An adaptive speech enhancement method for siren noise cancellation

Siren noises usually severely disturb the intelligibility of voice communication inside the cabs of police, paramedic and fire vehicles. It is often desired that such unwanted noise can be removed from the speech signal. In this paper, a new method is proposed to adaptively cancel siren noises and enhance speech signals. Based on the characteristics of siren noises, an anti-speech filter and a time delayer are employed in the single and dual channel noise cancellation systems to reduce the siren noises. Experiment results demonstrate that the effectiveness of the proposed method for canceling the siren noises and the performance of the enhanced speech signal is satisfying.     

Speech enhancement in dual-microphone mobile phones using Kalman filter

In this paper, a dual-microphone speech enhancement algorithm for the mobile phones is proposed. The adopted method exploits the coherence function algorithm and the Kalman filter. This algorithm has a simple implementation that does not need a prediction of interfering signals statistics. In addition, this algorithm can be used in small devices with so closely distance between the two microphones. Moreover, the use of such algorithm allows reducing multiple noise sources at many azimuths positions. Finally, the new algorithm proves its performances referring to the perceptual evaluation of speech quality and the time domain waveforms.     

非重叠背景噪声下的自适应维纳滤波模式识别方法

维纳滤波实现模式识别的关键问题是噪声知识的获得与估计。提出一种非重叠背景噪声的提取方法,首先将对噪声的粗略估计代入维纳滤波函数,得到相关峰。然后由相关峰的位置及参考图像的尺寸确定目标图像的位置和范围,从而提取出背景噪声图像。经过二次维纳滤波,得到改善的相关输出结果,实现了自适应过程。仿真结果表明在非重叠有色噪声环境下,与噪声估计法以及传统的维纳滤波方法相比,此方案具有较好的识别效果。     

Using power level difference for near field dual-microphone speech enhancement

In this contribution, a novel dual-channel speech enhancement technique is introduced. The proposed approach uses the dissimilarity between the power of received signals in the two channels as a criterion for speech enhancement and noise reduction. We claim that in near field conditions, where the distances between microphones and sound source are short, the difference in the received power levels at the two microphones is an estimate of the clean speech signal power. Then, apply this theory to present an optimum method for speech enhancement. Fortunately, the method has the ability to cope with problems such as transient noise and nearby microphones which are two of the main problems of the proposed dual-microphone speech enhancement techniques. Using objective speech quality measures and spectrogram analysis, we show that the proposed method results in improved speech quality.     

次级声源优化布放的局部空间有源降噪

针对区域有源降噪问题,为获得更优降噪效果,根据实际次级通路传递函数,提出次级声源优化布放的有源控制系统并详细比较了两种次级声源优化布放算法与次级声源均匀布放的实际降噪效果。应用的第一种次级声源优化算法是l₂范数约束的约束匹配追踪算法,第二种次级声源优化算法是l₁范数约束的稀疏正则化方法。在全消声室中利用扬声器线阵进行多通道有源降噪实验研究,实验结果表明,在200～1000 Hz,次级声源优化布放的控制系统的平均降噪量比次级声源均匀布放的控制系统的平均降噪量多5 dB左右;在1100～1900 Hz,次级声源优化布放的控制系统的平均降噪量比次级声源均匀布放的控制系统的平均降噪量多11～13 dB左右,次级声源优化布放的控制系统的降噪量分布更加均匀且次级声源输出能量更小。此外,两种优化算法中,稀疏正则化方法的降噪效果更佳。     

A noise cross PSD estimator based on improved minimum statistics method for two-microphone speech enhancement dedicated to a bilateral cochlear implant

