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语音库开发工具源码，例程程序创建机读SAPI.SpVoice对象开发语音库。(Speech development tools source , routine program to create machine-readable SAPI.SpVoice speech object development .)

一种用于语音或者心音去噪和提取包络的预处理方法，对于后面信号处理更加方便(A method for denoising and extract voice or heart sounds envelope pretreatment method, the signal processing is more convenient for the latter)

LMS自适应滤波算法是很常见的滤波算法。本代码包括常用的LMS自适应滤波算法，如基本LMS算法，解相关LMS算法，滤波型LMS算法，变换域LMS算法等。(LMS adaptive filtering algorithm is a common filtering algorithm. This code, including commonly used LMS adaptive filtering algorithms, such as the basic LMS algorithm, decorrelation LMS algorithm, filter-based LMS algorithm, transform domain LMS algorithm.)

观测实时模拟信号（语音）的频谱 用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图。 (Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw the spectrum map and voice waveform.)

语音信号处理，阵列为10阵元线阵，宽带和窄带波束形成(Speech signal processing, 10 sensors of microphone linear array,Broadband and narrowband beamforming)

这是一个在线签名的程序，使用了AR实现算法和DTW算法(This is an online signature process, using the AR algorithm and DTW algorithm)

利用所给的中心削波,计算自相关/AMDF代码,选取一段语音,使用Praat预估其基音周期范围,合理设置预估的极值点距离等参数,计算并画出语音的基音频率曲线(By using the given center clipping, the autocorrelation /AMDF code is calculated, a speech is selected, the range of the pitch period is estimated by Praat, the estimated distance of the extreme point is set up reasonably, and the pitch frequency curve of the speech is calculated and drawn.)

C / winform_ multiple forms transfer data operations between each othe

11阶自适应滤波器的LMS算法，并且与RLS进行了比较(11 steps LMS adaptive filter algorithm and the RLS were compared)

利用GMM训练模型，并用于说话人识别等相关领域，有不错的效果(GMM apply for text to speech)