A correlation test-based validity monitoring procedure for online detecting the quality of nonlinear adaptive noise cancellation

In the present study, a new correlation test-based nonlinear adaptive noise cancellation (ANC) validity monitoring procedure is proposed by following the insight and formulations which were developed by the authors for validating identified nonlinear dynamic models. The new method is based on the concept that if an ANC is valid, the recovered signal should be uncorrelated to the noise source. Then, a new correlation test between recovered signal and noise source is periodically computed to online check the validity of noise cancellers when ANCs are in operation. Simulation demonstrations on validity monitoring for recursive least squares-based ANC are conducted to illustrate the effectiveness and efficiency of the new procedure.     

基于小波旁瓣相消器的轴承故障特征提取

为解决强背景噪声下声信号提取的轴承故障特征不显著问题,提出一种基于小波旁瓣相消器的故障特征提取方法。该方法利用小波滤波器组将含噪故障轴承声信号变换到小波域,进行小波域阵列广义旁瓣相消自适应波束形成,再通过小波滤波器组重构增强后的故障轴承信号,最后对重构增强后的信号进行包络解调并提取故障特征频率进行故障诊断。实验结果表明,该方法能够在强背景噪声下有效提取滚动轴承故障特征,并且相较于传统的延时求和波束形成器具有更好的降噪和故障特征增强效果。     

Improved performance of impulse active noise control using active function threshold with absolute harmonic variable step-size algorithm

The Standard (conventional) adaptive algorithms exhibits low convergence rate and minimum noise suppression, or else the system becomes unstable under Gaussian and non-Gaussian (impulsive noise SαS distributions) noise environments. In order to overcome the drawback of traditional algorithms (i.e., to eliminate unwanted noise), the popular algorithm Filtered Cross Minimum Square (FxLMS) is used in Active Noise Control (ANC), not only to improve its efficiency but also to improve its performance. In this paper, we proposed two improvements: first, we proposed a novel method Active threshold function FxLMS (ATFxLMS) being employed in ANC in the paths of primary (reference) and error signals; a second proposal is employing the Variable Step-Size based on Absolute Harmonic Mean (AHMVSS) of error signal. The idea behind this method is that the step-size of the algorithm varies depending on the harmonic mean of error signal obtained from the error location. In comparison to the fixed step-size algorithm, the proposed ATF-AHMVSS provided an improved convergence rate for the desired ANC efficiency. Moreover computational complication of the proposed method was examined as it was found that the proposed algorithm provided stable condition for ANC systems. Computer simulation results are revealed that the proposed (AT & AHMVSS-FxLMS) algorithm have attained excellent performance in terms of convergence speed, noise reduction and minimum steady state error as compared to other existing algorithms under different noise inputs. The results obtained from the proposed algorithm show outperformance compared to traditional adaptive algorithms.     

一种基于小波变换的红外导引头信号检测方法

针对某型号红外导引头信号的检测问题,提出了一种基于离散平稳小波变换的微弱脉冲信号检测方法.根据有用脉冲信号与噪声信号在频谱特性上的差异,对导引头信号进行多尺度的离散平稳小波变换,利用分解后得到的低频近似信号逼近信号中的低频噪声来滤出低频噪声的干扰,同时采用阈值去噪的方法处理信号中的白噪声.将该方法应用于仿真信号和真实导引头信号检测,仿真实验结果表明:该方法在有效克服传统离散正交小波变换去噪时容易产生的Gibbs现象的前提下,极大地提高了导引头信号的信噪比,增强了导引头的探测能力.     

