• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.6
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• pp.714-719
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• 2005

The paper attempts to design and implement the system which can remotely check visitors' speech or Image by a mobile phone. This system is designed to recognize who a visitor is through the automatic calling service, not through a short message, via the mobile phone, even when the home owner is outside. In general, door locks are controlled through the home Server, but it is more effective to control door locks by using DTMF signal from a real-time point of view. The technology suggested in this paper makes it possible to communicate between the visiter and the home owner by making a phone call to tile home owner's mobile phone automatically when the visiter visits the house even if the home owner is outside, and if necessary, it allows for the home owner to control the door lock remotely. Thanks to the system, the home owner is not restricted by time or space for checking the visitor's identification and controlling the door lock. In addition, the security system is improved by changing from the existing password form to the combination of password and speaker verification lot the verification procedure required for controlling the door lock and setting the environment under consideration of any disadvantages which may occur when the mobile Phone is lost. Also, any existing problems such as reconnection to tile network for controlling tile door lock are solved by controlling the door lock in real time by use of DTMF signal while on the phone.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.51-56
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• 2000

Is this paper, we studied on the automatic speech control system in real-time windows environment using voice recognition. The applied reference pattern is the variable DMS model which is proposed to fasten execution speed and the one-stage DP algorithm using this model is used for recognition algorithm. The recognition vocabulary set is composed of control command words which are frequently used in windows environment. In this paper, an automatic speech period detection algorithm which is for on-line voice processing in windows environment is implemented. The variable DMS model which applies variable number of section in consideration of duration of the input signal is proposed. Sometimes, unnecessary recognition target word are generated. therefore model is reconstructed in on-line to handle this efficiently. The Perceptual Linear Predictive analysis method which generate feature vector from extracted feature of voice is applied. According to the experiment result, but recognition speech is fastened in the proposed model because of small loud of calculation. The multi-speaker-independent recognition rate and the multi-speaker-dependent recognition rate is 99.08% and 99.39% respectively. In the noisy environment the recognition rate is 96.25%.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.8
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• pp.726-734
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• 2007

Generally, a mobile robot is moved by original input programs. However, it is very hard for a non-expert to change the program generating the moving path of a mobile robot, because he doesn't know almost the teaching command and operating method for driving the robot. Therefore, the teaching method with speech command for a handicapped person without hands or a non-expert without an expert knowledge to generate the path is required gradually. In this study, for easily teaching the moving path of the autonomous mobile robot, the autonomous mobile robot with the function of speech recognition is developed. The use of human voice as the teaching method provides more convenient user-interface for mobile robot. To implement the teaching function, the designed robot system is composed of three separated control modules, which are speech preprocessing module, DC servo motor control module, and main control module. In this study, we design and implement a speaker dependent isolated word recognition system for creating moving path of an autonomous mobile robot in the unknown environment. The system uses word-level Hidden Markov Models(HMM) for designated command vocabularies to control a mobile robot, and it has postprocessing by neural network according to the condition based on confidence score. As the spectral analysis method, we use a filter-bank analysis model to extract of features of the voice. The proposed word recognition system is tested using 33 Korean words for control of the mobile robot navigation, and we also evaluate the performance of navigation of a mobile robot using only voice command.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1291-1295
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• 2004

This paper proposed a new approach for endpoint detection of isolated speech, which proves to significantly improve the endpoint detection performance. The proposed algorithm relies on the root mean square energy (rms energy), zero crossing rate and spectral characteristics of the speech signal where the Euclidean distance measure is adopted using cepstral coefficients to accurately detect the endpoint of isolated speech. The algorithm offers better performance than traditional energy-based algorithm. The vocabulary for the experiment includes English digit from one to nine. These experimental results were conducted by 360 utterances from a male speaker. Experimental results show that the accuracy of the algorithm is quite acceptable. Moreover, the computation overload of this algorithm is low since the cepstral coefficients parameters will be used in feature extraction later of speech recognition procedure.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1227-1230
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• 1998

Phase control methods are used to expand the sound image in general AV system. However, these methods are effective only to the signal under 1kHz, and the listener must be located in front center of the speaker system. In this paper, we realize the realtime processing system in which phase shifting method is dominant at low frequency and precedence effect is dominant at high frequency. Two sound cards are used to process the audio signal in realtime with 16 bits stereo channel of 44.1 kHz sampling frequency. And the analog circuit is designed to process the phase shifting. In experiments the usefulness of the proposed stereo system is confirmed.

