DocumentCode
3399863
Title
Comparison of LDM and HMM for an Application of a Speech
Author
Mane, Vikram A. ; Patil, Ajay B. ; Paradeshi, Kiran P.
Author_Institution
Dept. of E&TC, Annasaheb Dange COE, Ashta, India
fYear
2010
fDate
16-17 Oct. 2010
Firstpage
431
Lastpage
436
Abstract
Automatic speech recognition (ASR) has moved from science-fiction fantasy to daily reality for citizens of technological societies. Some people seek it out, preferring dictating to typing, or benefiting from voice control of aids such as wheel-chairs. Others find it embedded in their hi-tec gadgetry-in mobile phones and car navigation systems, or cropping up in what would have until recently been human roles such as telephone booking of cinema tickets. Wherever you may meet it, computer speech recognition is here, and it´s here to stay. Most of the automatic speech recognition (ASR) systems are based on hidden Markov Model in which Guassian Mixturess model is used. The output of this model depends on subphone states. Dynamic information is typically included by appending time-derivatives to feature vectors. This approach was quite successful. This approach makes the false assumption of framewise independence of the augmented feature vectors and ignores the spatial correlations in the parametrised speech signal. This is the short coming while applying HMM for acoustic modeling for ASR. Rather than modelling individual frames of data, LDMs characterize entire segments of speech. An auto-regressive state evolution through a continuous space gives a Markovian model. The underlying dynamics, and spatial correlations between feature dimensions. LDMs are well suited to modelling smoothly varying, continuous, yet noisy trajectories such as found in measured articulatory data.
Keywords
Kalman filters; hidden Markov models; speech recognition; Guassian Mixturess model; Markovian model; acoustic modeling; automatic speech recognition; autoregressive state evolution; car navigation system; dynamic information; feature vector; hi-tec gadgetry; hidden Markov Model; mobile phone; parametrised speech signal; subphone state; Data models; Equations; Hidden Markov models; Kalman filters; Noise measurement; Speech; Speech recognition; error covariance matrix; kalman gain; white noise;
fLanguage
English
Publisher
ieee
Conference_Titel
Advances in Recent Technologies in Communication and Computing (ARTCom), 2010 International Conference on
Conference_Location
Kottayam
Print_ISBN
978-1-4244-8093-7
Electronic_ISBN
978-0-7695-4201-0
Type
conf
DOI
10.1109/ARTCom.2010.65
Filename
5655590
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