Speech Data Rate Reduction Part II: Applicability of Sensitivity and Error Analysis

This paper deals with the application of modern estimation techniques to the problem of speech data rate reduction. It is desirable to adaptively identify and quantitize the parameters of the speech model. These paramaters cannot be identified and quantized exactly; the performance of the predictor is thereby degraded and this could prevent data reduction. In many cases it is desirable to emply a suboptimal predictor in order to simplify the algorithms, and predictor performance is again degraded. This paper develops sensitivity and error analysis as a potential method for determining quantitatively how speech data reduction system performance is degraded by imprecise parameter knowledge or suboptimal filtering. An intended use of the sensitivity and error analysis algorithms is to determine parameter identification and model structure requirements of configuration concepts for adaptive speech digitizers. First, sensitivity and error analysis algorithms are presented that form the basis for the remainder of the work. The algorithms are then used to determine how imprecise knowledge of vocal tract parameters degrades predictor performance in speech. Transversal filters have previously been proposed for this application. The sensitivity analysis algorithms are then used to determine when and by how much the transverse filter is suboptimal to the Kalman filter. In particular, the question of how effectively a higher order of all-pole model approximates a system with zeros is answered, as this question is of considerable importance in speech. Finally, the physical significance of the innovations process in speech data rate reduction is studied.     

基于延迟发送的高速信号遥测采编技术

针对导弹与目标相遇时刻引信产生的高速信号的数据分析需求,提出了一种基于延迟发送的遥测采编技术。数据采集采用了缓存后延迟发送的方式,在所需信号的特征未满足发送要求前,对数据通道持续检测,并将采集到的数据缓存入RAM(Random Access Memory,随机存取存储器)中,此时发送其余信号的遥测数据;当所需信号满足发送要求后,立即中断其余信号的发送,将RAM中存储的数据读出并发送:这样不仅降低了系统码速率,还可以避免高码速率实时发送带来的不利影响。数据编码采用了分层量化的方式,提高了数据分析时较为关注的小幅度信号的分辨率。该遥测采编技术方案具有缩短研制周期、节约研制成本、降低技术风险的优点,并在飞行试验中得到成功应用。     

音频信号矢量编码算法

为了在保证译码恢复的声音质量良好的前提下,减小编码的压缩率,以减小声音信号的存储空间,提出了一种将线性预测编码、SOM神经网络矢量编码以及Huffman编码相结合的声音信号编码算法,将1列声音信号转换为2列信号,这样就可以进行后续的矢量编码。实现了预测编码和矢量编码的结合。利用Matlab软件编程进行了声音信号编解码实验。实验结果表明,在保证声音质量的前提下,该编码方法的码率小于MEPG-1 Layer3的最低的64kbps标准码率,且算法简单。文章提出的编码算法在音频压缩编码方面将具有较高的研究价值和很好的应用前景。     

On the Accuracy And Resolution of Radar Signals

Information matrices are derived for estimates of the range parameters of moving targets as obtained by combining a priori information (if available) with reflected radar signals observed in the presence of additive white Gaussian noise. The inverse of the information matrix provides a lower bound on the covariance matrix of any unbiased parameter estimates. This bound can be approached with a high signal-to-noise ratio and optimum data processing (matched filters). Arbitrary frequency modulation, amplitude modulation, and target motion as well as various assumptions on processing the RF phase are considered. The multiple-target case makes possible investigation of a signal's resolution ability, as well as its accuracy potentials. Results for a carrier frequency much greater than the effective signal bandwidth are obtained as a special case. A main purpose of the paper is the reduction of the original radar problem to a linear model which is equivalent in the sense of having the same information matrix. These models provide valuable insight into the relative effects of multiple targets, choice of modulation, a priori information, and assumptions regarding RF phase and bandwidth. The linear equivalent model also leads to a valuable computational algorithm for investigations using digital or hybrid computers. The various special cases of interest are obtained by simple modifications of the general case, and thus the algorithm can provide a very versatile tool for evaluating and designing radar signals.     

Direction-of-arrival estimation using signal subspace modeling

A computationally viable algorithm for estimating the direction-of-arrival (DOA) of multiple wavefields that are incident on an array of sensors is developed. The geometry of the array is unrestricted and the incident wavefields may be generated by mixtures of incoherent and coherent emitting sources. In the approach taken, the sensor signals are modeled as a noise-contaminated linear combination of steering vectors which are functionally dependent on a set of DOA parameters. These parameters are to be chosen so that this sensor signal model is most compatible with empirically measured data (i.e., snapshot data). An iterative procedure is developed for selecting the most data compatible set of DOA parameters in both the snapshot domain and the array covariance matrix domain. Critical to the success of such iterative solution procedures is the generation of quality DOA parameters to initialize the algorithm. A sequential beamforming method for this initialization is presented. Numerical examples to illustrate the effectiveness of the proposed algorithmic approach are given     

