DOA_Estimation

ADPCM(Adaptive Pluse Coding Modular)便解码器，供大家学习参考(ADPCM (Adaptive Pluse Coding Modular) decoder will be available for everyone to learn the reference)

通常语音信号在增强时会出现混响现象，演讲者为了消除背景混响，不得不频繁地偏转头部的方向，这样会造成脉冲响应的不断改变。我们结合盲解卷法和频谱消去法来提高逆滤波器的滤波效果。我们利用输入语音信号间的相关系数矩阵计算出稳定、精确的室内脉冲响应的逆滤波器，而这些输入信号无需测量室内的脉冲响应就能被观测到。逆滤波能够消除早期的反射，这些反射包含混响中的绝大部分能量。之后，用频谱消去法来抑制逆滤波后的信号的尾部混响。本方法在实际适应性方面的表现通过具体的实验进行了验证，结果表明盲解卷法和频谱消去法的结合相较于单独使用一种方法，能够提供一个更优越的演说环境。(Usually voice signal will be enhanced when the reverberation phenomenon, the speaker in order to eliminate the background reverberation, and had to frequently deflect the direction of the head, this will cause the impulse response of changing. We combine the blind deconvolution and spectral elimination method to improve the filtering effect of the inverse filter. We use the correlation coefficient matrix between the input speech signal to calculate the stable, accurate inverse filter of the indoor impulse response, and these input signals do not need to measure the room impulse response can be observed. Inverse filtering to eliminate early reflections, these reflections contains most of the energy in the reverberation. After the spectrum elimination method to suppress the tail reverberation of the signal after the inverse filtering. Performance in the actual adaptation of the method by specific experimental validation results show that the blind deconvolution and spectrum to eliminate )

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)

谱减去，最常用的程序，实习除噪，语音增强(To achieve denoising)

使用子带谱熵进行的端点检测，将谱熵分为几个子带，检测效果不错(The use of sub-band entropy of the endpoint detection, the spectral entropy is divided into several sub-band to detect the effect of good)

模式识别k聚类算法源代码，c语言内容描述。(Pattern Recognition Chapter )

通过广义互相关法求时延 三种加权系数 ROTH SCOT PHAT(Delay by the generalized cross correlation method)

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.)

做语音识别时用到DTW 算法。便于初学者的学习。(DTW algorithm is used for speech recognition. Easy for beginners to learn.)

说明：  用MATLAB实现lms算法及改进的lms的源代码(Using MATLAB to achieve improved LMS algorithm and LMS source code)