Eriksson_method

数字语音处理及MATLAB仿真_张雪英编著第二版 源码(Digital audio processing and MATLAB simulation _ Zhang Xueying edited version second source code)

利用自适应滤波器对心音信号进行预测，包括了LMS算法，归一化LMS，变步长LMS和RLS算法，其中RLS的预测效果最好.(Heart sound signals using adaptive filter to predict, including the LMS algorithm, the normalized LMS, variable step size LMS and RLS algorithm, in which the best forecast of RLS.)

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)

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)

熵值越大则每个符号包含的平均信息量越大。有研究发现，在有噪声的语音信号中，语音信号的熵和噪声信号的熵存在着较大的差异，对噪声信号来说在整个频带内分布相对平坦，熵值小，语音信号集中在某些特定频段内，熵值大。因此利用这个差异可以区分噪音段和语音段。(The greater the entropy is, the greater the average information of each symbol is. It is found that, in noisy speech signals, the entropy of speech signals and the entropy of noise signals are quite different. For noisy signals, the distribution is relatively flat in the whole frequency band, and the entropy value is small. The speech signal is concentrated in some specific frequency bands, and the entropy value is large. So the difference can be used to distinguish the noise segment and the speech segment.)

全面介绍语音合成的知识比较系统很适合语音学习(A comprehensive comparative introduction speech synthesis system is suitable for knowledge language learning)

　介绍了一种针对语音增值服务开发的智能语音终端,该终端以EM78806A单片机为核心研发,借助公共电话网络,实 现了电话与局端语音信箱中心的通信与智能化控制。文中主要对智能语音终端的硬件结构、工作原理及程序设计方法进行 了论述。(Introduce a value-added voice services for the development of intelligent voice terminal, the terminal to EM78806A Singlechip as the core research and development, through the public telephone network, the realization of a telephone central office voice mail with the Center for Communications and intelligent control. The main text of the intelligent voice terminal hardware structure, working principle and procedure of design methods are discussed.)

提取MFCC的matlab代码，用于提取MFCC-梅尔倒普系数(MFCC extraction of matlab code, for the extraction of MFCC-Mel Pu inverted coefficient)

GMM说话人识别平台全套，从特征提取到模式识别，到识别率，一整套都有(GMM speaker recognition platform full, from feature extraction to pattern recognition, the recognition rate, set both)

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)