Speech-enhancement

功率倒置算法的LMS实现，因为自己遇到了点问题，本人一直在找这个程序，没找到，故自己重新检查把问题发现了，相信如果你要实现的话可能会遇到这个问题祝你好运！(Power inversion algorithm LMS implementation, because they encountered a point problem, I ve been looking for this program, did not find it to re-examine the problem yourself discovered, I believe if you want to achieve, then you may encounter this problem Good luck!)

高斯混合模型参数初始化程序，在对高斯混合模型的建立之前采用KMEANS算法进行初始化(Gaussian mixture model parameter initialization procedure, in the Gaussian mixture model is initialized before the algorithm used KMEANS)

使用高斯混合模型（GMM），对声音信号进行识别，包括对特征参数的采集、训练和识别过程(Using the Gaussian mixture model (GMM)to realize the recognition of the sound signal, including the acquisition of the characteristic parameters,the training and the recognition process)

这是基于隐马尔可夫模型的连续语音识别代码，不同于dtw,这是个完整的工程，我把需要的voicebox也放进去了，一共用到9个子函数，包括模板的训练算法何识别算法，是完全可以使用的。(This is based on the hidden Markov model of continuous voice recognition code, different the dtw, which is a complete project, I need to put the voicebox into a total of nine sub-functions, including the template training algorithm to identify the algorithm, Is fully available.)

这是一个xmms的3D音效插件。算法简单，效率很高，如果使用音箱来听效果会更好！(xmms This is a 3D audio plugin. Algorithm is simple, highly efficient, if used to listen to speakers results will be better!)

采用加权网络中节点强度和权重都是幂率分布的模型，计算两个矩阵之间的欧氏距离，实现了对10个数字音的识别。( Using weighted model nodes in the network strength and weight are power law distribution, Calculation of the Euclidean distance between the two matrices, To achieve the recognition of 10 digital sound.)

psola算法的设计与实现，经过调试已经可以运行(psola Algorithm Design and Implementation, after debugging has to run)

语音语谱图分析程序,提供了数字样本数据,值得参考.matlab程序.(language voice spectrum analysis procedures, and providing a digital sample data, a good reference. Matlab procedures.)

自己认真修改的程序代码，并且通过测试可以运行的！对HMM-GMM的初学者有用。(Own serious modify the program code, and the test can be run. HMM-GMM useful for beginners.)

在evmdm642平台下的一个语音采集-回放任务，用48K/秒的采样率采集立体声音频输入，用给定的FIR 滤波参数处理数据后再播放出来。注：内附个人整理的详细操作文档。(evmdm642 platform in a speech to the gathering-intervals task with 48 K/s sampling rate Acquisition stereo audio input, used to the FIR filter parameters for data processing later. Note : with a detailed personal finishing operations documentation.)