Turboequlization_algorithm

This file contains and display nancy s articulated humanoid body in anatomical static posture. Body segments can be accessed independently they are named with suffix: "_point", polygons are also independent with suffix: "_polygons". In addition, the variables "point" and "polygons" include the whole points and polygons respectively with the accumulative index in the case of polygons. After running the file, do a "whos" to see the actual names of body segments.

MUSIC算法[1]是一种基于矩阵特征空间分解的方法。从几何角度讲，信号处理的观测空间可以分解为信号子空间和噪声子空间，显然这两个空间是正交的。信号子空间由阵列接收到的数据协方差矩阵中与信号对应的特征向量组成，噪声子空间则由协方差矩阵中所有最小特征值（噪声方差）对应的特征向量组成。(MUSIC algorithm [1] is a feature space based on matrix decomposition method. From the geometric point of view, the signal processing can be decomposed observation space the signal subspace and the noise subspace, it is clear that the two spaces are orthogonal. Signal Subspace data received by the array covariance matrix and eigenvectors corresponding to the signal component, the noise subspace from the covariance matrix of all the smallest eigenvalue (noise variance) eigenvector components.)

ybus case distributed-generation_distribution-system_Newton-Laphson

说明：  一种带有梯度加速的粒子群算法，可以实现多种优化工作的需要哦。(With a gradient to accelerate the particle swarm algorithm, can achieve the needs of a variety of optimization Oh.)

特征提取SIFT算法，对放缩 旋转不变性使此算法的经典之处。模式识别的必会算法(SIFT feature extraction algorithm, scaling rotation invariance to the classic between the algorithm. Pattern recognition algorithm will be)

关于TSP模拟退火&遗传算法&相应的MATLAB算法程序( tsp Genetic algorithm and simulated annealing and corresponding MATLAB algorithm procedures)

说明：  由混合光学双稳模型生成混沌序列的matlab程序(By mix optics bistable model production chaos sequence )

关于matlab编写的语音端点检测的程序(Matlab prepared speech endpoint detection procedures)

摘 要：本论文主要研究了语音识别的基本原理,对语音识别系统的构成进行分析处理,其中包括预处理、特征参数提取、建立模块库、识别匹配几大部分。预处理又包括语音采样、预加重、加窗（汉明窗）、端点检测；特征提取的参数是梅尔频率倒谱系数MFCC。 该语音系统采用的是动态时间伸缩算法(DTW)，研究对象是特定人的语音识别，并在MATLAB平台上实现。为了进行后续研究，首先使用电脑中的录音系统录制了阿拉伯数字0—9的语音文件，并转化成 “.wav”格式的文件。(Abstract: This thesis mainly studied the basic principle of speech recognition, to analyze the composition of the speech recognition system, including the preprocessing, feature extraction, to set up the module library, identify several most matches. Pretreatment, including speech sampling, pre-emphasis, add window (hamming window), endpoint detection Feature extraction of MFCC MEL frequency cepstrum coefficient. The voice system USES a dynamic time scale (DTW) algorithm, the research object is the speaker-dependent speech recognition, and realized in MATLAB platform.To carry out the follow-up study, the first to use the recording in a computer system to record the audio files of Arabic Numbers 0-9, and translated into . Wav format file.)

会跳动的红心，比较可爱版，可以编译上自己的名字(Beating hearts, lovely versions, can compile their own names)