最终代码说话人识别

为均衡带限信号所引起失真的横向或格型自适应均衡器(其中横向FIR系统长M=11), 系统输入是取值为±1的随机序列 ，其均值为零；参考信号 ；信道具有脉冲响应： 式中w用来控制信道的幅度失真(w = 2～4，例如，取w = 2.9,3.1,3.3,3.5等),而且信道受到均值为零、方差为 (例如，取 ，相当于信噪比为30dB)的高斯白噪声 的干扰。试比较基于下列五种算法自适应均衡器在不同信道失真、不同噪声干扰下的收敛情况(对应于每一种情况，在同一坐标下画出其学习曲线)： 横向/格-梯型结构LMS算法[1-4] 横向/格-梯型结构RLS算法[1-4] ()

自制DMX512控制台,很不错的一个软件,他对灯光的控制,非常出色.(Homemade DMX512 console, a very good one software, he lighting control, very good.)

基于最小均方误差的助听器语音去噪算法, 用于以最大程度消除助听器接受的语音杂音.(Speech denoising algorithm based on the minimum mean square error of a hearing aid for the hearing aid acceptable voice murmur eliminated to the greatest extent.)

声源定位程序，语音信号与房间脉冲响应进行卷积得到混响语音信号，再通过互相关时延估计得到定位(Sound source localization procedure, the speech signal and the room impulse response are convoluted to get the reverberation speech signal, and then through the correlation time delay estimation to obtain the localization)

通过语音分帧、端点检测、pitch提取、DTW算法，实现歌曲识别(Through voice framing, endpoint detection, pit extraction, DTW algorithm, song recognition is realized)

本程序音频共有四个，借助鸡尾酒会问题思想，用ica算法实现四个音频的分离(The audio program for a total of four, with the help of the cocktail party problem, realizing the separation of four audio using ICA algorithm)

基于LPC的语音合成，可以实现变调变速功能，附带注释便于学习。(LPC-based speech synthesis, can achieve modulation speed function, annotated easy to learn.)

LMS算法多麦克风语音降噪，语音降噪处理，基于matlab(LMS algorithm for multi-microphone speech noise reduction, voice, noise reduction, based on matlab)

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.)

给出了通信系统中语音信号的MATLAB仿真文档(Given speech signal communication system documentation MATLAB simulation)