D类音频功放的可集成Pop-Click噪声抑制系统

设计实现了一种可集成于D类音频功放芯片内部的Pop-Click噪声抑制系统,通过全新的输出级软启动控制以及辅助反馈环路的采用,保证了软启动过程中的环路稳定性和输出驱动级的完整性,实现良好稳定的PopClick噪声抑制。采用0.5μm CMOS工艺实现了集成Pop-Click噪声抑制系统的2.0 W单声道D类音频功放。测试结果显示,在单位增益、8Ω喇叭负载下,该D类音频功放的Pop-Click噪声能被有效抑制至1.5 mV内。     

OFDM系统下脉冲噪声的时频联合抑制方法

本文分析了VHF或较低UHF频段下出现的脉冲噪声对OFDM系统下信号接收的影响.基于Mobile WiMAX的下行链路和Bemouli Gaussian脉冲噪声模型,提出时频二维相结合的脉冲噪声联合抑制算法.仿真结果表明通过联合抑制方法,符号错误率和比特错误概率得到了明显的改善.     

Phase noise floor suppression of the output carrier from double sideband-carrier suppressed modulation system

A theoretical model of all the influencing factors on the phase noise floor of the output carrier from the double sideband-carrier suppressed (DSB-CS) modulation system is presented. Based on the established DSB-CS modulation system with different cases, the phase noise of the input carrier and the output doubled frequency carrier is measured and calculated using the theoretical model, respectively. The calculation results basically agree with the measurement results. It is shown that the phase noise floor ...     

Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms

Stereophonic acoustic echo cancellation has generated much interest in recent years due to the nonuniqueness and misalignment problems that are caused by the strong interchannel signal coherence. In this paper, we introduce a novel adaptive filtering approach to reduce interchannel coherence which is based on a selective-tap updating procedure. This tap-selection technique is then applied to the normalized least-mean-square, affine projection and recursive least squares algorithms for stereophonic acoustic echo cancellation. Simulation results for the proposed algorithms have shown a significant improvement in convergence rate compared with existing techniques.     

Clustered and deep echo state networks for signal noise reduction

Machine Learning - Echo State Networks (ESNs) are Recurrent Neural Networks with fixed input and internal (hidden) weights, and adaptable output weights. The hidden part of an ESN can be considered...     

Robust extended multidelay filter and double-talk detector for acoustic echo cancellation

We propose an integrated acoustic echo cancellation solution based on a novel class of efficient and robust adaptive algorithms in the frequency domain, the extended multidelay filter (EMDF). The approach is tailored to very long adaptive filters and highly auto-correlated input signals as they arise in wideband full-duplex audio applications. The EMDF algorithm allows an attractive tradeoff between the well-known multidelay filter and the recursive least-squares algorithm. It exhibits fast convergence, superior tracking capabilities of the signal statistics, and very low delay. The low computational complexity of the conventional frequency-domain adaptive algorithms can be maintained thanks to efficient fast realizations. We also show how this approach can be combined efficiently with a suitable double-talk detector (DTD). We consider a corresponding extension of a recently proposed DTD based on a normalized cross-correlation vector whose performance was shown to be superior compared to other DTDs based on the cross-correlation coefficient. Since the resulting DTD also has an EMDF structure it is easy to implement, and the fast realization also carries over to the DTD scheme. Moreover, as the robustness issue during double talk is particularly crucial for fast-converging algorithms, we apply the concept of robust statistics into our extended frequency-domain approach. Due to the robust generalization of the cost function leading to a so-called M-estimator, the algorithms become inherently less sensitive to outliers, i.e., short bursts that may be caused by inevitable detection failures of a DTD. The proposed structure is also well suited for an efficient generalization to the multichannel case.     

