFIS (Bayesian adaptive neuro fuzzy inference system) is proposed. The BANFIS includes a Bayesian filter and an adaptive neuro fuzzy filter for mECG elimination and non-linear artefacts removal to yield high quality fECG signal. Kalman filtering frame work has been utilized to estimate the nonlinear transformed mECG component in the abdominal electrocardiogram (aECG). The adaptive neuro fuzzy filter is employed to discover the nonlinearity of the nonlinear transformed version of mECG and to align the estimated mECG signal with the maternal component in the aECG signal for annulment. The outcomes of the investigation by the proposed BANFIS system proved valuable for STAN system for efficient prediction of foetal hypoxia.     

基于子带分解的DFRFT自适应滤波语音增强算法

提出一种改进的语音增强方法,利用子带分解对带噪语音信号进行处理,再在离散分数傅里叶变换(DFRFT)域采用最小均方(LMs)自适应算法进行滤波,对滤波后的子带信号进行DFRFT逆变换,最后利用综合滤波器组合成增强后的语音信号。仿真结果表明,本算法明显提高了收敛速度,减少了计算时间。在主客观评价中均具有较好的语音增强效果。     

CW interference mitigation in GNSS receiver based on frequency-locked loop

Global navigation satellite system (GNSS) receivers are highly susceptible to continuous wave (CW) interference because the received signals are extremely weak. Current interference mitigation techniques mainly use a notch filter or transform domain. This paper proposes a computationally effective algorithm based on a frequency-locked loop (FLL) to mitigate interference in GNSS receivers. The performance of the algorithm is validated through an analysis of the characteristics of the interference reduction filter and interference estimation precision. A Monte Carlo simulation is used to compare the proposed algorithm with various previous algorithms: the adaptive IIR notch filter, adaptive linear-phase FIR filter, and N-sigma DFT algorithm. The simulation results show that the proposed algorithm exhibits excellent interference estimation precision and superior antijamming performance compared with the conventional algorithms.     

An enhanced empirical wavelet transform for noisy and non-stationary signal processing

As an alternative method of empirical mode decomposition (EMD), the empirical Wavelet transform (EWT) method was proposed to realize the signal decomposition by constructing an adaptive filter bank. Though the EWT method has been demonstrated its effectiveness in some applications, it becomes invalid in analyzing some noisy and non-stationary signals due to its improper segmentation in the frequency domain. In this paper, an enhanced empirical wavelet transform method is proposed. This method takes advantage of the waveform in the frequency domain of a signal to eliminate drawbacks of the EWT method in the spectrum segmentation. It modifies the segmentation algorithm by adopting the envelope approach based on the order statistics filter (OSF) and applying criteria to pick out useful peaks. With these measures, the proposed method obtains a perfect segmentation in decomposing noisy and non-stationary signals. Furthermore, simulated and experimental signals are used to verify the effectiveness of the proposed method.     

心电信号去噪的数学形态学滤波器

为探索验证一种基于数学形态滤波器的去除心电基线漂移和工频干扰的高性能滤波器设计方法,借鉴数学形态学一维信号滤波原理,提出自适应阈值ECG去噪算法的思路,讨论了3σ统计准则在ECG自适应阈值滤波中的作用,利用改进的算法对心电图中常见的工频干扰和基线漂移进行校正。通过对MIT-BIH心率变异数据库中多组数据的仿真验证研究,验证了该算法能有效实现心电信号的噪声预处理;数学形态学理论在心电信号处理中具有良好性能,是实时处理一维生物医学信号有潜力的工具。     

ICP－AES多色仪系统Kalman滤波软件包设计与应用研究

在ＩＣＰ－ＡＥＳ多色仪系统上实现Ｋａｌｍａｎ滤波。所建立的软件包具有直接读取扫描数据文件、导数光谱、自适应滤波、结果存储与输出等功能。实际样品分析结果表明，该软件可替代或补充原有数据采集与处理系统，在回收率、检测限等方面较常规的在峰法与离峰法有较大程度改进，特别是对于具有复杂光谱背景的痕量元素分析，优点更为明显。     

基于修正快速傅里叶变换算法的仪表着陆系统信号干扰抑制与定量分析

针对日益繁忙的机场空域和周边建筑对仪表着陆系统信号干扰的增加,以及传统模拟处理技术缺陷,提出一种基于修正快速傅里叶变换(FFT)频谱校正和最小均方差(LMS)算法自适应滤波器结合实现仪表着陆系统(ILS)信号鉴频分离的技术方案。该方案应用LMS自适应滤波器对ILS信号干扰进行抑制,完成滤波器的权值系数设定,并通过修正FFT技术在时域与频域内对ILS信号进行频谱分离与提取,修正其频谱幅值,以消除由于采样引起的频谱泄露和栅栏效应对信号的影响,使得提取信号逼近理想情况,提高调制度差(DDM)识别精度。针对ILS信号的干扰抑制与频域分离进行了仿真验证,结果表明所提出的信号处理系统技术方案能够有效地对干扰进行抑制,完成信号频域识别,为飞机着陆进近阶段提供准确可靠的导航信息。     