Coherence based methods have been successfully applied to dual-microphone noise reduction systems. These techniques showed good results when noise signals on two microphones were uncorrelated, but their performance decreased with correlated noises. It could be improved when the cross power spectral density (CPSD) of received noises is available.In this paper, an improved minimum tracking (IMT) technique for noise CPSD estimation was proposed. The performance of this technique was compared to two other noise CPSD estimators based on voice activity detection (VAD) and minimum tracking (MT) approaches. Evaluation was performed at four signal-to-noise ratios (SNR) and two interfering noise source configurations.Results showed a superiority of the IMT approach in terms of low computing time and quality indicated by the perceptual evaluation of speech quality (PESQ) scores. Then, subjective listening tests were carried out with 50 normal hearing listeners using a specific bilateral cochlear implant (BCI) simulator and utilizing the French Lafon database corrupted by additional babble noise. Results obtained with the proposed technique were better than the two previously mentioned noise CPSD estimators.     

一种新的卫星钟差Kalman滤波噪声协方差估计方法

采用Kalman滤波方法进行钟差参数计算和预报时, 需确定Kalman滤波噪声协方差矩阵. 针对这一问题, 提出了一种新的卫星钟差Kalman滤波噪声协方差估计方法, 通过建立新息的相关函数序列与未知的噪声参数间的线性函数模型, 采用最小二乘法进行噪声参数估计. 采用精密钟差数据进行钟差参数估计和预报分析, 结果表明, 该方法具有较好的收敛性, 并与顾及随机噪声模型的开窗分类因子自适应抗差估计方法进行对比分析, 验证了新方法的正确性和有效性.     

The effect of Laplacian filter in adaptive unsharp masking for infrared image enhancement

Image processing, in particular image enhancement techniques have been the focal point of considerable research activity in the last decade. With the aid of an existing image enhancement technique, adaptive unsharp masking (AUM), we propose a novel kernel to be used in AUM filtering in order to enhance discontinuities which occur on the edges of targets of interest in infrared (IR) images. The proposed method uses an adaptive filter approach where an objective function is minimized by using descent algorithms. The output IR image has better sharpness and contrast adjustment for the detection of targets in terms of objective quality metrics. Hence, the proposed method ensures that the edges of the targets in IR images are sharper and that the quality of contrast adjustment has its optimum level in terms of peak signal-to-noise ratios.     

Effects of colored noise and noise delay on a calcium oscillation system

As a calcium oscillations system is in steady state, the effects of colored noise and noise delay on the system is investigated using stochastic simulation methods. The results indicate that: (1) the colored noise can induce coherence bi-resonance phenomenon. (2) there exist three peaks in the R–_τ0$R\text{–}{\tau }_0$ (R$R$ is the reciprocal coefficient of variance, and _τ0${\tau }_0$ is the self-correlation time of the colored noise) curves. For the same noise intensity Q=1$Q=1$, the Gaussian colored noise can induce calcium spikes but the white noise cannot do this. (3) the delay time can improve noise induced spikes regularity as _τ0${\tau }_0$ is small, and R$R$ has a significant minimum with increasing τ$\tau$ as _τ0${\tau }_0$ is large. (4) large values of ζ$\zeta$ reduce noise induced spikes regularity.     

An active noise reduction earplug using a piezoelectric laterally radiating loudspeaker

This paper describes the electroacoustic behaviour of a new piezoelectric Active Noise Reduction (ANR) earplug device. The sensor is a microphone having suitable characteristics chosen from among commercially available electret microphones. The actuator is a cylindrical piezoelectric ceramic loaded with a thin fluid film. An analytic model of laterally radiating loudspeaker, based on fundamental equations of acoustics, which takes into account the effect of viscosity and heat conduction is adapted to describe the acoustic radiation of the actuator. Theoretical and experimental frequency responses in a small cavity and an example of the ANR that can be obtained with the earplug are presented.     

Selection of meaningless steady noise for masking of speech

This paper focuses on masking speech with meaningless steady noise as a way of realizing a comfortable sound environment. As a basis for research, meaningless steady noise at minimum sound pressure levels for masking of male or female meaningful speech is considered, based on psychological experiments using a method of adjustment. From the results, band-limited pink noise can be selected as the most effective noise for masking of speech. In the case of speech with a lower sound pressure level, the sound pressure level of the meaningless steady noise needs to be a little higher.     