On application of input data partitioning to Bayesian weighted averaging of biomedical signals

Averaging in the time domain may be used for noise attenuation in case of biomedical signals with a quasi‐cyclical character. Traditional arithmetic averaging technique assumes the constancy of the noise power cycle‐wise, however, most types of noise are not stationary and the variability of noise power is observed. It constitutes a motivation for using methods of weighted averaging, in particular Bayesian weighted averaging. This paper presents the computational study of Bayesian weighted averaging with traditional (sharp) and fuzzy partition of the input data in the presence of non‐stationary noise. There is presented the known empirical Bayesian weighted averaging method (EBWA), with the parameter p describing the probabilistic model, and its modification NBWA which eliminates the parameter. Both methods can be extended by partitioning of the input data. The performance of presented methods is experimentally evaluated for an analytical signal as well as a real ECG signal and compared with traditional arithmetic averaging method. However, the methods can be applied to any signal with a quasi‐cyclical character. The aim of the paper is to show the influence of the type of partition as well as the number of parts on the quality of the averaged signal.     

基于自适应噪声抵消的CZ单晶炉炉膛温度信号处理

为滤除CZ单晶炉炉膛温度信号在单晶生长过程中存在的低频干扰,提出一种基于噪声抵消技术的滤波方法.首先利用傅立叶级数构造出低频干扰的逼近信号,然后根据炉温信号的缓变特征建立能够获取低频干扰逼近信号的抵消器误差函数,最后利用一种改进的粒子群优化算法优化误差函数获得低频干扰的逼近信号,并用该逼近信号抵消低频干扰.实验结果表明,所提出的自适应噪声抵消滤波算法能够有效滤除CZ单晶炉炉膛温度信号中的低频干扰,并优于常用的滤波方法.     

An Improved Lung Sound De-noising Method by Wavelet Packet Transform with Pso-Based Threshold Selection

Lung abnormalities and respiratory diseases increase with the development of urban life. Lung sound analysis provides vital information of the present condition of the pulmonary. But lung sounds are easily interfered by noises in the transmission and record process, then it cannot be used for diagnosis of diseases. So the noised sound should be processed to reduce noises and to enhance the quality of signals received. On the basis of analyzing wavelet packet transform theory and the characteristics of traditional wavelet threshold de-noising method, we proposed a modified threshold selection method based on Particle Swarm Optimization (PSO) and support vector machine (SVM) to improve the quality of the signal, which has been polluted by noises. Experimental results show that the recognition accuracy of de-noised lung sounds by the improved de-noising method is 90.03%, which is much higher than by the other traditional de-noising methods. Meanwhile, the lung sound processed by the proposed method sounds better than by other methods. All results make it clear the modified threshold selection can obtain a better threshold vector and improve the quality of lung sounds.     

静息态下基于变分模态分解的生命体征检测

传统的无线生命体征监测方法在心跳和呼吸信号的分离方面容易存在谐波残留现象,针对这一情况,提出了一种基于变分模态分解（VMD）的生命信号检测方法。该方法使用毫米波段调频连续波（FMCW）雷达进行生命体征信号获取,根据心跳及呼吸的频率特征,使用VMD算法将主要信号分解为不同模态,保证了各模态之间信号频率范围互不重叠,分离出较为完整且无谐波残留的呼吸及心跳信号。实验结果表明,所提算法能够有效提取出目标的呼吸及心跳信号,且相比传统的模态分解算法具有更高的鲁棒性和稳定性,具有良好的信噪比（SNR）,提高了测量精度和距离。     

基于独立变量方法的典型舱音去噪分析研究

基于盲源信号处理的原理,提出基于ICA的盲源分离技术,对飞机驾驶舱内飞行员的语音源信号进行分离的分析方法。本文使用了OGWE法和最大信噪比法进行了盲源分离,并与Matlab的小波分析工具箱的去噪效果进行了比对,测试分析的结果验证了基于ICA的盲源分离方法可以用来有效地处理舱音信号,具有可靠性和准确性。     

