FFTofWaveFile

该文件为基于DTW的孤立词识别系统（特定人、小词汇量，未采用鲁棒语音识别技术）(The document DTW for isolated word recognition system based on (a specific person, small vocabulary, did not use robust speech recognition technology))

这是语音信号处理中求倒谱的程序,我自己编的,很简单,实现了一段序列求倒谱的功能(This is the voice signal processing for cepstrum procedures, I developed, is very simple. achieve a sequence for the functional cepstrum)

语音通信控件源码点对点专用版简介VSession语音通信控件源码点对点专用版，版本2.0，此版本集成了G729A压缩算法，实时传输协议，话音清晰流畅！使用简单易懂方便！此控件源码是在本人以前发布的控件件源码VSessionn2.0版本的基础上，加入点对点通信时的呼叫，应答，挂断等通话前后的同步功能，使其用于点对点通信更加方便！如果您想要在程序源码中自定义点对点间的通话联系方式，或者有一对多，多对多的 (Voice communications control source peer-to-peer special edition Introduction VSession voice communication control source peer-to-peer special edition, version 2.0, this version integration the G729A compression algorithm, Real-time Transport Protocol, clear and smooth voice! Easy to understand and easy to use! This control source is in the I control previously released the parts source VSessionn2.0 version based on peer-to-peer communication call, answer, hang up, etc. the synchronization function before and after the call to make it more convenient for point-to-point communication ! If you want to program source from the definition of point-to-point calls between Contact, or one-to-many, many-to-many)

语音识别程序，可以识别哪个声音是谁发出来的。(Voice recognition program, you can identify who sent the voice is coming.)

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.)

语音识别开源代码 sphinx 0.8 windows版本(sphinx 0.8，ASR)

从一段音频中识别说话人的性别。开发环境vs2005(To identify the audio from a speaker' s gender. Development Environment vs2005)

计算语音、音频信号的信噪比程序（包含音频样本文件）(Calculation of voice, audio program signal to noise ratio (including the audio sample file))

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)

目前2.4kbps的混合激励线性预测 (MELP) 语音编码方法已经被确定为美国新的联邦语音编码标准.本文提出了一种改进的MELP语音编码方法,利用滤波器相似度和基于LPC系数分类的矢量量化技术,可以把MELP的码率降到1.7kbps以下,仍有较好的合成语音质量.(An Improved MELP Speech Coder)