自己编写的用于传感器信号滤波的算法程序，并有真实信号处理的数据演示...

NIOS环境下的SD―CARD读写，模块-NIOS environment SD-CARD literacy, modules

这个是实时采集语音信号，以及小波消噪的运行程序，我用过的，还算可以，但是小波消噪的那个程序有点问题，大家如果运行后，发现问题出在什么地方，谢谢告诉我。-This is a real-time voice signal acquisition and wavelet denoising operating procedures, I used that figure, Wavelet denoising but that the process is flawed, and if we run, it is found that the problem is in place, Thank you to tell me.

实现了精简的FFT语音压缩 采取了一种新的算法有一定的参考价值-Implementation of speech compress used by simplified FFT algorithm.It is a new one,and useful to you.

txt2wav.rar文本语音（Text-to-Speech，以下简称TTS），它的作用就是把通过TTS引擎把文本转化为语音输出。代码示范了如何运用Microsoft Speech SDK 建立自己的文本语音转换应用程序。应该事先下载微软的语音SDK里面含微软的语音开发工具原来是sapi5.1现在好像升级了微软的MSDN上有下-txt2wav.rar text pronunciation (Text-to-Speech, hereafter refers to as TTS), its function is transforms through the TTS engine the text as the pronunciation output. How did the code demonstrate has established own using Microsoft Speech SDK text pronunciation transformation application procedure. Should beforehand download Microsoft inside pronunciation SDK to contain Microsoft the pronunciation development kit originally is sapi5.1 now looks like promotes on Microsoft s MSDN to have next

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

matlab中wave文件的播放功能，欢迎使用-Matlab documents were broadcast wave function, welcomed the use of

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.

使用gabor变换作语音识别，下载自剑桥的实验室的网站，有兴趣的可以去网站上找资料-used for voice recognition, Cambridge downloaded from the website of the laboratory, interested parties can go to find information on the web site

隐马尔 柯夫模型( HM M ) 的传统 训练方法 ——Baum-W elch 算法只能 得到一个局 部最优 模型, 从而影 响 最终的识别率。对于 CHM M , 分段 K  平均方法来取得一个初始值可以解决这一问题, 但对 DHM M 却改进不大。 而 基于全 局搜索的 进化计 算的一 个重要 特点便 是可以 得到次 优解乃 至全 局最优 解。本文 把进 化计算 引入 到 DHM M 的训练中去, 提出一种改 进的进化训练方法, 实验结果表 明, 这种训 练方法初 具了全局搜 索和快速收 敛 的特点, 得到的模型优于传统方法和直接用进化计算所得的模型, 提高了系统的识别率。

语音识别专辑七- Speech recognition special edition seven