基于统计特征的语种识别算法分析与实现。1.提取语音的客观统计特征；2、通过分类器建立训练学习模型；3、将模型运用于汉语、英语、日语等语种识别实验，与人的主观感觉做对比
Based on statistical language identification algorithm analysis and Realization of objective statistics. The extraction of speech features 2, the classifier built the training learning model 3, the model was applied in Chinese, English, Japanese and other languages recognition experiments, and human subjective sensation compared (2016-05-08, matlab, 7KB, 下载14次)
基于匹配滤波器的语音识别。本程序接触匹配滤波对加噪语音进行匹配输出，使得信号的信噪比得到提升，送入后端DTW的语音识别系统，实验证明，本程序对语音提升有很大的帮助
A speech recoginiton base on the matched filter. this code will use matched filter and get speech SNR output enhanced. then we will extrace MFCC, send the feature to the DTW recongition system, the test result verify that this method is good for speech recoginition SNR enhanced. (2015-01-10, matlab, 4547KB, 下载16次)
有交互式界面的语音信号处理器，可实现录音后对语音信号进行快放、慢放、声音放大等功能
There are interactive voice signal processor interface, can be realized after the recording of the voice signal fast, slow, sound amplification and other functions (2014-04-30, matlab, 3KB, 下载10次)
用matlab实现巴特沃斯带通滤波器的M文件。本例读取电脑自带的XP开机声音，加上噪声，然后滤波看效果。
Using matlab to achieve the M-file Butterworth bandpass filter. This case, read the computer comes with XP start-up sound, plus noise, and then filter to see the effect. (2013-05-11, matlab, 1KB, 下载30次)
基于svm分类器的汉语声调识别，实现了对声调四声的识别，纯净语音识别率100 ，噪声环境下识别率90
The svm classifier based Chinese tone recognition, the four tones of tone recognition, the pure voice recognition rate of 100 , 90 recognition rate in noisy environments (2013-01-23, matlab, 6KB, 下载114次)
基于LMS算法和NLMS算法的自适应声回波消除器AEC，实现的主要功能是：在用户对电视发出的指令混杂了电视扬声器的声学回声的情况下，滤掉电视声音得到较为纯净的用户语音指令信号。
The main function of the adaptive filter based on the LMS algorithm and the NLMS algorithm is mixed: in the instruction issued by the user on the television TV speakers acoustic echo case, the TV sound was removed by filtration to obtain relatively pure user voice command signal. (2012-10-29, matlab, 1KB, 下载69次)
本实例程序是基于matlab的语音识别程序，通过提取传统的MFCC特征集，采用HMM分类器，作为对比实验是一个不错的选择。
This example the program matlab speech recognition program to extract MFCC feature set, the HMM classifier, as a comparative experiment is a good choice.
(2012-07-04, matlab, 23480KB, 下载90次)
基于特定人情感的语音识别，语音情感特征提取；语音情感分类器的设计，完成了一个特定人语音情感识别的初步系统。内容很丰富
In this article, we have accomplished a system for emotion recognition of speech by establishing a database of speech with certain emotion by certain people analyzing emotion features and designing a sort system for emotion recognition (2012-04-18, matlab, 253KB, 下载107次)
IIR型切比学府滤波器 并且分析了他的归一化后的 频率响应 和特定输入信号经过滤波后的结果 得出零极点图
IIR model than learning filter and analysis cut the normalization of the frequency response and specific input signal through the filter results from zero pole figure
(2011-11-15, matlab, 1KB, 下载2次)
自适应滤波器的核心是自适应算法。最小均方算法(LMS) 和递归最小二次方算法RLS) 。通过对这两种算法的推导过程,可以清楚的了解到自适应算法的理论体系,
Adaptive filter is the core of adaptive algorithm. Least mean square algorithm (LMS) and recursive least quadratic algorithm RLS). Through the derivation of these two algorithms, adaptive algorithms can clearly understand the theoretical system,
(2011-09-23, matlab, 3KB, 下载7次)
这是一篇工学硕士论文，主要说关于说话人语音识别，，以支持向量机为分类器，分别以MFCC和IM-FCC为特征单独执行分类，将得到结果按某种方式融合，取两者之长，最后做出判决来提高说话人识别系统的性能。
Speaker recognition is a kind of technology that take the use of the
information contained in the speech signal to identify who is the speaker or to
confirm whether the speaker is the claimed one。Now in low-noise and low
distortion environment，speaker recognition has performed quite perfect．But noise is everywhere in the practical application environment，which make the speaker recognition rate dramatically declined．As a result,extracting the robust feature and designing the effective classifier to get a good performance in the noise environment have become a hotspot,and then realizing the speaker
recognition system come to practice from laboratory． (2011-08-13, matlab, 1914KB, 下载109次)
产生一个连续信号，包含低频，中频，高频分量，对其进行采样，进行频谱分析，分别设计三种高通，低通，带通滤波器对信号进行滤波处理，观察滤波后信号的频谱
NI FHHFJSDG (2009-12-26, matlab, 1KB, 下载88次)
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. (2009-12-20, matlab, 277KB, 下载419次)
本程序为一个用matlab实现的语音滤波程序，能计算并显示语音信号的幅频特性，巴特沃斯低通滤波器幅度和相位响应，引入噪声后的幅频特性，以及滤波后的DFT幅频特性。本程序附有语音文件以供试验。
This program is an implementation using matlab voice filtering procedures, can calculate and display the speech signal amplitude-frequency characteristics of Butterworth low-pass filter magnitude and phase response, after the introduction of noise, amplitude-frequency characteristics, as well as the filtered DFT amplitude-frequency characteristics. This procedure for testing with voice file. (2009-10-07, matlab, 4KB, 下载117次)
主动噪声控制的matlab仿真模型，使用了LMS滤波器
MATLAB simulink model of active noise cotnrol which applies LMS adaptive filter. (2009-08-25, matlab, 7KB, 下载108次)
改程序实现了根据简朴音符合成音乐的功能，并且对不同乐器的声音进行傅里叶分析，可以合成具有不同音色的音乐
Reform process to achieve a synthesis of musical notes based on simple features, and the voices of different instruments to Fourier analysis, can be synthesized with different sound of music (2009-07-13, matlab, 366KB, 下载108次)
使用自相关法提取基频法来判别男女声，并与带通滤波器幅度筛选法及幅度作差提取基频判别法分别作比较。最后的判别正确率接近90％。
The use of autocorrelation-based frequency extraction method to determine male and female voices, and with band-pass filter magnitude and rate of screening for differential extraction of fundamental frequency, respectively discriminance for comparison. Finally, the discriminant accuracy rate close to 90. (2008-12-18, matlab, 7213KB, 下载279次)
书上的lms算法的输出为估计量（滤波结果）和均方误差阵
而这个程序的输出结果是滤波器系数和误差系数
实际使用的时候，这样给出的结果应该更实用
Book on the LMS algorithm for estimating the volume of output (filter results) and the mean square error matrix and the output of the results of this procedure is the filter coefficients and the error coefficient of the actual use of the time, so given the results of which should be more practical (2008-05-16, matlab, 4KB, 下载32次)
比较经典的用lms算法实现自适应均衡器的matlab的程序，适合初学者学习
Comparison of the classical LMS algorithm with adaptive equalizer matlab procedures, suitable for beginners to learn (2007-11-26, matlab, 68KB, 下载185次)
TD-PSOLA MATLAB编写语音合成模拟程序
The simulation program of combined speech signal beased on TD-PSOLA MATLAB method (2005-06-05, matlab, 37KB, 下载468次)