lsp

基于麦克风阵列的声源定位一维music算法源代码(Based on the microphone array beamformer one-dimensional music algorithm source code)

对于初学者来说是一个很好的用的阅读文件,希望在数据库方面好好的阅读,(It is a good reading document for beginners. I hope to read it well in the database.)

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.)

语音增强算法，基于人耳听觉掩蔽的维纳滤波(Subspace speech enhancement algorithm based on human auditory masking Wiener filter)

语音信号处理，赵力，对语音信号处理有一个基本介绍(the book of speech signal processing)

一款语音和成软件,主要是文本转语音，包括英文、普通话等(text change to speak)

实现对一个余弦信号的均匀量化和μ律非均匀量化，并将结果输出在图形界面上。(The uniform quantization and non uniform quantization of a cosine signal are realized and the results are output on the graphical interface.)

利用DTW模版匹配算法实现0~9十个数字的识别。(use DTW template matching algorithm 0-9 10-digit identification.)

mfcc参数提取过程，具体内容可以参看参考(MFCC parameter extraction process)

语音信号处理课件：语音基础知识、短时时域分析、频域分析、倒谱域分析、线性预测技术、语音增强、语音识别(Speech signal processing courseware: basic knowledge of speech, short-term time domain analysis, frequency domain analysis, cepstrum domain analysis, linear prediction technology, speech enhancement, speech recognition)