Speech Separation Methodology for Hearing Aid

In the design of hearing aids (HA), the real-time speech-enhancement is done. The digital hearing aids should provide high signal-to-noise ratio, gain improvement and should eliminate feedback. In generic hearing aids the performance towards different frequencies varies and non uniform. Existing noise cancellation and speech separation methods drops the voice magnitude under the noise environment. The performance of the HA for frequency response is non uniform. Existing noise suppression methods reduce the required signal strength also. So, the performance of uniform sub band analysis is poor when hearing aid is concern. In this paper, a speech separation method using Non-negative Matrix Factorization (NMF) algorithm is proposed for wavelet decomposition. The Proposed non-uniform filter-bank was validated by parameters like band power, Signal-to-noise ratio (SNR), Mean Square Error (MSE), Signal to Noise and Distortion Ratio (SINAD), Spurious-free dynamic range (SFDR), error and time. The speech recordings before and after separation was evaluated for quality using objective speech quality measures International Telecommunication Union -Telecommunication standard ITU-T P.862.     

Active noise control in a forced-air cooling system

This paper discusses the design methodology for the active noise control of sound disturbances in a forced-air cooling system. The active sound cancellation algorithm uses the framework of output-error based optimization of a linearly parametrized filter for feedforward sound compensation to select microphone location and demonstrate the effectiveness of active noise cancellation in a small portable data projector. Successful implementation of the feedforward based active noise controller on a NEC LT170 data projector shows a 20–40 dB reduction per frequency point in the spectrum of external noise of the forced-air cooling system can be obtained over a broad frequency range from 1 to 5 kHz. A total noise reduction (unweighted) of 9.3 dB is achieved.     

一种MEMS水听器信号去噪的组合算法研究

针对MEMS水听器采集的数据"淹没"在强噪声场中的问题,提出采用LMS自适应噪声对消与Fourier变换滤波相结合的组合算法实现MEMS水听器的信噪分离。在信号频率已知的情况下,设计了一种自适应噪声对消和Fourier变换滤波组合算法的滤波器,对提取后的信号与理想信号做性能对比。仿真实验表明:该组合算法在-15 dB的强噪声场中仍有较高的分辨精度和提取效果,对搜寻类似于"黑匣子"等情况比较适宜,并将设计的滤波器用于中北汾机测试实验的信噪分离中,结果验证了该算法具有良好的高效性和实用性。     

改进的DSSS电磁波随钻测量信号载频估计算法

为了解决随钻测量中电磁波信号的载频估计问题,基于直接序列扩频（ DSSS）通信原理,建立了改进的电磁波信号载频估计法,该方法采用信号自适应干扰对消技术滤除噪声信号,然后再通过带通滤波技术获得DSSS信号的频带,从而提高了信号的信噪比（ SNR）。在此基础上,建立了System View仿真模型,仿真结果表明：改进的载频估计法可以有效滤除噪声信号,精确地检测出电磁波信号的载波频率。     

滤波器组自适应语音增强方法

双麦克风噪声抵消应用中,由于交叉串的存在,传统自适应算法降噪性能受到很大的影响。为了提高双麦克风算法降噪性能,使用两级自适应滤波系统消除交叉串扰问题。为提高自适应滤波器收敛性能,采用主从结构LMS算法自适应调节步长因子。同时为了适合窄带处理算法,将输入信号进行子带分析预处理,对每个子带独立进行抗交叉串绕自适应处理,将各子带增强信号合并得到增强语音信号。实验结果表明,该方消噪量大,语音损伤小,语音增强效果显著。     

FPGA implementation of ICA algorithm for blind signal separation and adaptive noise canceling

An field programmable gate array (FPGA) implementation of independent component analysis (ICA) algorithm is reported for blind signal separation (BSS) and adaptive noise canceling (ANC) in real time. In order to provide enormous computing power for ICA-based algorithms with multipath reverberation, a special digital processor is designed and implemented in FPGA. The chip design fully utilizes modular concept and several chips may be put together for complex applications with a large number of noise sources. Experimental results with a fabricated test board are reported for ANC only, BSS only, and simultaneous ANC/BSS, which demonstrates successful speech enhancement in real environments in real time.     

