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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.)

声纹识别和语音识别，语音识别的模块使用的讯飞云。声纹识别的代码则是参考了Dake前辈的代码。在此提出感谢。(Voiceprint recognition and speech recognition, voice recognition module for use hearing Fei. Voiceprint identification code is a reference to the Dake predecessors code. Presented in this thanks.)

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))

本文主要研究普通计算环境下语音自动识别(ASR)技术的软件实时实现，基于语音识别的基本原理和过程。在VC环境下建立了一套语音识别系统及其演示软件。实验表明，该系统对特定人情况有较高的识别率和一定的实时性(In this paper, under the computing environment on general automatic speech recognition (ASR) technology to achieve real-time software, speech recognition based on the basic principles and processes. VC environment in the establishment of a voice recognition system and its presentation software. The experiments show that the systems are in particular have a high recognition rate and a certain degree of real-time)

这是一个语音的文件的一个FFT转换文件,是用C++写的源程序,相信对大家有用(This is a voice of a paper document conversion FFT is written in C source files, it may be useful to think)

基于vc++开发，dtw语音识别,MFCC参数(DTW recognice based on vc++)

HMM语音识别,包含主要算法,进行语音识别(HMM speech recognition, contains the main algorithm for speech recognition)

C语言实现的端点检测,具有良好的效果,对于做语音识别者来说，是很好的参考(C language implementation of endpoint detection, with good results)

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.)