for speech recognition dictionary of Chinese tree, completed 60,000 contribution...

凌阳61做的语音报站，已经通过调试。-do voice stations, has passed testing. Haha

CMU大名鼎鼎的SPHINX－3大词汇量连续语音识别系统，应用于Unix和Windows，做语音识别的必备工具。-CMU famous SPHINX-3 large vocabulary continuous speech recognition system, used in Unix and Windows, so the essential voice recognition tools.

marsyas是进行音频分析的源码，可以识别音乐节奏、乐器等音乐语义特征，使用隐马尔可夫模型。-marsyas is a source of audio analysis to identify the music rhythm, semantic features music and musical instruments, the use of Hidden Markov Model.

语音信号处理中基音周期提取算法。运用传统的自相关方法。并加入了三电平削波的处理。-Speech Signal Processing pitch extraction algorithm. Using the traditional autocorrelation method. And joined the three-level clipping processing.

关于hmm仿真的资料，内容实用有效。希望对大家有所帮助-hmm simulation of the information, as practical and effective. We hope to help

CCITT G.711语音压缩算法 -CCITT G.711 voice compression algorithm

能有效的去除纯净语音中加入的噪声，效果不错。-Can effectively remove the pure voice added to the noise, good results.

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.

这是一个在计算机上用C语言写的ADPCM编解码程序。-This is a computer using C language to write procedures for the ADPCM codec.

voicebox语音处理matlab工具箱-Voicebox:the speech processing toolbox of matlab.