Digital Hearing Aid with the Lapped Transform

This paper presents frequency domain techniques based upon new lapped transforms for utilization in a digital hearing aid. The lapped transform has the ability to substantially reduce the blocking effects inherent in traditional frequency domain filtering. Incorporated into the digital hearing aid are the functions of frequency shaping, acoustic feedback cancellation, and periodic noise reduction. The frequency domain algorithm allows these three functions to be easily integrated into a complete and efficient system.     

多载波CDMA子载波频率盲估计算法及其性能分析*

针对频谱感知和多载波CDMA信号解调的实际应用,根据多载波CDMA信号的循环平稳特性,提出了一种利用高阶循环累积量估计多载波CDMA信号子载波频率的方法。由于高阶循环累积量可以有效地抑制平稳噪声和非平稳高斯噪声,通过理论分析可以证明在上述噪声背景下,子载波采用BPSK调制的多载波CDMA信号的四阶循环累积量仅在循环频率为子载波频率处存在,可以通过检测此循环频率来实现子载波的估计。考虑到多载波CDMA信号发端可以采用不同的窗函数以降低频谱泄露,以常见的几种窗函数为例进行了算法仿真,发现本算法对窗函数的变化不     

A Nonlinear Method for Stochastic Spectrum Estimation in the Modeling of Musical Sounds

We propose an original technique for separating the spectrum of the noisy component from that of the sinusoidal, quasi-deterministic one, for the sinusoids + transients + noise modeling of musical sounds. It also enables estimation of the time-domain noise envelope and detection of transients with standard techniques. The algorithm for spectrum separation relies on nonlinear transformations of the amplitude spectrum of the sampled signal obtained via fast Fourier transform, which allow to eliminate the dominant partials without the need for precisely tuned notch filters. The envelope estimation is performed by calculating the energy of the signal in the frequency domain, over a sliding time window. Several transformations (such as pitch shifting, time stretching, etc.) can be performed on the so-obtained stochastic spectrum prior to resynthesis. The synthesized sound is built via inverse fast Fourier transform with overlap-add method. The performance of the proposed algorithm is assessed on synthetic, instrumental, and natural sounds in terms of different quality measures     

基于机器学习的肺音分类技术的研究进展

肺音(Lung Sound) 信号是人体呼吸系统与外界在换气过程中产生的一种生理声信号,其因含有大量的生理和病理信息而具有很高的研究价值。近年来,频发的雾霾天气等环境问题所带来的呼吸道疾病发病率的提高,也使得对肺部疾病诊断的快速性与准确性的需求大幅提升。肺部听诊以其迅捷便利和无创等优良特性重新引发人们的广泛关注,而自动肺音诊断技术的发展无疑会对肺部疾病诊断带来重要的帮助。电子听诊器以及其他信号采集技术等硬件方面的发展进一步促进了现代肺音信号的分析和识别技术的研究与进步。主要介绍了肺音的概念、基于计算机的肺音信号处理和模式识别技术,并对近年来基于机器学习的肺音分类技术的发展状况进行了总结与列举;最后,对肺音分类技术的研究和应用发展趋势进行了展望。     

A Direct Link Minimal Resource Allocation Network for Adaptive Noise Cancellation

This paper presents a modification to the Minimal Resource Allocation Network (MRAN) of Yingwei et al. by introducing direct links from inputs to output and investigates its performance for noise cancellation problems. MRAN has the same structure as a Radial Basis Function network but uses a sequential learning algorithm that adds and prunes hidden neurons as input data is received sequentially so as to produce a parsimonious network. Earlier work by Sun Yonghong et al. has demonstrated the capability of MRAN to produce a compact network with excellent noise reduction properties. In this paper the capability of the direct link Minimal Resource Allocation Network (DMRAN) is evaluated by comparing it with MRAN on several nonlinear adaptive noise cancellation problems. The direct link MRAN uses the same learning algorithm as MRAN but with the introduction of direct links we are able to realise even smaller networks than MRAN with better noise reduction properties.