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自适应控制，自校正控制系统参数估计matlab程序，利用递推最小二乘法进行系统参数估计(Adaptive control, self-tuning control system parameters estimated matlab procedures, the use of recursive least squares method for system parameter estimation)

implement the Power Al- gorithm for the computation of themaximumeigenvalue of Aj . Write a function Power_Algorithm which takes as input A q (the initial approximation of the eigenvector) and scale (it was given as  in the lectures).

ER,WS和BA网络模型生成算法的matlab实现(ER, WS and BA network model generation algorithm of matlab )

四进制PSK信号的产生，方便大家进行相应信号的的调制实验。(Quaternary PSK signal, facilitate everybody carries on the corresponding signal modulation experiment. )

介绍了matlab-gui设计的基础知识，对于刚刚接触matlab-gui设计的人来说是一份不错的资料(Matlab-gui design introduces the basics, just contact matlab-gui for those who design information is a good)

横向拉应力和平面剪应力的损伤模型umat(Damage Model umat plane transverse tensile stress and shear stress)

一个分析3GP文件格式的程序，可以帮助大家了解3GP文件的格式(An analysis of 3GP file format of the program that can help us to know the 3GP file format)

以输入和输出信号的统计特性的估计为依据，采取特定算法自动地调整滤波器系数，使其达到最佳滤波特性的一种算法或装置。自适应滤波器可 自适应滤波器 以是连续域的或是离散域的。离散域自适应滤波器由一组抽头延迟线、可变加权系数和自动调整系数的机构组成。(The input and output signals of the statistical properties of the estimators for the basis, take the specific algorithm automatically adjusts the filter coefficients, so as to achieve the optimal filter characteristics of an algorithm or device. Adaptive filter adaptive filterIs continuous or discrete domain domain. Discrete domain adaptive filter is composed of a group of tapped delay line, a variable weighted coefficient and automatically adjust the coefficients of the mechanism.)

实现功能 1) 在用户界面设置录音时长并实时录音 2) 调节放音幅度、播放原声并进行时域、频域绘图 3) 对收集的声音进行倒放处理并播放 4) 对收集声音进行回声处理并播放，比对时域波形 5) 对收集声音进行回响处理并播放，比对时域波形 6) 对经过回声处理的声音进行回音消除并比对时域波形 7) 采用自适应滤波器消除采集声音中的环境噪音 8) 设计相应的滤波器组成均衡器对收集的声音进行处理并播放 9) 男女声识别 10) 设计参数可调的滤波器对采集的声音进行处理 (1) in the user interface to set the recording length and real-time recording 2) to adjust sound, played acoustic amplitude and time domain, frequency domain mapping 3) to upend the collection of sound and play 4) to collect sound echo processing and playback, than in the time domain waveform 5) to collect sound echoes, processing and playback, than in the time domain waveform 6) to echo and treated with echo s voice than in the time domain waveform 7) using adaptive filter to eliminate ambient noise in acquisition voice 8) to design the corresponding filter to process the voice of equalizer to collect and display 9) NanNvSheng recognition 10) tunable filter design parameters for voice processing )

SMO算法的m文件以及用svr测试该算法的m文件，以及smo算法文档详细介绍及使用说明(The SMO algorithm m file and test the algorithm using SVR m files, as well as the SMO algorithm document details and instructions)