zengqiang

SFS提供了计算语音本质的一系列工具。包括支持各种数据格式，信号处理和图形，频谱、共振峰分析，基频提取等等。在线文档说明请访问http://www.phon.ucl.ac.uk/resource/sfs/help/(SFS provides a voice calculated nature of a series of tools. Including support for various data formats, signal processing and graphics, spectrum, formant analysis, frequency, etc. extraction. Online document shows, please visit http :// www.phon.ucl.ac.uk/resource/sfs/help/)

语音通信，延迟小，质量高，游戏玩家专用哈。(Voice communications, delay, high quality, dedicated gamers Kazakhstan.)

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.)

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)

男女声识别程序，中科院数字信号处理第二次作业，分辨率百分之90 以上，带GUI，在前人基础上修改，保证能用，不能用联系我(Male and female vocal recognition program, digital signal processing operations of the Chinese Academy of Sciences, resolution of more than 90 , with a GUI, modify their predecessors on the basis of guaranteed to not use)

VAD去噪算法，可以有效的过滤静音和无效的语音，同时消除了一些背景噪声。(VAD denoising algorithm, can effectively filter invalid mute and voice, while eliminating some background noise.)

不错的盲源分离工具箱，实现对声源的分离,包含多个子程序(a nice toolbox of blind source separation used to separate multiple sources)

Chapter 2 Low Bit-Rate Speech Coding In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.(In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.)

Get color from RGB 从RGB中获得颜色(Get color from RGB gets colors from RGB)

本次课程设计主要介绍了语音信号的录制、语音信号的采集与分析、语音信号的采样、语音信号的频谱分析、加噪后的频谱分析、滤波器的设计以及对于语音信号的回放等知识。通过 Cool Edit录制一段语音信号，运用MATLAB对其进行时域与频域分析，然后设计滤波器加入噪声进行滤波处理，比较滤波前后始于波形或频谱的变化，最后实现对于语音的回放设计。(Digital voice processing filter design)