matlab

双极性归零码的功率谱密度(Bipolar power spectral density of NRZ)

Kriging插值工具箱，可做可历尽内插(Kriging interpolation toolbox, can be experienced all kinds of interpolation)

matlab语音信号处理工具箱 本工具箱可配合语音数据库使用，用于计算线性预测语音模型的参数，语音声调转换，语音自动解析分解，语音语速变换及更改发音的重音，音量，清晰度等。配套图书《Speech Processing and Synthesis Toolboxes》已经出版，翻译进行中 (matlab signal processing toolbox with the voice of the Toolbox database can be used for calculating linear predictive speech model parameters, voice tone conversion, automatic speech analysis decomposition, transformation and change of voice speech rate accented pronunciation, volume, clarity, etc. . Matching book " Speech Processing and Synthesis Toolboxes" has been published in translations)

说明：  雷利对流，用格子波尔兹曼方法以Z.Guo程序为基准，得出的近似值用于估计浮动过程，希望有人能够用得到。( Rayleigh Benard Convection, using a LB method,based on [Z.Guo, ]. Boussinesq approximation is used for the buoyancy term.)

关于半脆弱水印的几篇文章，其中包括小波变换的应用以及contourlet变换的应用等，涉及领域为半脆弱数字图像水印已经篡改检测。(Semi-fragile watermarking of several articles, including the application of wavelet transform and Contourlet transform application areas for semi-fragile digital image watermarking has tamper detection.)

蚁群算法很好的一个MATLAB程序资料，希望大家支持(ACO)

Least mean squares (LMS) algorithms is a type of adaptive filter used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean squares of the error signal (difference between the desired and the actual signal)

Sparse Signal Representation using Overlapping Frames (matlab toolbox)(Sparse Signal Representation using Overl apping Frames (Matlab toolbox))

矢量水声信号的matlab仿真程序，里面包含文字说明与程序代码(Vector underwater acoustic signal matlab simulation program, which contains the text description of the program code)

关于端点检测的几种方法，语音样本是自己录制的，对传统算法做了一些改进，加入了去噪，去噪之后再进行端点检测，均调通 vad0303：自己设置调整门限为一定值 vad0310：根据能量值和过零率设置门限，自适应门限值 vad0310_2：基于比例因子的门限自调整 vad0310.m加入了噪声，端点检测前都噪声进行了处理 entropy.m：基于自适应子带频谱熵的稳健性语音端点检测 可用于语音增强及端点检测 dbdoor.m：双门限算法，用于语音端点检测。可以通过调整门限值，并加入门限自适应算法，实现语音端点检测(Several methods on endpoint detection, recording their own voice samples, the traditional algorithm has made some improvements, adding a de-noising, and then after de-noising endpoint detection are transferred through)