• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2006.11a
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• pp.283-288
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• 2006

In the future television broadcasting a flood of information from various sources will not always be welcomed by everyone. The need of accessing specific information as required is becoming a necessity. We are interested to make the life of television consumer easier by providing an intelligent television set which can adaptively proposed certain shows to the viewer based on the user historical consumed shows. The TV watching history data consists of TV program titles with their respective genres, channels, watched times and durations, etc. The method proposed is by utilizing Hidden Markov Model (HMM) to model the user preference of kind of genres the viewer will watch based on recorded genres of several weeks time. We take watching schedule from 6 PM to midnight as boundary. The range thus divided into 3 independent time band of 2 hours each resulting in 3 time bands from 6 PM to 8 PM, 8 PM to 10 PM, and lastly 10 PM to midnight. Each time band will be represented by an HMM. From each HMM we can generate a sequence of predicted genre that the user will probably watch during corresponding time-band. Our approach assumes that the user shows a consistent behavior of watching pattern in week to week basis and during the moment of watching TV. To asses the method performance experiment is conducted using real data collected from December 2002 to May 2003. Some user's data are selected and based on that predictions are made. The resulting predictions are then compared with the actual user's history. The experiment shows satisfactory result for user with middle to high consistent behavior level.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.455-461
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• 2002

This paper presents a method that improves the performance of the personal voice dialling system in which speaker independent phoneme HMM's are used. Since the speaker independent phoneme HMM based voice dialing system uses only the phone transcription of the input sentence, the storage space could be reduced greatly. However, the performance of the system is worse than that of the system which uses the speaker dependent models due to the phone recognition errors generated when the speaker independent models are used. In order to solve this problem, a new method that jointly estimates transformation vectors for the speaker adaptation and transcriptions from training utterances is presented. The biases and transcriptions are estimated iteratively from the training data of each user with maximum likelihood approach to the stochastic matching using speaker-independent phone models. Experimental result shows that the proposed method is superior to the conventional method which used transcriptions only.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.20-26
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• 2000

In speaker verification system the HMM(hidden Markov model) parameter updating using small amount of data and the priori threshold decision are crucial factor for dealing with long-term variability in people voices. In the paper we present the speaker model updating technique which can be adaptable to the session-to-intra speaker variability and the priori threshold determining technique. The proposed technique decreases verification error rates which the session-to-session intra-speaker variability can bring by adapting new speech data to speaker model parameter through Baum Welch re-estimation. And in this study the proposed priori threshold determining technique is decided by a hybrid score measurement which combines the world model based technique and the cohen model based technique together. The results show that the proposed technique can lead a better performance and the difference of performance is small between the posteriori threshold decision based approach and the proposed priori threshold decision based approach.

• Journal of the Korea Society of Computer and Information
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• v.11 no.2 s.40
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• pp.159-167
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• 2006

Lipreading has been suggested as one of the methods to improve the performance of speech recognition in noisy environment. However, existing methods are developed and implemented only in software. This paper suggests a hardware design for real-time lipreading. For real-time processing and feasible implementation, we decompose the lipreading system into three parts; image acquisition module, feature vector extraction module, and recognition module. Image acquisition module capture input image by using CMOS image sensor. The feature vector extraction module extracts feature vector from the input image by using parallel block matching algorithm. The parallel block matching algorithm is coded and simulated for FPGA circuit. Recognition module uses HMM based recognition algorithm. The recognition algorithm is coded and simulated by using DSP chip. The simulation results show that a real-time lipreading system can be implemented in hardware.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.45 no.5
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• pp.8-14
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• 2008

There are two a approaches for the Cognitive Radio(CR) development. One is 'Full CR', which Joseph Mitola III proposed, and another is 'Spectrum CR', which is currently being standardized. The target approach of this paper is the latter and we develop a Cognitive Engine(CE) and simulated a channel set management(CSM), which is a core function of CE. The Channel set management evaluates channel quality and Incumbent User(IU) vacancy possibility and classifies the channel set, which is performed by using channel state history. Especially, a very important function for the channel set management is a channel state prediction and this paper proposed a Hidden Markov Model(HMM) based channel state prediction and a method for increasing performance. Also, we applied the proposed method into our simulator and simulated channel state prediction. Through the simulation, we verified as we applied our proposed scheme, the performance of channel state prediction gets better and through comparing with RS and SS, we verified the HMM based Channel state prediction is better.

