数据采集与处理系统芯片-ADuC845及应用

随着嵌入式系统的发展，模拟信号处理单元与MCU正逐步融为一体，数据采集与处理系统芯片-ADuC845是这一融合的具体体现和产物。现对ADI公司的ADuC845芯片及其特性、应用作详细介绍。     

Lexical Influences in Audiovisual Speech Perception.

Phoneme identification with audiovisually discrepant stimuli is influenced hy information in the visual signal (the McGurk effect). Additionally, lexical status affects identification of auditorily presented phonemes. The present study tested for lexical influences on the McGurk effect. Participants identified phonemes in audiovisually discrepant stimuli in which lexical status of the auditory component and of a visually influenced percept was independently varied. Visually influenced (McGurk) responses were more frequent when they formed a word and when the auditory signal was a nonword (Experiment 1). Lexical effects were larger for slow than for fast responses (Experiment 2), as with auditory speech, and were replicated with stimuli matched on physical properties (Experiment 3). These results are consistent with models in which lexical processing of speech is modality independent. (PsycINFO Database Record (c) 2010 APA, all rights reserved)     

Measurement accuracy enhancements for wideband modulated signals

Time‐domain waveform measurements of wideband modulated signals, although attractive for communication systems modeling applications, have until now lacked the required accuracy for quantitative analysis. New developments described here enhance the accuracy of these measurements, enabling their application to communication system characterization and modeling. © 2003 Wiley Periodicals, Inc. Int J RF and Microwave CAE 13: 32–39, 2003.     

Modeling Effects of Rhythmic Context on Perceived Duration: A Comparison of Interval and Entrainment Approaches to Short-Interval Timing.

Relative merits of interval and entrainment conceptions of the internal clock were assessed within a common theoretical framework by 4 time-judgment experiments. The timing of tone onsets marking the beginning and ending of standard and comparison time intervals relative to a context rhythm were manipulated: onsets were on time, early, or late relative to the implied rhythm, and 2 distinct accuracy patterns emerged. A quadratic ending profile indicated best performance when the standard ended on time and worst performance when it was early or late, whereas a flat beginning profile (Experiments 1-3) indicated uniform performance for the 3 expectancy conditions. Only in Experiment 4, in which deviations from expected onset times were large, did significant effects of beginning times appear in time-discrimination thresholds and points of subjective equality. Findings are discussed in the context of theoretical assumptions about clock resetting, the representation of time, and independence of successive time intervals. (PsycINFO Database Record (c) 2010 APA, all rights reserved)     

Method of increasing the output signal-to-noise ratio in passive multi-element antenna arrays

A method of processing complex signals in passive multi-element antenna arrays based on an analysis of the natural values of the correlation matrix of the signal is considered. By means of the method it is possible to substantially increase the output signal-to-noise ratio. An iterational process by means of which optimal estimators of both the signal and the amplitude-phase distribution may be obtained, is proposed. Theoretical estimators of the efficiency of the method are given. It is proved that, in terms of the efficiency, the approach considered in the article comes close to the technique of matched filtration. Results of statistical modeling that confirm the statistical results are presented.     

Optimal search for signals in a multichannel system: Arbitrary initial distribution in state space merging scheme

Markovian and semi-Markovian random processes are used to analyze the problem of optimal search for signals in a multichannel communication system with arbitrarily distributed random outputs. The search efficiency factor is found in explicit form based on state space merging, and a mathematical programming problem is set up to find a numerical suboptimal solution. __________ Translated from Kibernetika i Sistemnyi Analiz, No. 3, pp. 144–150, May–June 2007.     

Nonlinear suboptimal filtration of pseudorandom sequences

The critical importance of studies of alternative methods of representation and processing of signals constructed from pseudorandom sequences of maximal period is demonstrated. For the first time, it is proposed that a multidimensional nonlinear finite-difference representation of an arbitrary pseudorandom sequence be employed. A suboptimal algorithm for nonlinear discrete Kalman filtration of a pseudorandom sequence is obtained. A numerical example of filtration of a sequence formed on the basis of a given recurrence rule is considered and its efficiency is demonstrated.     

一种实用的内插FFT信号处理方法

基于信号样本可以重建信号频谱包络,本文提出一种简单、实用的内插方法,将此方法的有效性和FFT的优越性结合起来应用于细化频率分辨率,以适应需要获取信号较多频谱信息的应用场合。     

Research on resolution between multi-component LFM signals in the fractional Fourier domain

When the initial frequencies and chirp rates of multi-component linear frequency modulation (LFM or chirp) signals are close,the signals may not be distinguished in the fractional Fourier domain (FRFD).Consequently,some signals cannot be detected.In this paper,first,the spectral distribution characteristics of a continuous LFM signal in the FRFD are analyzed,and then the spectral distribution characteristics of a LFM signal in the discrete FRFD are analyzed.Second,the critical resolution distance between the peaks of two LFM signals in the FRFD is deduced,and the relationship between the dimensional normalization parameter and the distance between two LFM signals in the FRFD is also deduced.It is discovered that selecting a proper dimensional normalization parameter can increase the distance.Finally,a method to select the parameter is proposed,which can improve the resolution ability of the fractional Fourier transform (FRFT).Its effectiveness is verified by simulation results.     

针对语音变换的语音篡改检测

针对使用语音变换技术的语音篡改,提出一种自动检测方法。在分析语音变换基本模型和变换语音失真的基础上,提取语音信号的声道参数以及相关的信号统计量,并通过支持向量机递归特征消除法,选择出对语音变换比较敏感的特征作为分类特征,使用支持向量机进行语音变换检测和变换语音的说话人性别判别。对于一种语音变换软件的实验结果表明,该方法具有较高的检测准确率,其中语音变换检测的平均准确率为94.90%,变换语音的说话人性别判别平均准确率为92.09%。