VC编写的音频频谱柱状图可视化显示界面

中文语音朗读的程序-voice reading aloud procedures

adpcm压缩算法，测试过，效果不错-Adpcm compressing algorithm. Has good behaviour through testing.

这是关于盲信号源分离技术的经典文章，主要用于盲信号源的分离。-This is a blind signal source separation on the classic technical articles, mainly used for blind source separation.

语音技术（VOIP）中常用的编码解码器（codess）源码-voice technology (VoIP), commonly used codec (codess) FOSS

采用LMS算法的频域均衡技术研究,lms在频域均衡方面的应用。-LMS algorithm using frequency domain equalization technology research, lms balance in the frequency domain applications.

英文版G.729语音压缩标准。 GENERAL ASPECTS OF DIGITAL TRANSMISSION SYSTEMS CODING OF SPEECH AT 8 kbit/s USING CONJUGATE-STRUCTURE ALGEBRAIC-CODE-EXCITED LINEAR-PREDICTION (CS-ACELP)-The English version of G.729 voice compression standards. GENERAL ASPECTS OF DIGITAL TRANSMISSIONSYSTEMSCODING OF SPEECH AT 8 kbit/sUSING CONJUGATE-STRUCTUREALGEBRAIC-CODE-EXCITEDLINEAR-PREDICTION (CS-ACELP)

不错的FIR滤波器源代码！-good FIR filter source code!

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.

:高斯混合模型(GMM)是一种经典的说话人识别算法，本文在实现其算法的同时，主要模拟了不同噪声环境情况下高斯混合模型 (GMM)的杭嗓声性能，得到了一些有益结论。 -Gaussian mixture model (GMM) is a classic speaker recognition algorithms, this algorithm at the same time in fulfilling its main simulated environmental conditions under different noise Gaussian mixture model (GMM) of the Hang throat sound performance, and obtained some useful conclusions.

Intercom 是一个 Unix系统上灵活的语音传输软件。支持标准音频压缩比如GSM, G.711, and G.72x和其他音频编码。Intercom专为高速网络设计来传输高品质的语音，也支持窄带传输-Intercom is a Unix system flexible voice transmission software. Support audio compression standards such as GSM, G.711, and G.72x and other audio coding. Intercom specifically designed for high-speed network to deliver high-quality voice, also support narrowband transmission