yuyin

本资料是自己编写的基于MATLAB语音工具箱的DTW动态时间规整的说话人识别，具有很高的识别率(This information is written in their own voice-based MATLAB toolbox DTW dynamic time warping speaker recognition with high recognition rate)

（1）理解矢量量化的基本原理。 （2）掌握矢量量化器设计的LBG方法。 （3）运用LBG方法获取语音短时平均能量码书((1) to understand the basic principles of vector quantization. (2) master the vector quantizer design LBG method. (3) Using the LBG method to obtain the speech short time average energy code book)

语音通信控件源码点对点专用版简介VSession语音通信控件源码点对点专用版，版本2.0，此版本集成了G729A压缩算法，实时传输协议，话音清晰流畅！使用简单易懂方便！此控件源码是在本人以前发布的控件件源码VSessionn2.0版本的基础上，加入点对点通信时的呼叫，应答，挂断等通话前后的同步功能，使其用于点对点通信更加方便！如果您想要在程序源码中自定义点对点间的通话联系方式，或者有一对多，多对多的 (Voice communications control source peer-to-peer special edition Introduction VSession voice communication control source peer-to-peer special edition, version 2.0, this version integration the G729A compression algorithm, Real-time Transport Protocol, clear and smooth voice! Easy to understand and easy to use! This control source is in the I control previously released the parts source VSessionn2.0 version based on peer-to-peer communication call, answer, hang up, etc. the synchronization function before and after the call to make it more convenient for point-to-point communication ! If you want to program source from the definition of point-to-point calls between Contact, or one-to-many, many-to-many)

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] ()

完整的基于HMM的语音识别系统，时间序列数据分析中的梅林变换工具，最终的权值矩阵就是滤波器的系数。( Complete HMM-based speech recognition system, Time series data analysis Mellin transform tool, The final weight matrix is ??the filter coefficient.)

3、语音特征提取与分类 首先, 待识别语音转化为电信号后输入识别系统, 经过预处理后用数学方法提取语音特征信号, 提取出的语音特征信号可以看成该段语音的模式。(3, the voice feature extraction and classification First, to be recognized voice into electrical signals input recognition system, after pretreatment with a mathematical method to extract speech characteristic signal extracted speech characteristic signal can be seen as the voice segment model.)

LMS多麦克风语音降噪的主程序是lmsspdn.m(Multi-microphone noise reduction LMS voice is the main program lmsspdn.m)

单点定位的滤波程序，可以获得速度信息和位置信息,这是一个滤波程序，对语音信号滤波。(A single point positioning of the filtering process, get the speed of information and location information,this is a wiener filtering process to filter the speech signal.)

自己提取的讯飞输入法语音识别组件，可绿色运行，解压双击iFlyVoice.exe即可运行，语音识别能力比较强！(The voice recognition component extracted by Xunfei can be operated in green, decompressed and double-clicked iFlyVoice.exe can be operated, and the voice recognition ability is relatively strong!)

比较经典的用lms算法实现自适应均衡器的matlab的程序，适合初学者学习(Comparison of the classical LMS algorithm with adaptive equalizer matlab procedures, suitable for beginners to learn)