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1.
This paper focuses on the identification of multiple-input single-output output-error systems with unknown time-delays. Since the time-delays are unknown, an identification model with a high dimensional and sparse parameter vector is derived based on overparameterization. Traditional identification methods cannot get sparse solutions and require a large number of observations unless the time-delays are predetermined. Inspired by the sparse optimization and the greedy algorithms, an auxiliary model based orthogonal matching pursuit iterative (AM-OMPI) algorithm is proposed by using the orthogonal matching pursuit, and then based on the gradient search, an auxiliary model based gradient pursuit iterative algorithm is proposed, which is computationally more efficient than the AM-OMPI algorithm. The proposed methods can simultaneously estimate the parameters and time-delays from a small number of sampled data. A simulation example is used to illustrate the effectiveness of the proposed algorithms.  相似文献   

2.
张晓辉 《科教文汇》2011,(36):201-202
学校的多媒体电教室、会议室、报告厅、舞台演出都普遍使用了调音台、无线话筒等电子设备,但对激励器这一专业音响器材应用得还较少,提起未免有些陌生。激励器是现代电子技术和心理声学相结合的产物,它能营造出各种音乐信号的谐波,能够对放大器等各种电子设备对在音频信号进行处理过程中失落的细节(主要指失去的谐波和泛音)进行修复。这些看似微不足道的细节恰恰反映了音频信号中所表达的特殊的情感,给人以耳目一新的感觉,有着不容忽视的感染力和艺术效果。  相似文献   

3.
The problem of designing robust systems for the detection of stochastic signals in noise is considered for the large-sample-size, small-signal case. By applying two previously-established models for the detection of stochastic signals, known results for the robust detection of deterministic signals are extended on a limited basis to the stochastic- signal case. The proposed detectors are seen to be robust over a class of possible noise statistics, based on a Huber-Tukey mixture model, which contains noises characterized by heavy-tailed probability density functions. In addition, numerical results are presented which verify the robustness property of the proposed detectors over wider classes of noise mixtures.  相似文献   

4.
本文针对判断多达64路音频信号传输的信号通道故障及故障发生时间的定位和报警,设计出了一种基于AD536A真均方值转换芯片的多通道音频信号监测器。该监测器通过LED显示器和声音进行报警实现了对多通道音频信号进行准确检测,故障频道和故障时间定位,达到实时、高精度监测的目的。  相似文献   

5.
面向信息检索的汉语同义词自动识别和挖掘   总被引:3,自引:0,他引:3  
为了提高同义词自动挖掘的效率,本文提出了从词典释义中自动识别和挖掘同义词的方法,使用超链接分析算法和模式匹配算法,从不同的角度提取同义词:第一部分是把词汇之间注释与被注释的关系看成是一种链接关系。对给定的词汇进行分析,把与给定词汇具有链接关系的所有相关词汇构造一个词汇图,图中的每一个节点代表相关词,每条弧代表了词汇之间注释与被注释的关系。利用超链接分析方法并结合PageRank算法,计算词汇的PageRank值,把PageRank值看成是体现词汇之间语义相似性的衡量指标,最后为每一个词汇生成候选同义词集,并通过一定的筛选原则和方法,推荐出最佳的同义词。第二部分是利用词汇定义模式,对词汇的释义方式进行分析,归纳总结出在词典释义中同义词出现的模式,进而利用模式匹配方法识别和挖掘同义词。此外,利用模式匹配方法对Web网页和期刊论文中的同义词也进行了挖掘测试。测试结果表明,利用模式匹配和超链接分析方法来自动识别和挖掘同义词具有可行性和实用性。  相似文献   

6.
This study proposes a probabilistic model for automatically extracting English noun phrases without part-of-speech tagging or any syntactic analysis. The technique is based on a Markov model, whose initial parameters are estimated by a phrase lookup program with a phrase dictionary, then optimized by a set of maximum entropy (ME) parameters for a set of morphological features. Using the Viterbi algorithm with the trained Markov model, the program can dynamically extract noun phrases from input text. Experiments show that this technique is of comparable effectiveness with the best existing techniques.  相似文献   

