CMU sphinx，是美国卡内基梅隆开发的开源语音识别系统，该程序用C编写，模块结构非常清晰，函数接口明了，非常适合用来进行二次开发。...

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.

C50_melp.tar.gz为美军2400bps语音压缩编码算法，该压缩文件是算法在TI的TMS320V5X芯片上的实时实现算法。-C50_melp.tar.gz for U.S. 2400 voice compression encoding algorithm, the document compression algorithm is TMS320V5X in TI"s chip real-time algorithm.

本文件包含几个进行语音端点检测的小程序。运行时直接复制到matlab的m文件即可。经本人调试正确无误。-This document contains some of the voice activity detection applet. Directly copied to the run-time of the m file to matlab. After I debug correct.

matlab 求线性倒谱系数,用于语音识别-matlab linear cepstral coefficients for speech recognition

摘要说话人识别在自动身份鉴别方面具有重要的现实意义。文章在分析了说话人识别实现的可行性基础上*提 出采用长时线性预测倒谱 +,--.,)系数和矢量量化 /0)模型的不依赖于文本的说话人识别方法，并成功地应用于 一个数据库查询系统中。-65DL5=?9?AD=7 D66@?AD7?8=H?=D 78:D7?A?=E?G?E D@=?7?8=D

G711语音压缩源码-G711 voice compression FOSS

GSM语音压缩编码源代码] GSM语音压缩编码源代码-GSM voice compression coding source code

端点检测程序 多频带，熵 谱熵 teger能量 希望大家好好学习好好利用-endpoint detection procedures multiband, entropy spectral entropy teger energy hope that we learn to make good use

语音识别，能量计算，过零率检测，发生点检测，用于音符切分-Speech recognition, energy calculation, zero-crossing rate detection, point detection occurs for note segmentation

Epos, a freely distributable language independent Text-to-Speech system.