• Journal of KIISE:Software and Applications
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• v.34 no.11
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• pp.1001-1010
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• 2007

HMM tends to output high probability for not only the proper class data but confusable class data, since the modeling power increases as the number of parameters increases. Thus it may not be helpful for discrimination to simply increase the number of parameters of HMM. We proposed two methods in this paper. One is a CMC(Confusion Likelihood Model Selection Criterion) using confusion class data probability, the other is a new recognition method, RCM(Recognition Using Confusion Models). In the proposed recognition method, confusion models are constructed using confusable class data, then confusion models are used to depress misrecognition by confusion likelihood is subtracted from the corresponding standard model probability. We found that CMC showed better results using fewer number of parameters compared with ML, ALC2, and BIC. RCM recorded 93.08% recognition rate, which is 1.5% higher result by reducing 17.4% of errors than using standard model only.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.5
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• pp.696-700
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• 2007

A new method that improves the performance of the phonetic transcription based speech recognition system is presented with the speaker-independent phonetic recognizer. Since SI phoneme HMM based speech recognition system uses only the phoneme transcription of the input sentence, the storage space could be reduced greatly. However, the performance of the system is worse than that of the speaker dependent system due to the phoneme recognition errors generated from using SI models. A new training method that iteratively estimates the phonetic transcription and transformation vectors is presented to reduce the mismatch between the training utterances and a set of SI models using speaker adaptation techniques. For speaker adaptation the stochastic matching methods are used to estimate the transformation vectors. The experiments performed over actual telephone line shows that a reduction of about 45% in the error rates could be achieved as compared to the conventional method.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.3
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• pp.143-150
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• 2016

In this paper, we propose Korean monophthong recognition method optimizing muscle mixing based on facial surface EMG signals. We observed that EMG signal patterns and muscle activity may vary according to Korean monophthong pronunciation. We use RMS, VAR, MMAV1, MMAV2 which were shown high recognition accuracy in previous study and Cepstral Coefficients as feature extraction algorithm. And we classify Korean monophthong by QDA(Quadratic Discriminant Analysis) and HMM(Hidden Markov Model). Muscle mixing optimized using input data in training phase, optimized result is applied in recognition phase. Then New data are input, finally Korean monophthong are recognized. Experimental results show that the average recognition accuracy is 85.7% in QDA, 75.1% in HMM.

• The Journal of the Acoustical Society of Korea
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• v.25 no.7
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• pp.312-318
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• 2006

The conventional lip-synch system has a two-step process, speech segmentation and recognition. However, the difficulty of speech segmentation procedure and the inaccuracy of training data set due to the segmentation lead to a significant Performance degradation in the system. To cope with that, the connected vowel recognition method using Head-Body-Tail (HBT) model is proposed. The HBT model which is appropriate for handling relatively small sized vocabulary tasks reflects co-articulation effect efficiently. Moreover the 7 vowels are merged into 3 classes having similar lip shape while the system is optimized by employing a class dependent SCHMM structure. Additionally in both end sides of each word which has large variations, 8 components Gaussian mixture model is directly used to improve the ability of representation. Though the proposed method reveals similar performance with respect to the CHMM based on the HBT structure. the number of parameters is reduced by 33.92%. This reduction makes it a computationally efficient method enabling real time operation.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.175-182
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• 2004

There is a growing interest in speaker verification, which verifies someone by his/her voices. There are many speaker vitrification algorithms such as HMM and DTW. However, it is impossible to apply these algorithms to memory limited applications because of large number of feature vectors to register or verify users. In this paper we introduces a speaker verification system using SVM, which needs a little memory usage and computation time. Also we proposed hardware architecture for SVM. Experiments were conducted with Korean database which consists of four-digit strings. Although the error rate of SVM is slightly higher than that of HMM, SVM required much less computation time and small model size.