7.
This paper focuses on the parameter estimation for radial basis function-based state-dependent autoregressive models with moving average noises (RBF-ARMA models). An extended projection algorithm is derived based on the negative gradient search. In order to reduce the sensitivity of the algorithm to noise and reduce the fluctuations of the parameter estimation errors, a modified extended stochastic gradient algorithm is proposed. By introducing a moving data window, a modified moving data window-based extended stochastic gradient algorithm is further developed to improve the parameter estimation accuracy. The simulation results show that the proposed algorithms can effectively estimate the parameters of the RBF-ARMA models.  相似文献   

8.
This paper considers the parameter and order estimation for multiple-input single-output nonlinear systems. Since the orders of the system are unknown, a high-dimensional identification model and a sparse parameter vector are established to include all the valid inputs and basic parameters. Applying the data filtering technique, the input-output data are filtered and the original identification model with autoregressive noise is changed into the identification model with white noise. Based on the compressed sensing recovery theory, a data filtering-based orthogonal matching pursuit algorithm is presented for estimating the system parameters and the orders. The presented method can obtain highly accurate estimates from a small number of measurements by finding the highest absolute inner product. The simulation results confirm that the proposed algorithm is effective for recovering the model of the multiple-input single-output Hammerstein finite impulse response systems.  相似文献   

9.
以VC++6.0为开发平台,实现一个基于隐马尔可夫模型(Hidden Markov Model,简称HMM)非特定人的安多藏语孤立词语音识别系统。对有声段语音进行MFCC参数的提取,对提取后的MFCC参数进行矢量量化后训练HMM模型,形成特征模板库,最后进行识别。根据安多藏语的特点,改进端点检测的方法,提高了孤立词语音信号检测的准确性,并进一步提高了识别率。  相似文献   

10.
In almost all the work carried out in the area of automatic modulation classification (AMC), the dictionary of all possible modulations that can occur is assumed to be fixed and given. In this paper, we consider the problem of discovering the unknown digital amplitude-phase modulations when the dictionary is not given. A deconvolution based framework is proposed to estimate the distribution of the transmitted symbols, which completely characterizes the underlying signal constellation. The method involves computation of the empirical characteristic function (ECF) from the received signal samples, and employing constrained least squares (CLS) filtering in the frequency domain to reveal the unknown symbol set. The decoding of the received signals can then be carried out based on the estimate of the signal constellation. The proposed method can be implemented efficiently using fast Fourier transform (FFT) algorithms. In addition, we show that the distribution estimate of the transmitted symbols can be refined if the signal constellation is known to satisfy certain symmetry and independence properties.  相似文献   

11.
系统以16位数字信号处理器TMS320VC5502为核心,采用音频Codec芯片TLV320AIC23对语音信号进行采集和编码转换,通过端点检测、特征参数提取、DTW算法等关键技术实现特定人、小词汇量、孤立词的语音识别,最终根据LED闪灯次数检测数字0~9的识别结果。  相似文献   

12.
采用基于词典的正向增字最大匹配算法,分词词典采用改进的双层哈希表加动态数组的数据结构。在不提升已有典型词典机制空间复杂度与维护复杂度的情况下,一定程度上提高了中文分词的速度和效率。  相似文献   

13.
This paper focuses on the joint parameter and state estimation issue for observer canonical state-space systems with white noises in state equations and moving average noises in output equations. By means of the Kalman filtering and the gradient search, we derive a Kalman filtering based extended stochastic gradient algorithm. For purpose of achieving the higher parameter estimation accuracy, a Kalman filtering based multi-innovation extended stochastic gradient algorithm is proposed on the basis of the multi-innovation identification theory. Finally, the effectiveness of the proposed algorithms is validated through a numerical example.  相似文献   

