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hmm-GMM-KEAMS

  • Wavelet Subband coding for speaker recognition The fn will calculated subband energes as given in

    Wavelet Subband coding for speaker recognition The fn will calculated subband energes as given in the att tech paper of ruhi sarikaya and others. the fn also calculates the DCT part. using this fn and other algo for pattern classification(VQ,GMM) speaker identification could be achived. the progress in extraction is also indicated by progress bar.

    标签: recognition calculated Wavelet Subband

    上传时间: 2013-12-08

    上传用户:guanliya

  • 自己采用opencv编写的程序

    自己采用opencv编写的程序,该程序主要实现运动目标的检测,采用背景减法里面的GMM混合高斯模型

    标签: opencv 编写 程序

    上传时间: 2017-03-20

    上传用户:refent

  • 哈工大博士论文

    哈工大博士论文,基于HMM和ANN的汉语语音识别。

    标签: 论文

    上传时间: 2013-12-29

    上传用户:225588

  • 详细介绍了隐马尔科夫链的原理和matlab代码实现

    详细介绍了隐马尔科夫链的原理和matlab代码实现,可以运行其中的demo了解hmm的工作原理

    标签: matlab 详细介绍 代码 马尔科夫链

    上传时间: 2013-12-27

    上传用户:love_stanford

  • 隐含马尔可夫模型的入门资料

    隐含马尔可夫模型的入门资料,stanford机器学习课程资料 Introduction to the HMM model.

    标签: 马尔可夫模型

    上传时间: 2017-09-04

    上传用户:huangld

  • 这是一个模型介绍和常用算法的C语言的实现

    这是一个模型介绍和常用算法的C语言的实现,包过HMM算法,BP神经网络解决异或问题~~

    标签: 模型 C语言 算法

    上传时间: 2013-11-25

    上传用户:duoshen1989

  • 基于HMM的孤立字语音识别系统

    基于MATLAB的孤立词语音识别系统分析,可以参考一下

    标签: 孤立字

    上传时间: 2015-03-31

    上传用户:王金栋888

  • HMM code

    隐马尔科夫模型压缩包。。。隐马尔科夫模型的离散形式及连续形式的实现。。。

    标签: HMM

    上传时间: 2016-03-03

    上传用户:dsgadgad

  • Signal Processing for Telecommunications

    This paper presents a Hidden Markov Model (HMM)-based speech enhancement method, aiming at reducing non-stationary noise from speech signals. The system is based on the assumption that the speech and the noise are additive and uncorrelated. Cepstral features are used to extract statistical information from both the speech and the noise. A-priori statistical information is collected from long training sequences into ergodic hidden Markov models. Given the ergodic models for the speech and the noise, a compensated speech-noise model is created by means of parallel model combination, using a log-normal approximation. During the compensation, the mean of every mixture in the speech and noise model is stored. The stored means are then used in the enhancement process to create the most likely speech and noise power spectral distributions using the forward algorithm combined with mixture probability. The distributions are used to generate a Wiener filter for every observation. The paper includes a performance evaluation of the speech enhancer for stationary as well as non-stationary noise environment.

    标签: Telecommunications Processing Signal for

    上传时间: 2020-06-01

    上传用户:shancjb