• MALSORI
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• no.15_18
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• pp.19-34
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• 1989

There is general agreement that voice onset time (VOT) is functionally related with the glottal opening at the moment of the oral release of a stop. However, systematic investigations of tempo 8n4 the place of articulation as affecting the glottal opening and VOT have relatively neglected. Various instrumental techniques were used to verify the claim with BrEng and korean speakers, under controlled experimental conditions, tempo being one of them. It was found that voiceless aspiration (i.e. VOT) is not simply a function of the glottal area at the moment of the oral release of a stop as it is normally defined in the existing literature. Within a given place of articulation and across temper VOT was generally insignificantly related to the glottal area. It is inferred that the glottal adduction onset time for the following vowel is actively control led by the speaker to meet aerodynamic requirements in relation to class (i.e. aspirated and unaspirated) and tempo. Some possible underlying physiological mechanisms for various phonetic aspects of intervocalic stops, associated with the glottal area and VOT, were discussed.

• Korean Journal of Cleft Lip And Palate
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• v.5 no.1
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• pp.1-9
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• 2002

Some children with Cleft Palate have shown a speech disorders after repaired surgical operation. A diagnostic evaluation of speech in children with cleft palates is important in preventing speech disorders. However, standard speech evaluation form for children with cleft palates has not yet developed in Korea. The purpose of this study is to make the standard speech evaluation form for children with cleft palates. Thirty control children group and ten children with cleft palate participated in this experiment. The test words are composed of meaningless two syllabic words containing the three different types of korean stop consonants,

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1998.04a
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• pp.704-709
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• 1998

An automated calibration system of sound level meters was developed and tested. As a standard sound source, the speaker unit(Forstex FE208) cabineted by 440 * 390 * 490 mm$^{3}$(LHW) volume wood box was adopted. Including this source, the driving part was found out to have a good linearity of sound pressure output vs AC input. We use the Hybrid-Bisect, /Newton-Raphson method modified by the linearity as searching algorithm. Personal computer and program do the control, measurements, and calculations and finally do the accumulation of useful data and results. Several trials of automatic calibration using this developed system give reliable results.

• 제어로봇시스템학회:학술대회논문집
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• 1993.10a
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• pp.870-876
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• 1993

This paper presents a binary clustering network (BCN) and a heuristic algorithm to detect pitch for recognition of keywords in continuous speech. In order to classify nonlinear patterns, BCN separates patterns into binary clusters hierarchically and links same patterns at root level by using the supervised learning and the unsupervised learning. BCN has many desirable properties such as flexibility of dynamic structure, high classification accuracy, short learning time, and short recall time. Pitch Detection algorithm is a heuristic model that can solve the difficulties such as scaling invariance, time warping, time-shift invariance, and redundance. This recognition algorithm has shown recognition rates as high as 95% for speaker-dependent as well as multispeaker-dependent tests.

• 제어로봇시스템학회:학술대회논문집
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• 1996.10b
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• pp.399-402
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• 1996

The electrodynamic loudspeakers should have a wide dynamic range to reproduce various sound levels. When the input signal is small, the radiated sound from the loudspeaker is not so much distorted. However, for large input signal with low frequency component the radiated sound is significantly distorted due to the nonlinearities of the loudspeaker. The suspension, damping, and magnetic flux of loudspeaker are the main sources of the nonlinearity. Such electromechanical parameters related to harmonic distortion have been represented by a polynomial model for diaphragm displacement, while each of the polynomial coefficient is evaluated by using the principle of harmonic balance experimentally. Based on the polynomial model, we designed a compensator for nonlinear harmonic distortion of direct radiator loudspeaker. Than observer is used to estimate the displacement of the loudspeaker diaphragm, which is rather difficult to measure directly in the conventional setting. The usefulness of the designed compensator is demonstrated by numerical simulations. Simulation results show about 30db decrease at the second and third higher harmonic distortions. We carry out an experiment on speaker to verify designed controller and nonlinear observer.