Intelligibility enhancement for noisy whispered speech using asymmetric cost function

We proposed two whispered speech enhancement methods based on asymmetric cost functions in this paper to deal with the amplification and attenuation distortions of whispered speech distinctively.The modified Itakura-Saito（MIS）distance function provides more penalties to speech amplification distortion,whereas the Kullback-Leibler（KL）divergence function gives more penalties to speech attenuation distortion.The experimental results show that the MIS function based method achieves significant improvement of intelligibility in contrast to the conventional speech enhancement algorithms when the signal-to-noise ratio（SNR）falls below-6 dB,whereas the KL function based one achieves the similar result as the minimum mean square error（MMSE）speech enhancement method.The results show that the effects of the amplification and attenuation distortions on the intelligibility of the enhanced whisper are different,where larger attenuation distortion may result in better intelligibility of speech with low SNR.However,the attenuation distortion has small effects on intelligibility of speech with high SNR.     

改进维纳滤波器及其在目标方位估计中的应用

针对复杂的水声环境以及信噪比较低的目标信号导致方位估计性能较差的问题,本文提出了一种基于改进维纳滤波器和波束形成器的方位估计方法,该方法能够抑制噪声,提高目标方位估计性能。首先利用改进维纳滤波器抑制各通道接收数据中的噪声,提高输出信噪比。在此基础上,将改进维纳滤波器的输出通过波束形成器,获得目标的方位估计。改进维纳滤波器能够通过调整滤波器参数,控制滤波器的噪声抑制能力和信号失真。因此,针对不同的波束形成器对信号失真的敏感程度不同,可以通过调节改进维纳滤波器的参数,获得噪声抑制与信号失真之间的最佳折中,从而提高输出信噪比,降低目标方位估计的信噪比门限和均方根误差。仿真和实验结果验证了本文方法。     

皮层脑电的非线性降噪

引入基于对非线性动力学局部线性拟合的局部投影非线性降噪方法对Spragure-Dawley大鼠的皮层脑电进行降噪.为了提高降噪效果，利用返回图法对皮层脑电降噪时所需要的最佳局部邻域尺度进行了估计.首先以被50%的高斯白噪声污染的Lorenz方程x轴为例进行降噪，说明根据降噪理论所编写的计算程序的正确性.然后将此降噪方法分别应用于被麻醉的大鼠的皮层脑电和青霉素溶液诱发癫痫发作的皮层脑电时间序列，并采用非线性预报分析说明降噪的效果. 关键词： 皮层脑电 返回图法 非线性降噪 非线性预报     

复式降噪块地铁车轮减振降噪特性

为探究一种复式降噪块装置及其组合形式对某S型辐板地铁车轮的减振降噪效果和机理,在半消声室内,分别对1种自由状态下的标准车轮和3种形式的复式降噪块车轮开展阻尼特性及振动声辐射特性试验,并通过有限元建模对其进行了模态计算。结果表明:复式降噪块装置可在全频段内提高车轮阻尼比,并对车轮各部位有良好的减振效果,以轮辋和踏面的减振效果最为显著;其中,6个制振阻尼片形式的降噪块对车轮的降噪效果最显著,径向激励下的降噪量为13.1 dB(A),轴向激励下的降噪量为11.1 dB(A),降噪频段主要集中在1000 Hz以上中高频。该文研究结果是对列车降噪研究领域的补充和发展。     

Single-channel speech enhancement method using reconstructive NMF with spectrotemporal speech presence probabilities

In this paper, a novel single microphone channel-based speech enhancement technique is presented. While most of the conventional nonnegative matrix factorization-based approaches focus on generating a basis matrix of speech and noise for enhancement, the proposed algorithm performs an additional process to reconstruct speech from noisy speech when these two elements are highly overlapped in selected spectral bands. This process involves a log-spectral amplitude based estimator, which provides the spectrotemporal speech presence probability to obtain a more accurate reconstruction. Moreover, the proposed algorithm applies an unsupervised learning method to the input noise, so it is adaptable to any type of environmental noise without a pre-trained dictionary. The experimental results demonstrate that the proposed algorithm obtains improved speech enhancement performance compared with conventional single channel-based approaches.     