抑制温度信号噪声方法的研究

在空间相机中,温度传感器输出的信号易受到噪声的干扰从而造成有用信号的淹没,因此研究适用于温度信号的噪声抑制方法具有重要的实用价值.本文对干扰温度信号的噪声来源进行了综合分析,并针对温度信号所含噪声的特点,采用中值滤波法和小波阈值滤波的复合数字滤波算法相结合,对温度信号进行滤波,实现了信噪分离的目的,提高了信噪比,并通过MATLAB的仿真功能对输出的信号进行验证.仿真表明,这两种方法的混合使用对温度信号的噪声的抑制有明显的效果.     

Measurement of vorticity in the surface layer using an acoustic echo sounder array

The earth's atmospheric surface layer is usually defined as that region of the lower atmosphere (generally below about 10 m above the earth's surface) where surface friction causes vertical fluxes of heat, moisture and momentum to be constant with height. Within the surface layer, either in response to surface friction or to the atmosphere above, horizontal circular eddies often develop. These circular motions may provide the source of rotation for dust devils so often seen on hot and dry days particularly in desert regions. Also, at larger scales ($\text{～ 1 km in diameter}$) regions of warm and therefore buoyant upward moving air, called thermal plumes, may acquire rotation. These plumes may extend from the earth's surface to more than a kilometer in height on a warm afternoon. Fluid dynamicists quantify this horizontal rotation with a parameter known as the vertical component of vorticity. Vorticity is very difficult to measure in the earth's atmosphere at scales of close to a kilometer because the calculation involves wind-speed differences over horizontal distances of about 500 m. The winds must be measured quite accurately because the differences can be quite small and, therefore, the errors in these measurements are often quite large. This work describes a method of measuring vertical vorticity at scales down to 500 m using an array of three acoustic sounders about 2 m above the earth's surface which overcomes some of the accuracy problems mentioned above. We relate these vorticity measurements to other atmospheric parameters and compute temporal spectra of these quantities to help explain the relationship between vorticity, thermal plume activity, and the smaller-scale dust devils.     

Real-time speech enhancement algorithm for transient noise suppression

Multimedia Tools and Applications - To effectively restrain stationary noise and transient noise, a real-time single-channel speech enhancement algorithm is proposed. First, to evaluate stationary...     

隧道磁电阻传感器的噪声抑制设计与测试

针对隧道磁电阻传感器拥有很高的灵敏度却存在本底噪声过大的问题,设计了三轴线性隧道磁电阻传感器的探头电路、调制电路、低噪声放大电路。基于商用数字锁相放大器,对调制放大后的输出信号进行解调处理。经实验测试,系统三轴的噪声频谱密度在1 Hz处分别为0. 426,0. 497,0. 513n T/■,较传感器本底噪声有了明显降低。实验结果表明:系统的设计对噪声的抑制起到了一定的作用,适用于一般传感器的降噪处理。     

基于DSP的声反馈抑制系统的研究与实现

提出了一种基于TI公司TMS320C6713 DSP和移频法抑制声反馈的有效方法。该方法采用能与之无缝连接的TLV320AIC23 Codec芯片作为语音采集和回放工具,然后基于在Matlab进行仿真达到抑制啸叫相当理想的基础上完成了在DSP上的实时实现。最后,采用主观法和客观法评估了输出语音的质量。结果表明,该方法能有效抑制再生混响干扰,明显提高了扩声增益,且显著改善了频响特性和声音清晰度。     

频率-空间域EMD压制随机干扰与相干噪声

提出一种在f-x域中利用经验模式分解去除相干噪声与随机干扰的方法,将它用于迭后记录的去噪处理,并与使用自回归模型的f-x线性预测滤波结果做了比较,表明这种方法能有效去除迭后噪声。     

Performance analysis of under-modelling stereophonic acoustic echo cancellation by adaptive filtering LMS algorithm

This paper addresses the field of stereophonic acoustic echo cancellation (SAEC) with adaptive filtering algorithms. In SAEC applications, using the least mean square (LMS) algorithm, it is usually assumed that the lengths of the adaptive filters are equal to that of the unidentified system responses. Although, in many realistic situations, under-modelled lengths adaptive filters, whose lengths are less than that of the unidentified systems (under-modelled systems), are employed, and analysis results for the exact modelled stereophonic LMS algorithm are not automatically appropriate to the under-modeled lengths. In this paper, we present a statistical analysis of the under-modeled stereophonic LMS algorithm. Exact expressions and deterministic recursive equations to the mean coefficients behavior of the adaptive LMS filters are derived to completely characterize and assess the performances (transient and steady-state) of the under-modeling stereophonic LMS algorithm. The expected theoretical behaviour is compared with Monte Carlo simulations and practical experimental results, showing a very good agreement.     