基于AC4880xC—C声码器的话音通信设备的设计

介绍了基于奥科公司AC4880xC-C声码器和利用低速率话音压缩编码技术（ITU-T G.729A）设计低比特率话音通信设备的方法。理论与实验表明，该系统具有编码速率低，合成语音质量高，抗误码性能好等优点，可有效提高宽带利用率，具有很好的应用前景。     

A Distribution-Free Doppler Processor

A distribution-free Doppler processor (DFDP) is described. This procedure is applicable to the detection of signals with unknown phase or signals which undergo a change in phase from one observation to the next. The procedure has the characteristic that the false-alarm probability is a constant, independent of the probability distribution of the received data when no signal is present. Also, an estimate of the Doppler frequency, when applied to a Doppler radar processor, is provided.     

横流不稳定性条件下初始扰动增长实验研究

在具有近似等顺压梯度的45°后掠平板上，人工激励非定常扰动的条件下，进行了三维边界层承受横流不稳定性实验研究。在扰动增长的初始阶段，信号的信噪比非常低，本文首次成功测量了初始扰动的增长，将有用的信号有效地提取出来。对于不同的来流速度与不同的人工扰动频率与波数，一系列的实验研究提供了一组初始扰动的增长曲线。     

无线列车调度实现数据话音同传

为了保证行车安全,在无线列车调度系统中,利用话音压缩/扩展技术实现同一无线信道数据话音同时传输.在分析和比较了几种话音编码方式的基础上,确定本系统采用连续可变斜率增量调制方式(CVSD)编解码方式,该编解码方式具有较好的抗误码特性及量化信噪比.根据CVSD的编解码特性,提出了话音压缩/扩展技术在系统中的具体实现.采用TMS320C3X系列DSP芯片实现了该系统功能.     

Nonparametric Radar Extraction Using a Generalized Sign Test

A nonparametric procedure used in a constant false alarm rate (CFAR) radar extractor for detecting targets in a background of noise with unknown statistical properties is described. The detector is based on a generalization of the well-known two-sample sign test and thus requires a set of reference noise observations in addition to the set of observations being tested for signal presence. The detection performance against Gaussian noise is determined for a finite number of observations and asymptotically, for both nonfluctuating and pulse-to-pulse Rayleigh fluctuating target statistics. It is noted that the performance loss, as compared to the optimum parametric detector, depends critically on the number of reference noise observations available when the number of hits per target is not large. In the same case a much larger loss is also found for a pulse-to-pulse fluctuating target even though the asymptotic loss is the same as for a nonfluctuating target. A comparison is finally made with a detector based on the Mann-Whitney test, which usually is considered to be one of the better nonparametric procedures for the two-sample case.     

音频信号联合编码的通用码书研究

为了减小音频信号编码算法的运算量,提出了在线性预测编码、SOM神经网络矢量编码以及Huffman编码相结合的声音信号联合编码算法(以下简称联合编码)的基础上,将算法中原有的专用码书改为通用码书的算法。利用Matlab软件编程进行了专用码书和通用码书条件下的声音信号编解码实验。实验结果表明,在码率相等的条件下,使用通用码书和专用码书的联合编码方法可以得到相近的译码声音质量。使用通用码书的联合编码的最低码率在音频编码格式Opus码率的范围内,且接近Opus码率的下限,而且算法较Opus编码简单,因而实时性较好。文章提出的编码算法可为音频压缩编码的进一步研究提供参考。     

Multiplexing Asynchronous Signals Using Bit Stuffing and Retiming Techniques

This paper discusses the problem of multiplexing asynchronous signals which are clocked from independent sources. Two solutions are offered to this problem: one provides a continuous transmission over a fixed period of time proportional to the bit rate integrity duration, and the other utilizes a bit stuffing technique. The former approach is used when the asynchronous signals are clocked from highly stable clocks, and also when the bit rate integrity period is on the order of a few hours. The latter approach is employed when the signal clocks are relatively less stable or the bit rate integrity period is very long. The reliability of these techniques is analyzed, and their efficiencies are evaluated and found to be most satisfactory. Synchronization procedures and probabilities of false lock and failure to lock are also described. In addition, a multiplex system illustrating these techniques is presented. The system multiplexes four input channels at a one megabit rate for each.     

