An improved speech recognition VQ code I have written and improve the feeling of...

这是城市坐标的程序. 注释:这是本人自己编写的.不足之处请各位大哥大姐多多指教,另外请注意版权问题.-This the city coordinates procedures. Note : This is my own prepared. Deficiencies Department asked Big Brother Big Sister to the exhibitions, and to copyright issues.

基于CHMM模型的语音识别系统，能很好的实现汉语语音识别功能。

H。263标准的编解码程序代码，仅作参考， 甚用！-H. 263 codec standard code for information only, the very use!

实现 直接2型滤波器的设计 以及 仿真 功能-direct two-filter design and simulation function

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

一个语音识别技术的应用，用做的在职　　茜-A voice recognition technology, used to do the job Qian

The Yin algorithm was developed by Alain de Cheveigné of IRCAM-CNRS and Hideki Kawahara of Wakayama University. It allows for real-time continuous (for each sample) fundamental frequency estimation. It features a very low error rate and few tuning parameters. This implementation is a C++ template, with EMM support for faster processing. Developped in Visual C++ 6.0, but should compile with other compilers.-The Yin algorithm was developed by Alain de Cheveigné of IRCAM-CNRS and Hideki Kawahara of Wakayama University. It allows for real-time continuous (for each sample) fundamental frequency estimation. It features a very low error rate and few tuning parameters. This implementation is a C++ template, with EMM support for faster processing. Developped in Visual C++ 6.0, but should compile with other compilers.

人类精神智慧的精华，在你绝望的时候，在你懈怠的时候，给予你强大的力量-The essence of the wisdom of the human spirit, in your time of despair, before you slack time give you powerful

这是音频编解码源代码G11,G721,G731.-This is the audio codec source code G11, audio, G731.

语音信号的频域处理,语音虽然是一个时变、非平稳的随机过程。但在短时间内可近似看作是平稳的。因此如果能从带噪语音的短时谱中估计出“纯净”语音的短时谱，即可达到语音增强的目的。由于噪声也是随机过程，因此这种估计只能建立在统计模型基础上。利用人耳感知对语音频谱分量的相位不敏感的特性，这类语音增强算法主要针对短时谱的幅度估计。 -voice signals in the frequency domain processing, voice is a time-varying, nonstationary random process. But in a short period of time can be approximated as smooth. So if Noisy Speech from the short-term spectrum estimate "pure" voice of the short-term spectrum, and reached speech enhancement purposes. As the noise is random process, which can only be estimated based on statistical models based on. Use ear perception of voice spectrum component of the phase sensitive to the characteristics of such speech enhancement algorithms targeted at the rate of short-term spectral estimation.