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35973510VAD-DTW-HMM

于 2017-12-14 发布 文件大小:531KB
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下载积分: 1 下载次数: 12

代码说明:

  基于动态时间规整与隐马尔科夫模型的自动语音识别系统(Automatic speech recognition system based on dynamic time warping and hidden Markov model)

文件列表:

ch13
ch13\cdhmm
ch13\cdhmm\baum.m, 1511, 2001-11-12
ch13\cdhmm\getparam.m, 1990, 2002-02-28
ch13\cdhmm\hmm.mat, 63856, 2001-11-12
ch13\cdhmm\inithmm.m, 1266, 2002-02-28
ch13\cdhmm\main.m, 165, 2001-11-13
ch13\cdhmm\mfcc.m, 859, 2001-11-09
ch13\cdhmm\mixture.m, 357, 2001-11-11
ch13\cdhmm\pdf.m, 230, 2001-11-10
ch13\cdhmm\recog.m, 255, 2001-11-13
ch13\cdhmm\samples.mat, 630000, 2001-11-12
ch13\cdhmm\train.m, 839, 2001-11-13
ch13\cdhmm\vad.m, 1855, 2001-11-09
ch13\cdhmm\viterbi.m, 1026, 2001-11-11
ch13\dtw
ch13\dtw\dtw.m, 494, 2001-11-09
ch13\dtw\dtw2.m, 1648, 2001-11-09
ch13\dtw\mfcc.m, 859, 2001-11-09
ch13\dtw\testdtw.m, 647, 2001-11-09
ch13\dtw\vad.m, 1855, 2001-11-09
ch13\record
ch13\record\record.fig, 19008, 2001-11-12
ch13\record\record.m, 3974, 2001-11-12
ch13\record\sample.m, 147, 2001-11-12
www.pudn.com.txt, 218, 2007-01-22

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    这是语音信号处理中求倒谱的程序,我自己编的,很简单,实现了一段序列求倒谱的功能(This is the voice signal processing for cepstrum procedures, I developed, is very simple. achieve a sequence for the functional cepstrum)
    2006-10-11 20:12:24下载
    积分:1
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    为均衡带限信号所引起失真的横向或格型自适应均衡器(其中横向FIR系统长M=11), 系统输入是取值为±1的随机序列 ,其均值为零;参考信号 ;信道具有脉冲响应: 式中w用来控制信道的幅度失真(w = 2~4,例如,取w = 2.9,3.1,3.3,3.5等),而且信道受到均值为零、方差为 (例如,取 ,相当于信噪比为30dB)的高斯白噪声 的干扰。试比较基于下列五种算法自适应均衡器在不同信道失真、不同噪声干扰下的收敛情况(对应于每一种情况,在同一坐标下画出其学习曲线): 横向/格-梯型结构LMS算法[1-4] 横向/格-梯型结构RLS算法[1-4] ()
    2007-09-12 20:18:58下载
    积分:1
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    1、文件夹中包含了经典功率谱估计和自适应均衡算法两个实验的所有程序。 2、R.m、LMS.m、LMSmain.m为自适应均衡算法的程序: R.m用来计算输入信号的自相关矩阵及其特征值; LMS.m为时域LMS算法,用统计的方法仿真得出不同信道参数和不同步长下的学习曲线; LMSmain.m为实验主程序,按照实验要求中的具体数据得到实验结果和曲线。 3、functionx.m、fzhouqitu.m、spectrum.m、bt.m、bart_lett.m、welch.m、SPECTRUMmain.m为经典谱估计的程序: functionx.m产生需要进行谱估计的函数; fzhouqitu.m用来计算信号周期图的函数; spectrum.m是用周期图法进行谱估计的函数 bt.m是用BT图法进行谱估计的函数 bart_lett.m是用BARTLETT法进行谱估计的函数 welch.m是用WELCH法进行谱估计的函数 SPECTRUMmain.m是主程序,按照实验要求中的具体数据得到实验结果和曲线。 (a document folder contains the classic power spectrum estimation and adaptive equalization algorithm for the two experiments all the procedures. 2, R.m, LMS.m, LMSmain.m adaptive equalization algorithm for the procedure : R.m used to calculate the input signal and the correlation matrix eigenvalue; LMS.m too LMS algorithm, use statistical simulation method come to a different channel parameters and the synchronous learning curve; LMSmain.m main program for the experiment, according to the experimental requirements of the specific data and the curve of the experimental results. 3, functionx.m, fzhouqitu.m, spectrum.m. bt.m, bart_lett.m, welch.m. SPECTRUMmain.m classical spectrum estimation procedures : functionx.m have a need for spectrum estimation function; fzhouqitu.m used to calculate t)
    2007-04-29 17:54:54下载
    积分:1
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    G711标准的语音编解码算法。以便于对语音信号进行大量存储(speech coding)
    2010-10-10 20:15:03下载
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    2012-12-03 15:41:11下载
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    Praat4610 语音合成及分析源代码(Praat4610 speech synthesis and analysis of source code)
    2008-07-18 18:04:29下载
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    2013-04-02 16:51:36下载
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