▍1. hmm
自己认真修改的程序代码,并且通过测试可以运行的!对HMM-GMM的初学者有用。(Own serious modify the program code, and the test can be run. HMM-GMM useful for beginners.)
语音信号的盲源分离算法研究及应用,本程序利用matlab实现(Blind Source Separation Algorithm Research and Application of the voice signal, the program using matlab realize)
基于单通道的在线次级通道建模的主动 噪声控制FXLMS算法的MATLAB实现( a single channel feed-forward active noise control system based on On-line secondary path modellingthe FxLMS alogrithms You can find many information about this simulation in "Use of random noise for online transducer estimate in an adaptive attenuation system " written by L,J Eriksson and M.A Allie in 1989.)
运用HMM进行语音识别之后的孤立词识别,里面函数俱全。可以直接运用,简单方便(Speech recognition with HMM isolated word recognition, which function and taste. Can directly use, simple and convenient)
11阶自适应滤波器的LMS算法,并且与RLS进行了比较(11 steps LMS adaptive filter algorithm and the RLS were compared)
voice recognition using mfcc
the document explains speech recognition using mel frequency cpstrum coeefficient and noise reduction method
进行语音合成和分析,最基本的数据来源和可靠合成基础(Speech synthesis and analysis, basic data sources and reliable synthetic base)
matlab程序,从语音中提取lpcc,可以修改维数,带详细中文解释,方便阅读。(matlab program, extracted from voice lpcc, you can modify the dimension, with a detailed explanation of the Chinese, and easy to read.)
基于LMS算法和NLMS算法的自适应声回波消除器AEC,实现的主要功能是:在用户对电视发出的指令混杂了电视扬声器的声学回声的情况下,滤掉电视声音得到较为纯净的用户语音指令信号。(The main function of the adaptive filter based on the LMS algorithm and the NLMS algorithm is mixed: in the instruction issued by the user on the television TV speakers acoustic echo case, the TV sound was removed by filtration to obtain relatively pure user voice command signal.)
基于最小均方误差的助听器语音去噪算法, 用于以最大程度消除助听器接受的语音杂音.(Speech denoising algorithm based on the minimum mean square error of a hearing aid for the hearing aid acceptable voice murmur eliminated to the greatest extent.)
melp编解码 matlab 程序 (melp codec matlab program)
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)
科大讯飞关于语音识别开发SDK库,支持语音转汉字功能及根据读汉字功能。有兴趣的可以好好研究研究(IFLYTEK speech recognition development SDK library are interested in a good studies)
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.)
C#版语音识别系统,精准度超强,到得收藏(C# version of the speech recognition system, super accuracy to Favorites)
用于说话人识别(声纹识别)中训练过程和识别过程的高斯混合模型程序(GMM model for the training process or test process of speaker identification)
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!)