退极化效应对三次谐波产生的影响

用紧致密度矩阵和弛豫时间近似方法，研究退极化效应对三次谐波产生的影响。给出三次谐波产生系数表达式。在半抛物量子阱中，给出三次谐波产生的数值结果。退极化效应破坏了三共振，修改了三次谐波产生谱。使原来的三共振峰退化为一双共振峰和一单共振峰。     

高阶最优LPC根值筛选的共振峰估计算法研究

为解决线性预测（LP）共振峰检测误差较大的问题,针对高阶LP提取共振峰中伪根难以去除及极点交互带来的频谱混叠,提出一种基于高阶最优LP系数根值筛选的共振峰估计算法。考虑提高LP的阶次,获取含有更高线性峰值拟合精度的线性系统根值。采用语音数字共振模型约束共振峰的根幅值范围,滤除非共振峰频率值对应的伪根。结合功率加权来预加重信号的主要频谱成分,提高频谱峰值频率区分度。实验结果表明,所提算法消除了高阶邻近极点间的相互干扰,去除了非共振峰谐波对应的高阶伪根,与其他算法相比,所提算法定位语音信号共振峰频率的误差更低,更能稳定精确地检测共振峰。     

基于BP神经网络的耳语音转换为正常语音的研究

提出了一种基于BP神经网络的汉语耳语音转换为正常语音的方法.首先提取正常语音、耳语音的共振峰参数,使用BP神经网络训练出耳语音到正常语音共振峰参数的转换模型;然后根据模型求出与耳语音对应的正常语音共振峰参数,采用共振峰合成的方法将耳语音转换为正常语音.实验结果表明:使用该方法转换的正常语音DRT得分为80%,MOS得分为3.5,在可懂度和音质方面均达到了满意的效果.     

频域内的卡尔曼滤波语音增强算法

提出了在频域内实现的卡尔曼滤波算法,该算法利用语音和噪声幅度谱的时变特性,先对语音幅度谱进行初步修正,提取较为准确的LPC系数,然后在每一频率点下对语音幅度用卡尔曼滤波进行递推估计,最终得到效果更好的增强语音.实验结果表明,本文算法有效地提高了增强语音的SNR,尤其是在高信噪比的情况下,效果更加明显.     

传感器与固体电子学中非线性函数多项式拟合的规范化

多项式可用于非线性信号的拟合,关键在于求解其各项系数。对于任何非线性函数,文中提出都有一个规范化的拟合方法。相应有一个规范化的多项式。该规范化多项式是以整数n为底的幂级数,最大幂次nmax是x坐标区间的等分数,其系数可用一个规范化的矩阵积得到。我们给出了固体电子学中的两个应用实例。当x坐标区间分段拟合应用时,还讨论了函数及其导数计算值的连续性条件,并以正弦函数不同区间的展开为例,作了演示。     

基于微环谐振腔的耦合谐振光波导反射镜

研究了一种基于微环谐振腔的耦合共振光波导反射镜.基于耦合模理论,对这种新型波导反射镜的反射光谱进行了数值分析,详细讨论了直波导与微环谐振腔之间的耦合系数和相邻微环谐振腔之间的耦合系数对反射光谱的影响.计算结果表明这种反射镜的反射光谱存在多峰结构,在弱耦合的情况下可以实现波长选择性反射滤波,并且发现在这种波导反射镜中存在类似于原子系统中电磁诱导透明现象的耦合共振诱导透明现象.     

基于ARLPM改进的海面目标检测新方法

舰载(或岸基)雷达对海面小目标检测是一个重要而且困难的课题。最大极点幅度检测(ARLPM)是目前实用性较强的一种方法,特别在短时检测方面有着不错的效果。但该算法的缺陷在于只利用了海杂波AR谱的最大幅度极点的谱宽特性,文中在ARLPM的基础上提出了一种新的基于AR模型的方法——双参量检测,它同时利用到了极点的谱宽和谱峰。最后,利用真实海杂波数据对该方法作了实验,通过比较表明,双参量检测优于原方法。     