自适应提升小波变换与信号去噪

文章引入了基于提升法的自适应离散小波变换,根据LMS自适应法使伯恩斯坦预测算子自适应匹配特定的数据序列,而且应用该方法于信号的软域值去噪,数值仿真实验表明自适应提升小波变换同经典的小波变换相比,去噪后信号的信噪比效率相近,提升方法的优点在于其设计上的灵活性和计算简单。     

时变单音干扰下GPS系统的性能分析及干扰抑制方法

针对实际系统中干扰源的移动特性,将干扰信道建模为时变瑞利衰落信道,分析了时变单音干扰下多普勒频移对GPS系统性能的影响并推导了误码率公式,提出了一种基于短时傅里叶的自适应时频域干扰抑制算法。该算法首先对接收到的GPS信号进行时频分析,检测出受到干扰的频点,然后设置合理的干扰剔除门限,最后通过自适应二阶格型陷波器将所受干扰的频点剔除。并与基于lms的时域自适应滤波算法进行了对比,实验证明,所提算法对时变单音干扰具有较好的干扰抑制能力。     

基于改进粒子群算法的自适应陷波器设计

在信号处理中,接收信号常伴随着干扰和噪声,这就需要最优滤波器来实现,其中工频干扰的消除则以自适应陷波器为最优。利用粒子群算法自适应地调节其权值,得到与干扰信号接近的期望信号,最终达到消除干扰得到有用信号的目的。同时,针对此算法存在局部收敛和收敛速度不高的问题,提出了改进方法。计算机仿真结果表明了该改进粒子群算法在自适应陷波器设计上的有效性,并取得了较高的效率。     

Nonlinear and adaptive undecimated hierarchical multiresolution analysis for real valued discrete time signals via empirical mode decomposition approach

Hierarchical multiresolution analysis is an important tool for the analysis of signals. Since this multiresolution representation provides a pyramid like framework for representing signals, it can extract signal information effectively via levels by levels. On the other hand, a signal can be nonlinearly and adaptively represented as a sum of intrinsic mode functions (IMFs) via the empirical mode decomposition (EMD) algorithm. Nevertheless, as the IMFs are obtained only when the EMD algorithm converges, no further iterative sifting process will be performed directly when the EMD algorithm is applied to an IMF. As a result, the same IMF will be resulted and further level decompositions of the IMFs cannot be obtained directly by the EMD algorithm. In other words, the hierarchical multiresolution analysis cannot be performed via the EMD algorithm directly. This paper is to address this issue by performing a nonlinear and adaptive hierarchical multiresolution analysis based on the EMD algorithm via a frequency domain approach. In the beginning, an IMF is expressed in the frequency domain by applying discrete Fourier transform (DFT) to it. Next, zeros are inserted to the DFT sequence and a conjugate symmetric zero padded DFT sequence is obtained. Then, inverse discrete Fourier transform (IDFT) is applied to the zero padded DFT sequence and a new signal expressed in the time domain is obtained. Actually, the next level IMFs can be obtained by applying the EMD algorithm to this signal. However, the lengths of these next level IMFs are increased. To reduce these lengths, first DFT is applied to each next level IMF. Second, the DFT coefficients of each next level IMF at the positions where the zeros are inserted before are removed. Finally, by applying IDFT to the shorten DFT sequence of each next level IMF, the final set of next level IMFs are obtained. It is shown in this paper that the original IMF can be perfectly reconstructed. Moreover, computer numerical simulation results show that our proposed method can reach a component with less number of levels of decomposition compared to that of the conventional linear and nonadaptive wavelets and filter bank approaches. Also, as no filter is involved in our proposed method, there is no spectral leakage in various levels of decomposition introduced by our proposed method. Whereas there could be some significant leakage components in the various levels of decomposition introduced by the wavelets and filter bank approaches.     

颤振试验中加速度计信号的时频滤波方法研究

针对飞机颤振试飞试验数据信噪比偏低的问题,提出了两种加速度计信号的时频域滤波算法,分别借助小波和分数阶傅里叶变换对数据进行时频分析,并在时频域内滤波.该类算法的共同思想是利用扫频信号在时频域内的聚焦特性,有效提取真实响应信号,达到了信噪分离的目的.文中给出了具体的滤波算法,并通过仿真算例和实际试飞数据检验了滤波效果.结果表明两种方法均可显著提高加速度计信号的信噪比.其中,小波变换方法的通用性较好,而分数阶傅里叶变换在处理线性扫频激励数据时表现了更优的去噪效果.     

