语音信号处理中的提取算法，基本算法与…

这是小弟在实验室做的说话人身份识别程序，采用MFCC参数提取，并建立GMM模型来进行识别。希望会对大家有帮助。-This is the boy in the laboratory to do the speaker identification procedure for the extraction of MFCC parameters and to establish GMM model to be identified. I hope everyone will help.

提供信号处理中的Rasta变换的实现过程，并可以给出相应参数。开发环境为VC。- Provides in the signal processing the Rasta transformation realization process, and may produce the corresponding parameter. The development environment is VC.

利用全极点模型产生语音信号.rar-use all-pole model generated voice signal. Rar

用matlab录音的程序，可实现。快下啊 -using Matlab recording procedures can be realized. Express under ah!

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.

该文章是语音去噪方面的，比较新，希望对大家有帮助-The article is a voice de-noising aspects, relatively new, we want to help

语音信号分帧加窗程序，许多程序包里都没有的！-Speech signal windowed framing procedure, many programs do not have the bag!

利用微软的语音引擎使你的程序会朗读，需要安装微软语音引擎或者金山词霸- Causes your procedure using Microsoft s pronunciation engine to be able to read aloud, needs to install the Microsoft pronunciation engine or the Jinshan word tyrant

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&

这个函数是用来计算离散余弦变换的matlab程序。注意输入的参数要是实数-This function is used to calculate the discrete cosine transform matlab program. Note that if the real number input parameters