A hybrid microphone array post-filter in a diffuse noise field

In this paper, a hybrid post-filter for microphone arrays with the assumption of a diffuse noise field is proposed to suppress correlated as well as uncorrelated noise. In the proposed post-filter, a modified Zelinski post-filter, which is estimated using the signals on the microphone pairs on which noises are uncorrelated by considering the correlation characteristics of noise impinging on different microphone pairs, is applied to the high frequencies to suppress spatially uncorrelated noise; a single-channel Wiener post-filter is applied to the low frequencies for cancellation of spatially correlated noise. In theory, the proposed post-filter is a Wiener post-filter. In practice, experiments using multi-channel recordings were conducted, and experimental results demonstrate the usefulness and superiority of the proposed post-filter compared to other post-filters using speech quality measures and speech recognition rate.     

Spectral estimation method for noisy data using a noise reference

The development of array processing methods to extract the useful characteristics of acoustic sources such as their locations and absolute levels, starting from the measured sound field is one of the main issues in aero-acoustics. The conventional beamforming method is a very popular technique investigated to solve the power level estimation problem. It has the advantage of being robust, easy to implement and cheap in computation time. However, this technique is also known for having poor spatial resolution capabilities which prevents the correct source levels being obtained for numerous practical applications. Deconvolution techniques of the result computed with CBF, with the point spread function of the array manifold, may restore the power levels of the acoustic sources that would be observed in the absence of the array resolution effects. However, the accuracy of the results provided by deconvolution methods is very sensitive to background noise, always present in acoustic measurements. This process should be carried out after the additive noise has been suitably attenuated and, ideally, the deconvolution operator should amplify the noise as little as possible. Another approach is described in the article. It consists in using a noise reference and a new technique called spectral estimation method with additive noise to remove both the smearing effect produced by the array response and the background noise. The technique has been applied to computer and experimental simulations conducted both in an anechoic chamber and in the test section of an open wind tunnel involving acoustic sources radiating in a noisy environment. The levels of the sources were found with a good level of accuracy and the background noise greatly reduced, confirming the validity of the approach and the satisfactory performance of the method proposed.     

A wavelet-based forward BSS algorithm for acoustic noise reduction and speech enhancement

In this paper, we address the problem of noise reduction and speech enhancement by adaptive filtering algorithm. Recently, the well known forward blind source separation (FBSS) structure has been largely studied and intensively used to reduce acoustic noise components and to enhance speech signal. The FBSS structure is often combined with adaptive algorithms to accelerate the adaptation of the cross-filters, and to improve noise suppression at the output. In this paper, we propose to use a wavelet transform decomposition in the FBSS structure by using a two-channel forward wavelet symmetric adaptive decorrelating (WFSAD) algorithm. The proposed WFSAD algorithm provides a better compromise between time and frequency resolution and improves robustness of the noise reduction process when compared with the classical two-channel forward symmetric adaptive decorrelating (FSAD) algorithm. Simulation results prove the efficiency of the proposed WFBSS algorithm in comparison with conventional ones in terms of several objective and subjective criteria.     

Convex combination of nonlinear adaptive filters for active noise control

This paper proposes a nonlinear active noise control (ANC) system based on convex combination of a functional link artificial neural network (FLANN) and a Volterra filter. Simulation study reveals enhanced noise cancelation performance of the proposed ANC system over the ones based on its component filters.