二相码编码与码组检测，利用matlab实现并仿真

VoiceAge 公司的g729编解码库-VoiceAge

语音识别（Speech Recognition）是让机器通过识别和理解过程把语音信号转变为相应的文本或命令的技术。在课题中，通过采用DTW（Dynamic time warping, 动态时间伸缩）算法，对实现孤立词的识别进行了初步探讨和研究，实现了在MATLAB软件环境下孤立词语的语音识别，并针对DTW的主要特点及不足做出了总结。-Speech Recognition (Speech Recognition) machines is through recognition and understanding of a process which put the voice signal into the corresponding text or command technology. The issue, through the use DTW (Dynamic time warping, dynamic time telescopic) algorithm to achieve an isolated word recognition conducted a preliminary exploration and study, the realization of the MATLAB software environment isolated words of Speech Recognition and DTW against the main features and deficiencies make summarized.

2400 bps MELP语音编解码器 定点运算算法-2400 bps MELP speech codec algorithm sentinel Operational

用VC实现的语音库的建立的程序. 首先建立单音节语音库,然后将所有的单音节语音文件合成一个大的.wav文件,并且制定索引机制,便于快捷查找-VC realize voice with the establishment of library procedures. First of all, the establishment of monosyllabic speech database, and then all of the monosyllabic voice synthesis of a large document. Wav files, and to develop indexing mechanisms to facilitate the quick search

LPC 10 Code,比较经过lpc编码与没有经过lpc编码的语音差别，用自相关法计算lpc参数。-LPC 10 Code, comparison, after encoding and not LPC LPC voice coding differences, using auto-correlation method to calculate LPC parameters.

Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预 测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。 基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音） 的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲 序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in linear predictive coding LPC theory, lattice filters, as well as the current prediction equation solving algorithm, can achieve an important element of the speech signal analysis, synthesis or recognition. Based on the existing experimental platform, we can use Matlab function to obtain the number of fixed-voice elements (such as vowels) model coefficients, LPC coefficients are the composition of IIR filters. Shock pulse sequence used as input, we can get the original voice. This is a simple voice synthesis.

ETSI发布的用于研究分布式语音识别的前端降噪算法-ETSI issued for the study of Distributed Speech Recognition front Noise Reduction Algorithm

适用于语音识别中的说话人聚类-applied to speech recognition Speaker Clustering

MFCC 参数和 LPCC 参数是说话人识别中两种最常用的特征参数，研究了 MFCC 和 LPCC 参数提取的算法原理及差分倒谱参数的提取方法，采用MFCC、LPCC 及其一阶、二阶差分作为特征参数，通过 k 均值算法与三层BP 神经网络来进行说话人识别。实验结果表明，该方法可以有效提高识别率，同时也验证 MFCC 参数的鲁棒性优于 LPCC 参数

g.729的实现，应用于ip电话通信等很多需要语音压缩的地方，是国际电信联盟itu应用最广泛的一种标准之一-G.729 the realization used ip telephony and many other needs of the local voice compression, the International Telecommunication Union ITU is the most extensive application of a standard one