LPC方法提防语音信号共振峰的分析

通过对ＬＰＣ（线性预测编码）方法提取语音信号共振峰进行的研究表明,采用相一频特性与对数幅－频特性同样能提取语音信号共振峰。与对数据－频特性的二次导数相比,相－频特性的三导数有更高的频率分辨率,能更有效地解决共振峰合并的问题,撮更精确的共振峰参数。     

一类2次多项式混沌系统的均匀化方法研究

该文给出了一般的2次多项式混沌系统与Tent映射拓扑共轭的充分条件,并依据该条件,给出了一类2次多项式混沌系统及其概率密度函数;进一步得到了能够将这类系统均匀化的变换函数;给出了一个新的2次多项式混沌系统并进行均匀化处理,对其产生的序列进行了信息熵、Kolmogorov熵和离散熵分析,结果显示该均匀化方法的均匀化效果显著且不改变序列混沌程度。     

细胞色素C还原态与氧化态的激光共振喇曼谱和红外吸收谱研究

本文用Ar离子激光器的514.5nm作为激发波长,激发细胞色素C还原态(CytochromeC(Ⅱ)吸收谱中对应于β带(～520nm)的π→π~*电子跃迁,测量了它的共振Raman谱及其偏振特性,标识了各个峰所对应的键振动。观察并讨论了产生共振Raman谱的条件。测量在还原态Cytochrome C(Ⅱ)和氧化态Cytochrome C(Ⅲ)混合状态时的Raman谱,得到了     

A time-varying analysis method for rapid transitions in speech

A linear predictive coding (LPC) model based on time-dependent poles which has yielded promising results when applied to synthetic data is applied to real speech data. The data are processed pitch-synchronously using a simple procedure to identify regions of the data that best fit the model. The maximum-likelihood technique, which has been found to be robust in the presence of noise, is used to estimate the parameters. Resulting formant estimates for several diphthongs are presented. The algorithm tracks the formants well, both in stable regions and in regions of transition. This ability to track formant variation within analysis intervals is a definite advantage over traditional LPC. Results from speech data involving final stop consonants are presented. Rapid changes, particularly in the first and second formants, in the region immediately prior to the stop are detected. Such abrupt transitions are often not detected by traditional time-invariant methods     

基于Chebyshev法实现LPC与LSF相互转换

在语音信号的处理中,常需要将线性预测系数(LPC)与线谱对频率参数(LSF)相互转换.利用余弦函数特性改进的Chebyshev多项式求根算法,将余弦函数转换为高次幂函数再进行搜索求根来实现语音特征的LPC参数与LSF参数的转化.     

A formant tracking system toward automatic recognition of speech

This paper describes the implementation of a Speech Understanding System component which tracks the formants of pseudo-syllabic nuclei containing voiced consonants. The nuclei are isolated from continuous speech after a precategorical classification in which feature extraction is carried out by modules organized in a hierarchy of levels. FFT and LPC spectra are the input to the formant tracking system. It works under the control of rules specifying the possible formant evolutions given previously hypothesized phonetic features and produces fuzzy graphs rather than usual formant patterns because formants are not always evident in the spectrogram pattern.     

Algorithm for speech model root location

A novel algorithm for locating the roots of the speech model filter polynomial derived from LPC analysis is proposed. The technique is guaranteed to converge and can be implemented in a fixed time interval. When several iterations are performed, very accurate root estimation can be achieved     

LPC与LSF转换算法的比较研究

线性预测法是语音信号处理中的核心技术。在语音信号的处理中,常常需要将线性预测的LPC系数与LSF参数相互转换。本文根据Chebyshev多项式求根法,研究了几种由LPC求解LSF的算法,分析了它们各自的特点及相互关系。分析并推导了由LSF求解LPC的算法。     

结合牛顿-拉夫森函数计算语音线谱对参数的高效算法

提出了计算语音信号线谱对（LSP）参数的高效算法NRSPF。首先利用牛顿法-拉夫森函数及斯蒂芬森加速求高阶非线性方程的一个实根,再使用多项式综合除法降阶,最后采用费拉里算法求其余的根,即得LSP参数。通过TI-DSP平台的实例研究表明,NRSPF算法与APF算法相比,迭代次数减少、收敛速度加快,计算量小,并且在精度提高10倍、100倍和1000倍情况下,APF算法可能出现被零除错误和死循环,而NRSPF算法不仅避免了该错误,而且迭代次数增加很少,收敛速度仍然很快,得到更精确的结果。本文提出的算法高效、可靠、实时性强,可应用于超低码率语音实时通信系统、语音编解码器等。      

