基于matlab 提取语音信号中的特征参数mfcc的源程序。

语音识别:语音矢量化及算法及与原文件的矢量对比功能源代码-Speech Recognition : Speech vector algorithm and the original document and the contrast vector function source code

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.

为微软SAPI(Text-to-Speech API)封装的类- Kind (Text-to-Speech API) seals which for Microsoft SAPI

语音处理中的LP算法的MATLAB程序，希望可以帮到需要的人-speech processing algorithm in MATLAB LP procedure, the desire to help people in need

G711语音压缩源码-G711 voice compression FOSS

通过ACTIVEX控件与MATLAB进行双向通信实现实时语音端点检测-through ACTIVEX control and MATLAB for two-way real-time voice communication endpoint detection

OTOMATİ K KONUŞ MA TANIMA ALGORİ TMALARININ UYGULAMALARI Kö ksal Ö CAL Ankara Ü niversitesi Fen Bilimleri Enstitüsü Elektronik Mühendisliğ i Anabilim Dalı Danı ş man : Yrd. Doç . Dr. H. Gö khan İ LK Bu ç alı ş mada, SMM (Saklı Markov Model) tabanlı izole bir kelime tanı ma sistemi geliş tirilerek, sesin akustik parametreleri LPC (Linear Predictive Coding), LPCC (LPC Cepstrum), CEPS (Ayrı k Fourier dö nüş ümü tabanlı cepstrum) ve MFCC (Mel Frequency Cepstral Coefficients) ‘nin konuş macı dan bağ ı msı z konuş ma tanı ma sistemlerindeki performansları değ erlendirilmiş tir. Değ iş ik akustik parametrelerle birlikte değ iş ik S-OTOMATİ K KONUŞ MA TANIMA ALGORİ TMALARININ UYGULAMALARI Kö ksal Ö CAL Ankara Ü niversitesi Fen Bilimleri Enstitüsü Elektronik Mühendisliğ i Anabilim Dalı Dan&

写mp3文件格式的matlab 源程序-Mp3 file format to write the matlab source code

语音频率分析使用matlab实现，很经典了-Voice frequency analysis using matlab implementation, it is the classic of the Hehe

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.