语音识别与合成入门5，word或压缩包中内有matlab源码

dtw（动态时间规整）语音识别的matlab源码-dtw (Dynamic Time Warping) speech recognition matlab source code

语音编码的原程序,是在国外的网站下来的,没有改动可以直接在软电话下使用!-speech coding the original procedure is outside the site down, no changes may be directly in the use of soft phone!

自编的语音语谱图程序，适用于语音信号处理以及语音增强处理，-self Speech Spectrogram procedures applicable to the speech signal processing and speech enhancement.

编译环境：vc++.net2003 名称：语音合成软件 作者：yxifu-Program environment: VC++.net2003 Name: program of speech synthesize Programer: yxifu

提供信号处理中的Rasta变换的实现过程，并可以给出相应参数。开发环境为VC。- Provides in the signal processing the Rasta transformation realization process, and may produce the corresponding parameter. The development environment is VC.

LZW压缩解压的完整程序，不可多得，连输入输出参数都实现了。可立即下在编译-LZW compression decompression integrity procedures, rare, and even input and output parameters are realized. Immediately under the compiler

运用SAPI将文本转化成语音的demo-use SAPI text into voice demo

用labview实现的基于mfcc的说话人识别，测试的结果不是太理想，但体现的思想应该对labview的学习有所帮助-Using LabVIEW realize the MFCC-based speaker recognition, the test results are not too ideal, but should embody the idea of learning LabVIEW help

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.