Distortion estimates for adaptive lifting transforms with noise

Multimedia analysis, enhancement and coding methods often resort to adaptive transforms that exploit local characteristics of the input source. Following the signal decomposition stage, the produced transform coefficients and the adaptive transform parameters can be subject to quantization and/or data corruption (e.g. due to transmission or storage limitations). As a result, mismatches between the analysis- and synthesis-side transform coefficients and adaptive parameters may occur, severely impacting the reconstructed signal and therefore affecting the quality of the subsequent analysis, processing and display task. Hence, a thorough understanding of the quality degradation ensuing from such mismatches is essential for multimedia applications that rely on adaptive signal decompositions. This paper focuses on lifting-based adaptive transforms that represent a broad class of adaptive decompositions. By viewing the mismatches in the transform coefficients and the adaptive parameters as perturbations in the synthesis system, we derive analytic expressions for the expected reconstruction distortion. Our theoretical results are experimentally assessed using 1D adaptive decompositions and motion-adaptive temporal decompositions of video signals.     

Adaptive GLR-, Rao- and Wald-based CFAR detectors for a subspace signal embedded in structured Gaussian interference

The problem of detecting a subspace signal embedded in subspace Gaussian interference and thermal noise is studied in this paper. In this problem, both the signal-independent and signal-dependent interferences are assumed to be present, therefore the overall interference subspace covers the signal subspace. The approach of this paper extends previous works involving either of those two kinds of interferences. A set of secondary data containing only interference plus noise is employed to estimate the interference covariance matrix and the noise power. Three new detectors are designed via the generalized likelihood ratio (GLR), Rao and Wald tests, respectively. Their probabilities of false alarms (PFAs) and detections are analytically derived. The PFAs show that the new detectors have the constant false alarm rate (CFAR) property against the interference and noise. Numerical results show that the new detectors outperform their counterparts for the studied problem. Furthermore, the new detectors are less sensitive to the secondary data size and to the mismatched subspace signal than some other detectors, such as the GLR detector (GLRD), the adaptive matched filter (AMF), the adaptive subspace detector (ASD), etc.     

An adaptive NR filter for spectrum analysis

An adaptive non-recursive (NR) filter with application to spectrum analysis is presented. The adaptive nature is implemented with a weight adjustment algorithm (LMS) on the filter that performs a steepest descent minimization of mean square error. The difference between an “interference” or coherent representative of it and the raw signal under analysis via its spectral content is defined as the error. A priori knowledge of the disturbances consists of the spectral content of the disturbances without regard to amplitude or phase information.The algorithm is manifest with a digital arithmetic unit using high speed logic in a robust 512 weight digital filter fabricated with current integrated circuit technology. The experimental configuration is described along with improvements currently attainable with advancements in technology. Applications to spectrum analysis, whereby unknown disturbances degrade interpretation of spectrum signatures in bearing studies, acoustic phenomena, and sonar signal analysis, is described.Adaptive filter behavior with stationary as well as non-stationary input signals are presented. It is shown that for some narrowband as well as broadband disturbances the filter tracks and adaptively rejects disturbance in real-time so as to present uncluttered renditions of the spectrum signatures or spectrograms for analysis.     

基于傅立叶变换的语音频谱平滑算法

传统的基于源-滤波器模型的语音频谱平滑算法,需要将语音信号分解为激励源和对应的声道滤波器,这样做会引入误差,最终导致合成语音质量的下降。研究了一种基于傅立叶变换的语音频谱平滑算法,避免了上述的分解步骤。实验表明,这种方法能够较好地进行语音频谱平滑并且使合成语音的质量没有受到太大影响。     

Stability analysis of periodic orbit of an adaptive notch filter for frequency estimation of a periodic signal

Online frequency estimation of a sinusoidal signal is a classical problem and has many practical applications. Recently an adaptive notch filter (ANF) with global convergence property has been developed for frequency estimation of a pure sinusoidal signal. This paper addresses a modified ANF structure that can estimate the fundamental frequency of any periodic signal including pure sinusoidal signals. To prove the stability of the modified ANF, the paper introduces a new theorem that shows for any periodic signal, there exists a locally asymptotically stable periodic orbit of this ANF by which the frequency estimation becomes feasible. This alternative stability proof is simple and uses widely known mathematical tools, and therefore alleviates the problem complexity even when the input signal is a pure sinusoidal signal. A further contribution of this paper is obtaining a necessary and sufficient condition in terms of design parameters for local asymptotical stability of the modified ANF. This condition, obtained from the numerical study of Floquet multipliers of a linear time-varying periodic system, provides a strict stability region in the modified ANF design parameters space.     

基于谱减和LMS的自适应语音增强

针对宽带噪声背景下的语音增强问题,将短时语音视为非平稳或宽平稳信号,基于谱减法和自适应滤波的最小均方（LMS）算法,提出了一种FIR型自适应滤波算法（SSLMS）：用减谱法由短时噪声观测语音估计期望信号,作为滤波器输出信号的参考信号;用滤波器的输出与参考信号的差值为误差信号,用LMS算法求得滤波器权系数修正量,并修正滤波器。权系数最速下降调整中,采用了归一化LMS、符号LMS、块LMS技术,以简化保证权系数收敛的步长选择、减少权系数修正的运算量,从而提高自适应速度。对不同的语音在各种信噪比下仿真实验,并与改进的谱减法比较,结果表明,该法增强效果优于谱减法;在信噪比为3 dB时该法的增强效果仍然令人满意。