WienerScalart

LMS自适应滤波器的matlab源码 文件大小不够，所以加了个图(LMS adaptive filter matlab source code file size not enough, so the addition of a map)

基于单通道的在线次级通道建模的主动 噪声控制FXLMS算法的MATLAB实现( a single channel feed-forward active noise control system based on On-line secondary path modellingthe FxLMS alogrithms You can find many information about this simulation in "Use of random noise for online transducer estimate in an adaptive attenuation system " written by L,J Eriksson and M.A Allie in 1989.)

语音信号的录音、读入、放音等：利用函数wavread对语音信号进行采样,记住采样频率和采样点数，给出以下语音的波形图（2.wav），wavread的用法参见mablab帮助文件。利用wavplay或soundview放音。也可以利用wavrecord自己录制一段语音，并进行以上操作(需要话筒)。 (Speech signal recording, reading, playback, etc.: the voice signal using the function wavread sampling, sampling frequency and sampling points to remember, given the following speech waveform (2.wav), wavread usage see mablab help file. Using wavplay or soundview playback. Can also record a voice using wavrecord own, and the above operation (requires microphone).)

C语音实现Matlab corrcoef（求两个信号的互相关系数），包括了Matlab的对比验证源代码(Calculate the correlation coefficient refer the Matlab function:corrcoef)

GMM说话人识别平台全套，从特征提取到模式识别，到识别率，一整套都有(GMM speaker recognition platform full, from feature extraction to pattern recognition, the recognition rate, set both)

基于LMS算法和NLMS算法的自适应声回波消除器AEC，实现的主要功能是：在用户对电视发出的指令混杂了电视扬声器的声学回声的情况下，滤掉电视声音得到较为纯净的用户语音指令信号。(The main function of the adaptive filter based on the LMS algorithm and the NLMS algorithm is mixed: in the instruction issued by the user on the television TV speakers acoustic echo case, the TV sound was removed by filtration to obtain relatively pure user voice command signal.)

本次课程设计主要介绍了语音信号的录制、语音信号的采集与分析、语音信号的采样、语音信号的频谱分析、加噪后的频谱分析、滤波器的设计以及对于语音信号的回放等知识。通过 Cool Edit录制一段语音信号，运用MATLAB对其进行时域与频域分析，然后设计滤波器加入噪声进行滤波处理，比较滤波前后始于波形或频谱的变化，最后实现对于语音的回放设计。(Digital voice processing filter design)

压缩包中包括3个用于对语音增强中，增强语音质量测试的程序：分段信噪比提高测量、对数谱距离测量、信噪比测量。具体说明，请看文件文件说明。(Compression package, including three for speech enhancement, the enhanced voice quality testing procedures: Sub-noise ratio improve measurement of the distance spectrum measurement, signal to noise ratio measurement. Specify, see the paper documentation.)

说话人识别系统,界面友好，可以对实时录音的说话人进行识别(Speaker recognition system, user-friendly, real-time recording can be carried out to identify the speaker)

基于负熵的盲源分离算法，可以用于语音的盲分离，提高语音信号质量(Negative entropy algorithm based on blind source separation, can be used to blind separation of speech, improve the quality of voice signal)