Further Exploring the Strength of Prediction in the Factorization of Soft-Decision Reed–Solomon Decoding

Reed-Solomon (RS) codes are among the most widely utilized error-correcting codes in digital communication and storage systems. Among the decoding algorithms of RS codes, the recently developed Koetter-Vardy (KV) soft-decision decoding algorithm can achieve substantial coding gain, while has a polynomial complexity. One of the major steps of the KV algorithm is the factorization. Each iteration of the factorization mainly consists of root computations over finite fields and polynomial updating. To speed up the factorization step, a fast factorization architecture has been proposed to circumvent the exhaustive-search-based root computation from the second iteration level by using a root-order prediction scheme. Based on this scheme, a partial parallel factorization architecture was proposed to combine the polynomial updating in adjacent iteration levels. However, in both of these architectures, the root computation in the first iteration level is still carried out by exhaustive search, which accounts for a significant part of the overall factorization latency. In this paper, a novel iterative prediction scheme is proposed for the root computation in the first iteration level. The proposed scheme can substantially reduce the latency of the factorization, while only incurs negligible area overhead. Applying this scheme to a (255, 239) RS code, speedups of 36% and 46% can be achieved over the fast factorization and partial parallel factorization architectures, respectively.     

A wideband SAW resonator and its application to a VCO for mobileradio transceivers

A new wideband SAW resonator has been developed for SAW-VCOs used in mobile radio communication transceivers. The resonator is based on a repeated structure of pairs of electrically connected IDTs, which provides a high Q and a wide frequency bandwidth. An experimental SAW-VCO module measuring 8×8 mm² offers a frequency-variation bandwidth of about 20 MHz, a C/N of 75 dB, and an output power of 3-4 dBm at 900 MHz, satisfying the specifications for Japanese MCA and business-use radio transceivers     

谐振腔光纤陀螺信号检测方法的研究

谐振腔光纤陀螺(R—FOG)是利用光学Sagnac效应实现对转动检测的一种高精度的惯性传感器件。在谐振腔光纤陀螺系统中，信号检测系统占有非常重要的地位，其检测精度的大小直接影响陀螺的分辨率。光纤环形谐振腔是谐振腔光纤陀螺的核心敏感部件。采用两种频率的锯齿波组合调制，考虑激光器有限光谱线宽条件下，采用洛仑兹线型描写光纤环形谐振腔的输出光强表达式。针对输出光强与谐振频率偏差在靠近谐振点附近的近似线性关系，利用多次反馈频率操作来依次跟踪谐振腔光纤陀螺顺时针和逆时针光束的谐振点，从而避免了谐振频率偏差复杂的求根算法。仿真结果表明，多次反馈频率操作，可以较快地锁定到谐振点。     

CAD applying the finite-element method for dielectric-resonatorfilters

The progress of numerical techniques now permit us to analyze rigorously complex devices such as dual-mode cavity multipole filters or planar passive elements for coplanar monolithic microwave integrated circuits (MMICs). In this paper, we describe a rigorous design of dielectric resonator (DR) filters applying the finite-element method (FEM). We first present a dual mode coupling technique which replaces classical DRs, coupling, and tuning screws, which are commonly used in dual-mode filters, by slotted DRs. Next, a new theoretical analysis based on the contribution to the dual-mode filter response of the first DR hybrid mode and of higher order modes is described. This analysis can be applied to any type of microwave dual-mode filter. It allows us to define a procedure which explains the presence and controls the position of the two transmission zeros in the filter responses. In this paper, this procedure has been applied to improve filtering performances of a dual-mode DR filter. Finally, a synthesis method is developed to rigorously design for the first time, a four- and an eight-pole slotted DR elliptic filters. The experimental results were obtained with no tuning and the theoretical ones show good agreement