Digital Whitening Techniques for Improving Spread Spectrum Communications Performance in the Presence of Narrowband Jamming and Interference

Narrowband jamming and interference due to other users in spread spectrum communication systems can be effectively suppressed by using digital whitening. As a result, an impressive improvement in receiver performance can be obtained. The digital whitening is accomplished using a transversal filter whose coefficients are selected by either a Wiener algorithm or a maximum entropy algorithm. Filters obtained by use of these algorithms are evaluated for various jamming and signaling conditions and are found to exhibit comparable performance over a wide range of input signal-to-noise ratios.     

IrBurst Modelling and Performance Analysis in the Presence of Transmission Errors

IrBurst is a session/presentation layer protocol dedicated for high speed large volume information transfer over IrDA links. The protocol is developed by IrDA in order to complement the existing protocol OBEX which is best suited for small files. This paper presents a mathematical model for IrBurst over the IrDA protocol stacks taking into account the presence of bit errors and multiple simultaneously applications. The performance of IrBurst is examined considering TinyTP buffer sizes and the number of IrBurst application connections as parameters. The contribution of this work is to develop a mathematical model for IrBurst, investigate the compatibility of IrBurst to low layer protocols, as well as providing suitable design guidelines for IrDA devices for high performance of IrBurst. Pi Huang received the B.Sc. degree in Electrical and Electronic Engineering from University of Central Lancashire, U.K., in 2001, the M.Sc. degree in Telecommunications from University College London, U.K., in 2002 and the Ph.D degree in the Personal Wireless Networks and Outdoor Optical Links from Bournemouth University, U.K., in 2006. He is currently working in wireless solution division of British Telecom. His research focuses on performance modelling and analysis as well as discrete-event simulation of wireless communication protocols and wireless communication networks. He has published over 20 papers in the areas of wireless communications. Anthony C. Boucouvalas has worked at GEC Hirst Research Centre, and became Group Leader and Divisional Chief Scientist until 1987, when he joined Hewlett Packard (HP) Laboratories as Project Manager. He joined Bournemouth University in 1994 and became a Professor in Multimedia Communications in 1996, and in 1999 became Director of the Microelectronics and Multimedia Research Centre. His current research interests span the fields of wireless communications, optical fibre communications and components, multimedia communications, and human-computer interfaces, where he has published over 200 papers. He has contributed to the formation of IrDA as an industry standard and he is now a Member of the IrDA Architectures Council. He is a Fellow of Fellow of the Royal Society for the encouragement of Arts, Manufacturers and Commerce, (FRSA) and a Fellow of IEE, (FIEE). In 2002 he became a Fellow of the Institute of Electrical and Electronic Engineers (FIEEE), for contributions to optical fibre components and optical wireless communications. He is an Editor of numerous Journals and in the Organising committee of many conferences.     

Speech Interpolation in Digital Transmission Systems

In a telephone conversation a channel is activated by the voice only 25 percent of the time. It is therefore possible to provide a number ofmconnections with a number ofnchannels, wherenis less thanm. When the number of connections is large,m/n(gain) tends to the inverse of the activity. The interpolation procedures used in overload conditions and the resulting degradations, as well as the typical circuits of a voice interpolation apparatus, are examined. Also, two very highly interesting developments are herein illustrated, the ATIC equipment and a single-channel digital speech interpolation (DSI) developed only for measurement purposes. The results obtained from measurement studies carried out with the aforementioned apparatus have enabled the determination of the criteria which must be followed in the selection of the most suitable interpolation procedures. Besides examining the complexity and the reliability which the DSI equipment calls for, it has been possible to analyze the precise working characteristics under normal operating conditions.     

Techniques for Reducing the Visibility of Transmission Errors in Digitally Encoded Video Signals

This paper is concerned with the problem of transmission errors in digitally encoded television signals. Various techniques for reducing the visibility of transmission error effects in the reconstructed signals have been evaluated using computer-produced film sequences. It is found that the visibility of errors in element difference coders is sharply reduced if an erroneous line is replaced by an average of adjacent lines. It is shown that the visibility of errors in two coders that use spatial prediction is low and can be further reduced by using a "leaky" predictor. The implications of these results for in-frame, interframe, and facsimile encoders are discussed.     

数字图像处理中几种平滑技术的研究比较

平滑技术用于平滑图像中的噪声．他是图像处理中的重要组成部分。本文针对均值滤波、中值滤波和自适应滤波三种平滑技术进行了分析比较。     

A Novel Digital Speech Detector for Improving Effective Satellite Capacity

A probabilistic analysis has been done by comparing cumulative distributions of voice and typical telephone-line noise, and the characteristics of voice waveform have been investigated in order to determine a new criterion for distinguishing the two signals. This digital speech detector is extremely sensitive to the onset of voice signal but much less sensitive to typical telephoneline noise.     

基于小波变换的语音质量评价方法研究

主要研究了用于小波变换对语音质量的评价方法。影响语音质量的主要特征表现在噪声部分,即信号高频部分,而小波系数体现了信号的高频;通过构造语音信号的小波范数信噪比,可以定量区分同一语音信号经过不同器件的质量差异,从而确定信号质量;文中通过实验数据验证了该算法在理论上的正确性。     

码分多址数字蜂窝系统容量提高的几种关键技术

个人通信对容量的需求越来越显示出码分多址技术的重要性。本文对照频分多址和蝗分多址系统的频谱效率，阐述了码分多址系统提高系统容量的几项关键技术。     

数字技术与音频指标的关系及提高伴音质量的思路

由数字技术的基本知识入手,简要分析了取样频率、量化精度、系统同步等主要数字技术与频率响应、信噪比、动态范围、失真度等主要模拟音频指标的关系,并结合电视伴音的制作、存储、播出等环节,提出了提高数字电视伴音质量的方法.     

