几种语音增强关于mmse算法，维纳滤波，掩蔽增强

语音合成与语音处理-voice synthesis and voice processing

此程序为自相关法进行基音检测的程序，主要应用于语音识别。-program for petch detection using auto-correlation method, mainly for speech recognition

数字音频格式有很多种，其质量与采样频率和采样精度两个参数有关。频率的表示单位为赫兹〔Hz〕，它表示每秒采样次数。采样频率越高，音质就越好。采样精度为每次采样所存储的数据数量，它决定每个数字信号所能够表示的离散振幅的数量。存储每个样本的数据越多，音质就越好。但是高品质的声音需要占用大量的内存和磁盘空间。考虑到网络带宽，在Internet连接上传输就需要花费很长的时间。对于Applet来说，保证声音文件的最小化是极为重要的。-digital audio format there are many, the quality and frequency of sampling and sampling accuracy of the two parameters. The frequency of Hertz unit of [Hz], it said the number of samples per second. The higher sampling frequency, the sound quality better. For each sampling precision sampling by the amount of data storage, it was decided that each digital signal can be expressed as the number of discrete amplitude. Each sample storage of more data, better sound quality. But high-quality voice would take a lot of memory and disk space. Consideration of the network bandwidth, the transmission on the Internet connection on the need to spend a very long time. Applet for instance, to ensure that voices of the smallest document is very important.

N best hypothesis document N best hypothesis as used H-N best hypothesis document N best hypothesis as used HMM

GSM语音压缩编码源代码] GSM语音压缩编码源代码-GSM voice compression coding 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.

本程序的压缩编码是G.729的编解码程序，其的算法是基于CS-ACELP的算法基础上的编码。用在语音通信中，例如手机中-This program s arithmetic is base on ITU G.729 Speech Vocoder, which is a CS-ACELP compressing coder, usually use in speech comunication, such as mobile phone.

WAVE文件作为多媒体中使用的声波文件格式之一，它是以RIFF格式为标准的。RIFF是英文Resource Interchange File Format的缩写，每个WAVE文件的头四个字节便是“RIFF”。WAVE文件由文件头和数据体两大部分组成。其中文件头又分为RIFF／WAV文件标识段和声音数据格式说明段两部分。WAVE文件各部分内容及格式见附表。-WAVE as a multimedia documents used in the acoustic format, it is based on RIFF format as a standard. RIFF English Resource Interchange File Format of the initials, each WAVE document for the first four bytes is "RIFF." WAVE document from the document and data on the first two major components. The first document which is divided into RIFF/logo of WAV files and voice data formats note of two parts. WAVE part of the document content and format of the attached tables.

大规模连续语音识别工具，比较实用。可以做实际建模-large-scale continuous speech recognition tools, more practical. Modeling can do practical

此文件是好将声音发送到 dsk 6700 和处理的任何声音。您可以配置此代码由 dsk 海缆无源探测格式然后通过调整可以将声音发送到 dsk 的 aic23 编解码器寄存器的端口处理它，然后回 aic23 和 dsk 的输出中听到它。它是任何一个想要使用 dsk 来学习如何初始化 dsk 的好榜样。