floating

这是自己毕业设计做的PCA和LDA的结合，训练和识别别过程都有，用的是ORL库里的图像，具有较高的识别率！ (This is a combination of PCA and LDA graduation design, training and do the process of the identification with ORL library of images, has a high recognition rate!)

程序功能可以基本上实现基于C语言的G.729A语音算法的解码功能，添加了部分本人的注释和理解，便于读者理解(Program features can basically C language-based decoding algorithm G.729A voice, add some of my notes and understanding, easier to understand)

这是语音播放接口组件，将语音数据输入，会播放声音。经过测试确保可用。(This is the voice player interface module, the voice data input, will play the sound. After testing to ensure that the.)

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

产生双音多频信号。输入任何参数就可得到所要的结果。(Generate DTMF signal. Input any parameters, you will get what you need.)

这 里主要对LMS算法及一些改进的LMS算法（NLMS算法、变步长LMS算法、变换域LMS算法）之间的不同点进行了比较，，在传统的LMS算法的基础上发 展了LMS算法的应用。另一方面又从RLS算法的分析析中对其与LMS算法的不同特性进行了比较。 (Here the main difference between the LMS algorithm and improved LMS algorithm (NLMS algorithm, variable step size LMS algorithm, the transform domain LMS algorithm) comparison, the traditional LMS algorithm based on the development of the application of the LMS algorithm . On the other hand and from its different characteristics of the LMS algorithm of the analytical analysis of the RLS algorithm.)

这是一个与语音识别与端点检测有关的文件,是用C++写的源程序,相信对大家有用(This is a speech recognition with the Endpoint Detection of the documents are written in C source files, it may be useful to think)

语音AEC的demo程序和库源代码，在win7下验证可行。(audio aec demo in windows7.)

语音信号处理中的所有基本的算法的代码（C语言版）(Speech signal processing algorithms in all the basic code (C language version))

语音合成程序源码！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，的到短时谱与短时谱包络。短时谱除以短时谱包络的到声源短时谱，对声源短时谱的实部与与虚部分别进行线性插值，就能达到改变变语音信号基频的目的，然后再进行频域反变换，可的到变换后的短时语音信号。短时谱包络部分也能独立改变，以达到改变音色的目的。 (Voice synthesis program source! psalo frequency domain pitch synchronous superposition method. It was first carried out on the original speech signal a short-time frequency-domain transform, to the short-time spectrum and short-time spectrum envelope. The short-time spectrum divided by the short-time spectrum envelope of the short time spectrum of the sound source, the short time spectrum of the real part of the sound source, and the imaginary parts of the linear interpolation, can achieve the purpose of changing the fundamental frequency of the alternating speech signal, and then then the inverse transform of the frequency domain, can be to the short-time speech signal after conversion. The short-time spectral envelope section can be varied independently, in order to achieve the purpose of changing the tone.)