ADPCM压缩解压编码，实现部分为C语言编写，把它用C++封装为两个接口，编码和解码，方便使用。8K 16bit。...

SoundRuler是一个用来分析和图形化重复声音的工具(生物声学)。它识别和测量文件中每种声音的时间和光谱属性。执行的结果可以保存到电子数据表，也可以导出波形图。-SoundRuler is uses for to analyze and the graph repetition sound tool (biological acoustics). It distinguishes and surveys in the document each kind of sound time and the spectrum attribute. The execution result may preserve the electronic data sheet, also may derive the oscillogram.

共振峰合成非常好用-A very convenient project on formant synthesis.

A wonderful ica toollbox about blind source separation for the beginners

基于g.723.1的音频压缩程序，ITU的源码-The voice compress program file which base on G.723.1 is own to the original source code from ITU.

语音的正弦模型程序，用户接口比较全。包括正弦模型的分析合成，模块比较多，可以参考网上的说明。-voice sinusoidal model procedures, the entire user interface comparison. Including the sinusoidal model of synthesis, more modules, the Internet can refer to the note.

MATLAB编写的用于语音识别的程序 主要内容为隐马尔可夫程序-MATLAB prepared for the speech recognition process mainly as Hidden Markov proce

LMS 算法源程序，绘制性能曲线图，误差性能曲线-LMS algorithm source code, rendering the performance curves, the error performance curve

cdma2000 1x语音编解码程序，并带有测试用例，很好用-cdma2000 1x voice codec procedures and with test cases, very good use

Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预 测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。 基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音） 的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲 序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in linear predictive coding LPC theory, lattice filters, as well as the current prediction equation solving algorithm, can achieve an important element of the speech signal analysis, synthesis or recognition. Based on the existing experimental platform, we can use Matlab function to obtain the number of fixed-voice elements (such as vowels) model coefficients, LPC coefficients are the composition of IIR filters. Shock pulse sequence used as input, we can get the original voice. This is a simple voice synthesis.

本文将现实中采集到的一段语音加上了高斯白噪声，利用了LMS算法，将噪声滤除，仿真效果良好-This article will be collected in reality a section of voice coupled with a Gaussian white noise, using the LMS algorithm, the noise filtering, simulation results