bpsk_baseband

kaiser window for filter matlab code with example

一篇经典论文的代码，“Saliency,Scale,image description”，用于寻找视觉显著性的区域，matlab代码(A classic papers of the code, " Saliency, Scale, image description" , for the visual search of the area significantly, matlab code)

说明：  频域的模态参数识别方法之一，可以对振动信号进行分析处理，得出结构的固有频率，阻尼比，振型系数，方法很好，程序完整，(Frequency-domain modal parameter identification methods that can analyze the vibration signal processing, draw the structure of natural frequencies, damping ratio, mode shape coefficient, very good method, program integrity,)

MLE Linear regression code is an important tool used in pattern recognition

SIFT特征和正交DLT算法在物体识别中的应用.(SIFT features and orthogonal DLT algorithm in object recognition.)

Using a progressive intuitive/mathematical approach, this introduction to computer vision provides necessary theory and examples for practitioners who work in fields where significant information must be extracted automatically from images-- including those interested in multimedia, art and design, geographic information systems, and image databases, in addition to the traditional areas of automation, image science, medical imaging, remote sensing and computer cartography. The book provides a basic (Computer Vision- Linda Shapiro.pdf)

本程序可以简单地求出两个信号之间的相位差，操作简单(This procedure can be simply the phase difference between the two signals is obtained, simple operation)

说明：  控制系统中如何实现分数阶控制器MATLAB例程(Fractional order controller MATLAB routine)

N4ITK Bias Correction. this code is usefull for MRI bias Field removing.

实现维纳滤波用于语音增强。第一步采用频域谱函数计算最佳的维纳滤波器，再对输入带噪信号进行滤波；第二步是在此基础上的改进，用了自适应的思想，及时根据当前帧的状态更新修正滤波器的系统函数，得到更精确的无噪语音信号。第三步是采用自适应LMS方法，在时域迭代计算滤波器系数，迭代次数足够多时可得到较准确的有用信号。(Implement wiener filtering used in speech enhancement. First step optimum wiener filter is obtained by using the frequency spectrum function, again to input the signal with noise filtering The second step is the improvement on the basis of this, with the idea of adaptive, and in a timely manner according to the current frame updates correction system function of the filter, get a more accurate without noise speech signal. The third step is to use the adaptive LMS method, iterative calculation in time domain filter coefficient, the number of iterations enough to get the useful signal accurately.)