数字电视传输质量指标研究与监测

随着数字电视技术的发展和功能的不断增加,对信号传输和保障的要求也变得更高。尽管付出了更多的努力,但传输方面的故障还是较以前有所增多,处理故障的时间也明显增加,这些都影响着信号的质量,降低了用户的体验。对数字电视传输质量指标进行研究梳理,提出日常监测的方向,为及时发现和快速处理故障提供依据,可以使排查故障的思路变得更加清晰,提高排除故障的效率,从根本上提高数字电视的传输质量。     

Subjective Quality of the Same Speech Transmission Conditions in Seven Different Countries

Because of the diversity of subjective testing methods, it is not possible, in general, to compare speech quality measurements from different laboratories. A recent experiment had the aim of determining whether comparable results can be obtained when the same test is performed in several different countries. Tape recordings of speech samples in the native languages of Britain, Canada, France, Italy, Japan, Norway, and the United States were processed by 38 communications circuits in the Communications Methods Research Department of Bell Laboratories. Recordings of the processed speech were returned to each location for evaluation, on a five-point scale, by native listeners. The communications circuits included white noise and speechmodulated noise references, pulse code modulation (PCM) references and four adaptive differential PCM (ADPCM) test circuits, with bit rates between 16 and 48 kbits/s, each with four binary error rates. An important part of the data analysis was estimation of the quality of the 16 ADPCM conditions at locationB, given measurements of ADPCM quality at locationAand knowledge of reference-circuit quality measured atAand atB. The subjective testing data indicate that a reasonably accurate estimate of ADPCM quality at locationBis the quality measured atAadjusted by an additive constant which is the difference between average reference-circuit measurements atBand atA. The accuracy of this estimate is equivalent to that of a subjective test at locationBwith about seven evaluations of each transmission condition. Three physical measurements are less reliable guides to subjective quality. They lead to estimates with less accuracy than the average of three subjective ratings per circuit.     

Approximation Quality Improvement Techniques in Progressive Image Transmission

Progressive transmission of images has proved to be an efficient way of reaching effective bit rate reductions in highly interactive contexts such as telebrowsing systems, or low bandwidth communication channels. However, the lack of quality of the intermediate approximations has so far limited the obtained effective compression to a small factor. Three different and compatible techniques are presented to improve this quality: use of smoother interpolation schemes in its generation, a nonuniform decomposition procedure to favor the early refinement of relevant areas of the image, and an extension of the transmission hierarchy to pixel quantization producing successive refinement of the grey level resolution of the approximations. Combined use of these approaches yields effective bit rate reductions of up to 12.5:1, considerably extending the range and power of progressive transmission methods.     

Comparison of Performance of Digital Modulation Techniques in the Presence of Adjacent Channel Interference

The performance of four digital modulation techniques are compared to one another when each is received in the presence of adjacent channel interference. The interfering waveforms that each modulation format sees are identical to the modulation format under consideration. The four types of modulation are QPSK, 8-PSK, 16QASK, and QPR. Upper bounds to the probability of error are derived and used to evaluate the relative merits of the different schemes.     

A Comparison of Digital Transmission Techniques for Standard FM Mobile Radio Sets

Digital transmission for existing standard FM mobile radio sets is discussed. The transmission properties of such standard sets are presented. Various data pre-encoding techniques and the corresponding decoders are considered and their performances compared, The comparison includes consideration of the radiated FM spectra, the measured bit error rate curves, and the necessary complexity.     

一种图像自适应平滑与增强算法

图像的平滑与增强是数字图像处理过程中的重要步聚,图像平滑可以消减噪声,但会丢失细节,增强可以突出边缘但同时会放大噪声。为了解决图像平滑与增强之间的矛盾关系,提出一种图像的自适应平滑与增强方法,在进行滤波时,动态确定所处理像素点的滤波权值,在平滑了图像区域内部的同时,又对图像中的区域边缘进行锐化增强,有效解决了图像平滑与增强的矛盾。     

Simulation of MC-CDMA in the Presence of Timing Errors

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Performance of MPEG Codecs in the Presence of Errors

MPEG is emerging as a major international standard for applications in compressed video storage, transmission, and interactive communications. MPEG will be used in applications which involve imperfect channel conditions or storage defects. This work studies the effects of different types of errors on MPEG-encoded bit streams and possible approaches to minimize such effects. The applicability and tradeoff of forward error correction, block interleaving, and error concealment to MPEG-encoded bit streams are also investigated.     

改善移动通信高楼通话质量的方案

通过对无线网络参数、网络结构、天线馈线参数的适当调整等方法以及采用微蜂窝和多层网设计等技术 ,能有效地抑制无线干扰 ,改善移动通信高楼通话的质量。     

考虑载机位置误差的CLS多机无源定位算法

多机无源定位中存在载机位置误差却不予考虑时必然会降低目标的定位跟踪精度。为了解决存在载机位置误差情况下的定位问题,提出了一种考虑载机位置误差的约束最小二乘（CLS）多机无源定位算法。该算法对伪线性观测方程中由于测量误差和载机位置误差而导致的增广系数矩阵的误差协方差阵进行约束,并对伪线性观测方程的误差进行约束最小二乘处理,最终转化为对一组矩阵束的广义特征分解问题。仿真结果表明,相对于最小二乘（LS）算法和扩展卡尔曼滤波（EKF）算法,该算法具有更快的收敛速度和较高的定位精度,并且受载机位置误差影响小,在观测噪声比较大时仍能保持良好的定位性能。