Efficient blind dereverberation and echo cancellation based on independent component analysis for actual acoustic signals

This letter presents a new algorithm for blind dereverberation and echo cancellation based on independent component analysis (ICA) for actual acoustic signals. We focus on frequency domain ICA (FD-ICA) because its computational cost and speed of learning convergence are sufficiently reasonable for practical applications such as hands-free speech recognition. In applying conventional FD-ICA as a preprocessing of automatic speech recognition in noisy environments, one of the most critical problems is how to cope with reverberations. To extract a clean signal from the reverberant observation, we model the separation process in the short-time Fourier transform domain and apply the multiple input/output inverse-filtering theorem (MINT) to the FD-ICA separation model. A naive implementation of this method is computationally expensive, because its time complexity is the second order of reverberation time. Therefore, the main issue in dereverberation is to reduce the high computational cost of ICA. In this letter, we reduce the computational complexity to the linear order of the reverberation time by using two techniques: (1) a separation model based on the independence of delayed observed signals with MINT and (2) spatial sphering for preprocessing. Experiments show that the computational cost grows in proportion to the linear order of the reverberation time and that our method improves the word correctness of automatic speech recognition by 10 to 20 points in a RT??= 670 ms reverberant environment.     

反射强度调制式光纤声音传感器优化设计与研究

反射强度调制式光纤声音传感器具有结构简单、灵敏度高、抗电磁场和射频干扰等优点。对反射式光纤声音传感器膜片的振动特性进行了理论推导和有限元分析,得出振动膜片几何参数和振动特性之间的关系及其膜片的设计方法。设计了结构简单的传感器探头,并以1310 nm波长的激光器为光源研制了光纤声音传感器。对所研制的光纤声音传感器进行了实验研究,实验测得传感器的频响为350 Hz~7.5 kHz;灵敏度为416.7mV/Pa。     

Integrated topology optimization and optimal control for vibration suppression in structural design

This paper presents a topology design methodology for structures with active vibration control. The methodology takes into account the structural effect of the control forces and includes the modal control design. Location points for the actuation forces are chosen a priori. Structural topology optimization is used for distributing material on a fixed domain, using continuum finite element discretization for the static and free vibration analyses. Optimal control for transient response in modal space is used to derive the control force. The cost function for the optimization combines the strain energy and the control energy. Results of the numerical simulations validates the proposed methodology.     

含乘性噪声的声发射信号降噪方法研究

针对声发射噪声信号的特点,研究了适合含乘性噪声的声发射信号的自适应滤波器和小波分解降噪方法,并对这2种降噪方法的降噪仿真结果进行了对比,结果表明自适应滤波器降噪方法更适合含乘性噪声的声发射信号的降噪处理。     

An efficient stabilized fast Newton adaptive filtering algorithm for stereophonic acoustic echo cancellation SAEC

This paper addresses the field of stereophonic acoustic echo cancellation (SAEC) by adaptive filtering algorithms. Recently, we have proposed a new version of the fast Newton transversal FNTF algorithm for SAEC applications. In this paper, we propose an efficient modification of this algorithm for the same applications. This new algorithm uses a new proposed and simplified numerical stabilization technique and takes into account the cross-correlation between the inputs of the channels. The basic idea is to introduce a small nonlinearity into each channel that has the effect of reducing the inter-channel coherence while not being noticeable for speech due to self masking. The complexity of the proposed algorithm does not alter the complexity of the original version and is kept less than half the complexity of the fastest two-channel FTF filter version. Simulation results and comparisons with the extended two-channel normalized least mean square NLMS and FTF algorithms are presented.