14.
《Journal of The Franklin Institute》2022,359(17):10145-10171
Considering the colored noises from the process environments, the parameter estimation problems for the feedback nonlinear equation-error systems interfered by moving average noises are addressed in this paper. Due to small computational burden, the gradient search principle is adopted to the feedback nonlinear systems and an overall extended stochastic gradient algorithm is derived for parameter estimation. Introducing the innovation length, the scalar innovation is expanded into the innovation vector and a multi-innovation extended stochastic gradient algorithm is further developed to reach the high estimation accuracy by utilizing more dynamical observed data. Furthermore, to assure the convergence of the proposed algorithms, their convergence properties are analyzed through the stochastic process theory. Finally, the experimental results indicate the effectiveness of the proposed algorithms.  相似文献   

15.
This paper presents a novel switching predefined-time parameter identification algorithm with a relaxed excitation condition based on the dynamic regressor extension and mixing (DREM) method. DREM often requires the persistent excitation (PE) of the extended square regressor's determinant to ensure exponential parameter convergence. Unlike the classical DREM method, a new parameter identification algorithm configured with a two-layer filter technique is proposed under a relaxed initial excitation (IE) condition, rather than strict PE. A key point in choosing IE instead of PE is the introduction of a smooth switching function that dominates the pure integral action and filter behavior of the extended square regressor. The proposed algorithm relies on the predefined-time stability theorem and the settling-time of the identification algorithm is set a priori as a system parameter. The contributions of this paper are a novel switching predefined-time parameter estimation algorithm that 1) relaxes the stringent PE condition, 2) achieves predefined-time convergence, and 3) guarantees the monotonicity of each element of the parameter error inherited from the classical DREM method. Comparative simulation results are presented to illustrate the effectiveness of the proposed algorithm.  相似文献   

16.
This paper is a study of models of spaces of duration-limited signals considered as set functions. Models for the space of bounded signals, signals of bounded area and signals of finite energy are considered. It is shown that by using set function models and a Stieltjes integral for set functions one can derive integral representations for all bounded linear operators on each of these spaces. It is also shown that impulse functions can be treated rigorously as signals of bounded area in these models without introducing singularities, or requiring impulses to be considered as functionals on a signal space rather than as signals. Finally, analogues to differentiation and indefinite integration are introduced. The applications of these models to physical problems are illustrated by reformulating lumped parameter electrical network theory in a set function formalism, and approximate solutions to the resulting equations are shown.  相似文献   

17.
语音信号处理过程包括对需处理的音频信息进行过滤、解析和音频特征提取三个步骤,本文主要针对音频特征中的音调提取算法进行改进。  相似文献   

18.
A nearest neighbour search procedure is described for the automatic correction of misspellings. The procedure involves the replacement of a misspelt word by that word in a dictionary which best matches the misspelling, the degree of match being calculated using a similarity coefficient based on the number of trigrams common to the two words. Experiments with a collection of 1544 misspellings and a dictionary of 64,636 words suggest that the procedure results in the unique identification of the correct spelling for over 75% of the misspellings if the correct form of the word is in the dictionary, and that this figure may be increased to over 90% if near, rather than nearest, neighbours are acceptable.  相似文献   

19.
提出了一种基于经验模式分解去斑和顶帽变换背景不均匀的预处理方法. 经验模式分解去斑算法先对图像每一列进行经验模式分解得到IMF函数,然后将原信号与第一、二模态相减得到初步处理图像,再对该图像每一行重复该操作从而得到去斑图像,该算法有效地去除斑点噪声;顶帽变换则有效地补偿了海浪带来的局部不均匀的背景亮度,提高了图像的信杂比,有利于目标的检测. 仿真结果证明了算法的有效性.  相似文献   

20.
向婷婷 《科技广场》2014,(1):99-102
音乐是由音符、节拍组成的。每个音符有固定的频率。若单片机定时器产生与音符相同的频率,同时采用延控制节拍,就能播放音乐。利用AT89C51单片机的P1.0、P1.1引脚同时输出音频信号,产生和弦音乐。  相似文献   

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