hmm

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.)

这是开源语音合成espeak的源代码，对研究TTS的同学很有帮助。(This source code is open source speech synthesizer espeak, helpful studies TTS classmates.)

这是基于隐马尔可夫模型的连续语音识别代码，不同于dtw,这是个完整的工程，我把需要的voicebox也放进去了，一共用到9个子函数，包括模板的训练算法何识别算法，是完全可以使用的。(This is based on the hidden Markov model of continuous voice recognition code, different the dtw, which is a complete project, I need to put the voicebox into a total of nine sub-functions, including the template training algorithm to identify the algorithm, Is fully available.)

Praat4610 语音合成及分析源代码(Praat4610 speech synthesis and analysis of source code)

c语言写的wav音频文件简单处理函数源码，包括wav_io处理，提取lpc系数、plar系数，可用于语音识别和合成中。可直接调用。(wav audio files simple c language handler source, including wav_io processing, extraction lpc coefficient plar coefficient, can be used for speech recognition and synthesis. Can be called directly. )

语音的识别的前端技术，语音的基音周期检测(Front-end voice recognition technology, Voice Pitch Detection)

基于负熵的盲源分离算法，可以用于语音的盲分离，提高语音信号质量(Negative entropy algorithm based on blind source separation, can be used to blind separation of speech, improve the quality of voice signal)

程序实现的功能是：一种改进型的LMS算法！具有收敛速度快的特点(Procedures for the realization of the functions are: an improved algorithm of LMS! With the characteristics of fast convergence)

经典麦克风语音增强算法，当阵元数目越多，语音增强效果越好(Classical microphone speech enhancement algorithm, the more the number of elements, the better the speech enhancement effect)

基于动态时间规整与隐马尔科夫模型的自动语音识别系统(Automatic speech recognition system based on dynamic time warping and hidden Markov model)