SpecSubTD_Boll79

针对信号自适应chirplet分解未知参数多、实现起来比较困难的特点，文献[1]提出了一种新的chirplet分解快速算法。该算法利用计算信号的二次相位函数，得到其能量分布集中于信号的调频率曲线上的结论，此时通过谱峰检测可同时获得chirplet调频率、时间中心和幅度的估计；然后通过解线性调频技术获得其初始频率和宽度的估计，仿真结果验证了本文算法的有效性。(this code includs one method for chirplet analasis,that is used in voice and ultra-wave analasys,etc.)

speech specgram code

台湾国立中央大学开发的EMD-HHT算法,其中,EMD-HHT的创始人为该中心的主任.(EMD-HHT IS SPECIALLY DESIGNED FOR PROCESSING NONSTATIONARY AND NONLINEAR SIGNALS. IT CAN DECOMPSE SIGNALS AND THEN RECONSTRUCT SIGNALS ACCORDING TO SOME CRITERIA. AFTER THE PROCESS, SIGNALS WOULD HAVE HIGH SIGNALTO NOISE RATIO. IT HAS BEEN APPLIED TO VARIOUS TYPES OF SINALS FROM INDUSTRY AND ACHIEVED GREAT SUCCESS. IT EVOLVES FROM THREE SOURCES: FRANCE SCIENTIST G RILLING, MATLAB FILE EXCHANGE CENTRE BY MR TAN AND TAIWAN NATIONAL CENTRAL UNIVERSITY CENTRE.)

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)

基于谱减法的语音降噪处理，里面有声音文件，可以直接运行观察效果(Based on the spectral subtraction speech denoising processing, there are sound files, can be run directly observed effect)

从wave中读取数据，通过mfcc提取特征系数。从txt文件中读取基于GMM的分类器，最终识别为那种类型(Read data from the wave through the MFCC feature extraction coefficient. Txt file to read from the GMM-based classifier, and ultimately identified as the type)

ADPCM(Adaptive Pluse Coding Modular)便解码器，供大家学习参考(ADPCM (Adaptive Pluse Coding Modular) decoder will be available for everyone to learn the reference)

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/)

1、文件夹中包含了经典功率谱估计和自适应均衡算法两个实验的所有程序。 2、R.m、LMS.m、LMSmain.m为自适应均衡算法的程序： R.m用来计算输入信号的自相关矩阵及其特征值； LMS.m为时域LMS算法，用统计的方法仿真得出不同信道参数和不同步长下的学习曲线； LMSmain.m为实验主程序，按照实验要求中的具体数据得到实验结果和曲线。 3、functionx.m、fzhouqitu.m、spectrum.m、bt.m、bart_lett.m、welch.m、SPECTRUMmain.m为经典谱估计的程序： functionx.m产生需要进行谱估计的函数； fzhouqitu.m用来计算信号周期图的函数； spectrum.m是用周期图法进行谱估计的函数 bt.m是用BT图法进行谱估计的函数 bart_lett.m是用BARTLETT法进行谱估计的函数 welch.m是用WELCH法进行谱估计的函数 SPECTRUMmain.m是主程序，按照实验要求中的具体数据得到实验结果和曲线。 (a document folder contains the classic power spectrum estimation and adaptive equalization algorithm for the two experiments all the procedures. 2, R.m, LMS.m, LMSmain.m adaptive equalization algorithm for the procedure : R.m used to calculate the input signal and the correlation matrix eigenvalue; LMS.m too LMS algorithm, use statistical simulation method come to a different channel parameters and the synchronous learning curve; LMSmain.m main program for the experiment, according to the experimental requirements of the specific data and the curve of the experimental results. 3, functionx.m, fzhouqitu.m, spectrum.m. bt.m, bart_lett.m, welch.m. SPECTRUMmain.m classical spectrum estimation procedures : functionx.m have a need for spectrum estimation function; fzhouqitu.m used to calculate t)

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)