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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.

    標簽: recognition calculated Wavelet Subband

    上傳時間: 2013-12-08

    上傳用戶:guanliya

  • 自己采用opencv編寫的程序

    自己采用opencv編寫的程序,該程序主要實現運動目標的檢測,采用背景減法里面的GMM混合高斯模型

    標簽: opencv 編寫 程序

    上傳時間: 2017-03-20

    上傳用戶:refent

  • 哈工大博士論文

    哈工大博士論文,基于HMM和ANN的漢語語音識別。

    標簽: 論文

    上傳時間: 2013-12-29

    上傳用戶:225588

  • 詳細介紹了隱馬爾科夫鏈的原理和matlab代碼實現

    詳細介紹了隱馬爾科夫鏈的原理和matlab代碼實現,可以運行其中的demo了解hmm的工作原理

    標簽: matlab 詳細介紹 代碼 馬爾科夫鏈

    上傳時間: 2013-12-27

    上傳用戶:love_stanford

  • 隱含馬爾可夫模型的入門資料

    隱含馬爾可夫模型的入門資料,stanford機器學習課程資料 Introduction to the HMM model.

    標簽: 馬爾可夫模型

    上傳時間: 2017-09-04

    上傳用戶:huangld

  • 這是一個模型介紹和常用算法的C語言的實現

    這是一個模型介紹和常用算法的C語言的實現,包過HMM算法,BP神經網絡解決異或問題~~

    標簽: 模型 C語言 算法

    上傳時間: 2013-11-25

    上傳用戶:duoshen1989

  • 基于HMM的孤立字語音識別系統

    基于MATLAB的孤立詞語音識別系統分析,可以參考一下

    標簽: 孤立字

    上傳時間: 2015-03-31

    上傳用戶:王金棟888

  • HMM code

    隱馬爾科夫模型壓縮包。。。隱馬爾科夫模型的離散形式及連續形式的實現。。。

    標簽: HMM

    上傳時間: 2016-03-03

    上傳用戶: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.

    標簽: Telecommunications Processing Signal for

    上傳時間: 2020-06-01

    上傳用戶:shancjb

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