语音通信降噪研究

语音通信系统中，语音通过信道传输将不可避免地引入码间串扰和信号畸变，同时受到噪声污染。本文在分析自适应盲均衡算法CMA(constant modulus algorithm）和改进盲均衡算法的基础上，考虑到自适应盲均衡技术在语音噪声控制方面能力有限，将自适应盲均衡技术与小波包掩蔽阈值降噪算法联合使用，形成一种基带语音增强新方法。仿真试验结果显示自适应盲均衡技术可以使星座图变得清晰而紧凑，有效减小误码率。研究证实该方法在语音信号ISI和畸变严重情况下，在白噪及有色噪声不同的噪声环境中都具有稳定的降噪能力，消噪同时可获得汉语普通话良好的听觉效果。     

基于灰度冗余的红外图像自适应输出窗技术

根据红外图像的特点，提出了一种基于灰度冗余的红外图像自适应输出窗技术.该技术通过对图像灰度直方图的统计，设定适当的阈值，将灰度像素分布为零的灰度级进行完全压缩，将小于阈值的灰度级映射到阈值灰度级，然后将有效灰度级在整个可显示灰度级范围内作等间距排列.该算法在压缩灰度冗余的同时，实现了对有效图像灰度级的无损均衡，提高了图像质量，并能实现实时处理.     

Optimized analysis of the critical behavior in polymer mixtures from Monte Carlo simulations

A complete outline is given for how to determine the critical properties of polymer mixtures with extrapolation methods similar to the Ferrenberg-Swendsen techniques recently devised for spin systems. By measuring not only averages but the whole distribution of the quantities of interest, it is possible to extrapolate the data obtained in only a few simulations nearT _c over the entire critical region, thereby saving at least 90% of the computer time normally needed to locate susceptibility peaks or cumulant intersections and still getting more precise results. A complete picture of the critical properties of polymer mixtures in the thermodynamic limit is then obtained with finite-size scaling functions. Since the amount of information extracted from a simulation in this way is drastically increased as compared to conventional methods, the investigation of mixtures with long chains or built-in asymmetries is now possible. As an example, the critical points, exponents, and amplitudes of dense, symmetric polymer mixtures with chain lengths ranging fromN=16 up toN=256 are determined within the framework of the 3D bond fluctuation model using grand canonical simulation techniques. As an example for an asymmetry, the generalization of the method to asymmetric monomer potentials is briefly discussed.     

一种新型频域均衡器在水声通信中的应用

在传统频域均衡器的基础上,结合多载波频移键控调制方式的频谱结构,利用接收信号和参考信号的相关性及均衡器的滤波系数,提出了一种兼具均衡与解调功能的新型频域均衡器结构.该均衡器简单、有效地克服了水声信道中多径传播产生的码间干扰和频率选择性衰落等问题,简化了水声通信系统接收机的复杂度. 新型频域均衡器的解调原理以非相关解调算法为基础,但由于利用了均衡器的计算结果,其差错性能优于传统的非相关解调算法,计算机仿真试验表明,改进后的方法使差错性能提高了2 dB;实际水声通信实验表明,该均衡器能够有效地提高水声通信系统的可靠性,具有很好的应用前景.     

环多肽晶体的浮动电荷极化力场模拟

利用原子键电负性均衡结合分子力场方法(ABEEM/MM)对五种环多肽晶体进行了研究. 与传统力场相比, 该方法中的静电势包含了分子内和分子间的静电极化作用, 以及分子内电荷转移影响, 同时加入了化学键等非原子中心电荷位点, 合理地体现了分子中的电荷分布. 相对其他极化力场模型, 具有计算量较小的特点. 该模型下计算得到的环多肽分子单元相对实验测得的结构的原子位置、氢键长度和二面角的均方根偏差分别为0.009 nm、0.013 nm和5.16°, 能够很好地重复实验结果. 总体上, 其结果优于或相当于其他力场模型, 适用于对实际蛋白质体系的模拟和研究.     

Effective contrast enhancement method for color night vision

Image fusion refers to the techniques that integrate complementary information from multiple image sensors’ data in a way that makes the new images more suitable for human visual perception. The paper focuses on the low color contrast problem of linear fusion algorithms with color transfer method. Firstly, the contrast of infrared and visible images is enhanced using local histogram equalization and median filter. Then the two enhanced images are fused into the three components of a Lab image in terms of a simple linear fusion strategy. To enhance the color contrast between the target and the background, the scaling factor is introduced into the transferring equation in the b channel. Experimental results based on three different data sets show that the hot and cold targets are all popped out with intense colors while the background details present natural color appearance. Target detection experiments through target recognition area, detection rate, target-background discrimination also show that the presented method has a better performance than the former methods.     

水声通信中基于信道辨识的盲turbo均衡方法

水声信道多途效应明显,造成接收信号存在严重的码间干扰(ISI,Intersymbol interference)。基于最小均方误差(MMSE,Minimum mean square error)准则的turbo均衡器级联了均衡和信道译码,能够有效去除ISI,并获得优良的性能。由于水声信道的时变性,传统MMSE-turbo均衡需要周期性的训练序列,以实现连续可靠的通信。训练序列虽然提高了通信的可靠性,但降低了信息的有效传输速率。因此,为提高通信效率,本文提出了一种盲turbo均衡方法,该方法通过引入新的盲信道辨识器来同时获得信道估计响应和已去除部分ISI的初步均衡输出信号,并为turbo均衡提供初始的响应参数和比特软信息。与水声通信中应用较多的盲判决反馈均衡器(DFE,Decision feedback equalizer)相比,海上实验结果证明本文提出的盲turbo均衡方法抗信道多途衰落的能力较强,并且与传统MMSE-turbo均衡相比无需训练序列,因此提高了信息的有效传输速率。     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.     

Experimental assessment of a multicarrier underwater acoustic communication system

This paper presents a multicarrier communication system which transmits information on independent subcarriers to achieve an increased data rate. For this system, a passive-phase conjugation (PPC) based receiver structure is assessed by processing data collected in sea trials. Based on temporal diversity (pulse compression) exploited by PPC processing, an adaptive multichannel decision feedback equalizer is used to remove intersymbol interference, where spatial diversity is exploited by adaptive multichannel combination. The digital phase-locked loop (DPLL) technique is implemented for carrier-phase tracking. In the scenario of low input signal-to-noise ratios (SNRs), the receiver structure achieved superior performance using a common DPLL. In a depth-fluctuated environment, two sea experiments were conducted over ranges of 2 km and 4 km, respectively, and this communication system was assessed with a data rate of 4 kbps. In terms of mean square error, output SNR and bit error rate, this receiver structure has demonstrated its performance for the multicarrier communication system.