齿轮故障诊断中的单齿分析技术

齿轮故障诊断通常先对振动信号进行时域同步平均降噪处理,进而利用频谱分析或包络分析等方法,得出故障特征信息。本文在此基础上提出了单齿分析技术。它将振动平均信号等距分段(所分段数与被监测齿轮齿数相等),然后对分段信号进行频谱分析等。比较各段信号的异同以揭示故障齿的确切位置。该方法对齿轮的表面局部损伤故障的早期诊断和断齿故障早期诊断十分有效,可以准确地辨别出有故障齿的位置,为齿轮故障诊断提供了一种有效手段。      

Radar CFAR Thresholding in Clutter and Multiple Target Situations

Radar detection procedures involve the comparison of the received signal amplitude to a threshold. In order to obtain a constant false-alarm rate (CFAR), an adaptive threshold must be applied reflecting the local clutter situation. The cell averaging approach, for example, is an adaptive procedure. A CFAR method is discussed using as the CFAR threshold one single value selected from the so-called ordered statistic (this method is fundamentally different from a rank statistic). This procedure has some advantages over cell averaging CFAR, especially in cases where more than one target is present within the reference window on which estimation of the local clutter situation is based, or where this reference window is crossing clutter edges.     

基于相关函数的盲信号分离准则

盲信号分离的依据通常是信号的独立性或利用它们的高阶累积量。考虑了双通道的盲分离问题，提出了一种基于相关函数的盲分离准则。证明了在源信号不相关的条件下，可以实现源信号的盲分离。并给出了一种可直接求解二次方程组的盲分离算法。     

基于旋转轴向阵列的风扇宽频噪声实验

航空发动机降噪研究迫切需要一种叶轮机械管道内宽频噪声测量方法来指导降噪设计。本文通过对阵列测量的声压信号进行互相关分析,得到管道内顺流和逆流传播的模态声功率结果。安装在风扇实验台进口段的传声器阵列由2排周向间隔180°的轴向阵列组成,每排阵列有14个等间距的传声器。阵列安装在可周向旋转的测量段上,实验中测量段每隔6°旋转一次,共获得840个测点位置的声场信号。结果表明入射波与反射波最大可相差10dB。模态分解结果表明,转静干涉模态是转子通过频率及其谐频处的主导模态。利用不同参考信号计算出的声场结果相同,说明该实验测试方法对参考信号位置没有特殊要求,进一步说明该方法有很好的适用性。     

减速器振动信号的经验模态分解

时域同步平均是直升机减速器诊断技术的基础,目前这种方法依赖于转速传感器提供相位同步信号。探讨了应用经验模态分解代替时域同步平均分析减速器振动信号的方法。构建了一个减速器振动信号模型,提取了故障特征信号。对经验模态分解过程进行了理论推导,证明经验模态分解可以分离出故障特征信号,给出了信号分离的充分条件。将这种方法应用于直升机减速器的两种故障(点蚀和裂纹)振动数据,结果表明经验模态分解正确地分离出了故障特征信号,信号特征更为显著。     

Impulse feature extraction method for machinery fault detection using fusion sparse coding and online dictionary learning

Impulse components in vibration signals are important fault features of complex machines. Sparse coding(SC) algorithm has been introduced as an impulse feature extraction method, but it could not guarantee a satisfactory performance in processing vibration signals with heavy background noises. In this paper, a method based on fusion sparse coding(FSC) and online dictionary learning is proposed to extract impulses efficiently. Firstly, fusion scheme of different sparse coding algorithms is presented to ensure higher reconstruction accuracy. Then, an improved online dictionary learning method using FSC scheme is established to obtain redundant dictionary and it can capture specific features of training samples and reconstruct the sparse approximation of vibration signals. Simulation shows that this method has a good performance in solving sparse coefficients and training redundant dictionary compared with other methods. Lastly, the proposed method is further applied to processing aircraft engine rotor vibration signals. Compared with other feature extraction approaches, our method can extract impulse features accurately and efficiently from heavy noisy vibration signal, which has significant supports for machinery fault detection and diagnosis.     

基于拉普拉斯正态混合分布概率密度函数估计的语音分离算法

由拉普拉斯分布概率密度函数推导得出的符号函数作为语音信号分离算法中的激活函数,目前广泛应用在基于独立分量分析的语音分离算法之中.为了提高语音分离算法的收敛速度以及分离性能,提出把拉普拉斯正态混合分布概率密度函数作为语音信号概率密度函数的估计,得到一个更加适合语音信号分离的激活函数,基于此函数提出一种快速语音分离算法.把推导出的激活函数应用于独立分量分析(ICA)的自然梯度算法中进行计算机仿真实验,可验证本文算法的收敛性能和分离性能.     

信号周期分段处理法诊断单级传动齿轮副故障

为了诊断单级传动齿轮副故障,本文提出信号周期分段处理的方法。首先同步整周期采集振动信号,然后对采集的振动信号做时域平均,进行降噪和凸显故障脉冲的处理。利用周期分段处理法,将振动平均信号等距分段,分别对各段信号进行频谱分析。比较各段信号的时域波形和频谱的异同,可以准确诊断出单级传动齿轮副中同步齿轮故障位置,且还能诊断出非同步齿轮故障位置。数字仿真试验以及模拟故障实验结果都证实了信号周期分段处理法诊断齿轮故障的可行性和有效性;它为诊断单级传动齿轮副故障提供一种有效手段。