说话人识别是自动识别的过程,他在个人的基础信息包含在演讲。这种技术可以使用扬声器的声音来验证他们的身份和访问控制服务,如语音拨号,银行通过电话、电话购物、数据库访问服务、信息服务、语音邮件、安全控制领域的机密信息,和远程访问电脑。
Speaker recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. This technique makes it possible to use the speaker s voice to verify their identity and control access to services such as voice dialing, banking by telephone, telephone shopping, access services, information services, voice mail, security control for confidential information areas, and remote access to computers. (2014-10-27, matlab, 53KB, 下载7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，采用当前流行的SVM分类器，作为对比实验是一个不错的选择。
This example program is popular SVM classifier matlab speech recognition program to extract MFCC feature set, as a comparative experiment is a good choice.
(2012-07-04, matlab, 23968KB, 下载305次)
通常语音信号在增强时会出现混响现象，演讲者为了消除背景混响，不得不频繁地偏转头部的方向，这样会造成脉冲响应的不断改变。我们结合盲解卷法和频谱消去法来提高逆滤波器的滤波效果。我们利用输入语音信号间的相关系数矩阵计算出稳定、精确的室内脉冲响应的逆滤波器，而这些输入信号无需测量室内的脉冲响应就能被观测到。逆滤波能够消除早期的反射，这些反射包含混响中的绝大部分能量。之后，用频谱消去法来抑制逆滤波后的信号的尾部混响。本方法在实际适应性方面的表现通过具体的实验进行了验证，结果表明盲解卷法和频谱消去法的结合相较于单独使用一种方法，能够提供一个更优越的演说环境。
Usually voice signal will be enhanced when the reverberation phenomenon, the speaker in order to eliminate the background reverberation, and had to frequently deflect the direction of the head, this will cause the impulse response of changing. We combine the blind deconvolution and spectral elimination method to improve the filtering effect of the inverse filter. We use the correlation coefficient matrix between the input speech signal to calculate the stable, accurate inverse filter of the indoor impulse response, and these input signals do not need to measure the room impulse response can be observed. Inverse filtering to eliminate early reflections, these reflections contains most of the energy in the reverberation. After the spectrum elimination method to suppress the tail reverberation of the signal after the inverse filtering. Performance in the actual adaptation of the method by specific experimental validation results show that the blind deconvolution and spectrum to eliminate (2012-04-17, matlab, 39KB, 下载214次)
语音延长、变速处理程序，实现语音信号时域延长；
主要步骤是先对原始语音信号进行加窗，再对加窗后的语音信号进行时域扩展；
以上处理之后语音信号可能会带进新的噪声，所以再对处理后的语音信号，使用FIR滤波器进行去噪处理。
Voice extending process, extending the time-domain voice signal
The first major step is added to the original speech signal window, and then the windowed speech signal after the time-domain extension
After the above process may be brought into the new speech signal noise, so again the speech signal processing, the use of FIR filters for denoising. (2011-12-13, matlab, 254KB, 下载19次)
本工具箱利用lpc方法可以准确的提取出共振峰频率，并且解决了共振峰检测时滤波器阶数对所有说话人都一样的问题。本工具箱由美国卡内基梅隆大学一位教授编写，因此所有解释均为英文原版。
The toolbox to use LPC method can extract accurate formant frequency, and the resolution of formant filter order testing of all the speaker are the same problems. The toolbox by the United States a professor at Carnegie Mellon University to prepare, so all the explanations are the original English edition. (2008-12-16, matlab, 43KB, 下载144次)
自适应横向滤波器有两个权量，输入随机信号的样本间相互独立，且它的平均功率为0.01 ，信号周期为16个样点。利用计算机给出均方误差性能曲面，并在图中标注出最佳权向量解
Adaptive transversal filter has two right amount of input random signal between the samples independent of each other, and its average power of 0.01, the signal cycle for 16 samples. Given the use of computers MSE performance surface, and in the figure marked the best weight vector solution (2008-11-23, matlab, 1KB, 下载59次)
用matlab实现高斯混合模型的前期处理和分类训练！
Using Gaussian mixture model matlab realize the pre-treatment and classification of training! (2008-03-18, matlab, 59KB, 下载1759次)