• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.9-12
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• 2000

Voice dialing system can recognize the speaker's command and dial the destinate phone number automatically. Such a system is useful for wireless handsets and portable communication devices. As a personal voice dialing system, all the commands are used to train the HMM for speech recognition based on owner-selected phrases. Its implementation requires much less memory space and computation resource compared to a speaker-independent system. Since only two or three training utterances per command are used in this system, it is difficult to estimate exact state duration distribution to improve the recognition performance. Therefore a post-processor is presented to improve the performance. Experiments which use the database collected through the telephone line showed that the proposed post-processor improves the recognition system performance.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.4 no.2
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• pp.176-181
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• 2004

This paper introduces a Web site login system that uses user's native voice to improve the bother of remembering the ID and password in order to login the Web site. The DTW method that applies fuzzy inference is used as the speaker recognition algorithm. We get the ACC(Average Cepstrum Coefficient) membership function by each degree, by using the LPC that models the vocal chords, to block the recorded voice that is problem for the speaker recognition. We infer the existence of the recorded voice by setting on the basis of the number of zeros that is the value of the ACC membership function, and on the basis of the average value of the ACC membership function. We experiment the six Web sites for the six subjects and get the result that protects the recorded voice about 98% that is recorded by the digital recorder.

• Knowledge Management Research
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• v.19 no.1
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• pp.147-174
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• 2018

Recently, firms have been putting forth significant efforts to fulfill various demands and high expectations of customers. The role and importance of customer centers as a direct contact point for customer relationship management are more emphasized than previously. A customer center draws attention as a new alternative to secure corporate competitiveness as it contributes to sales increase, being in a position to satisfy customers' needs by ensuring customers' access to information. A customer center is an aggregation of various information and communication technologies. In particular, a voice recognition/analysis technology based on big data can elaborate customer services further, enhance customer satisfaction, and trigger constant interactions with customers. A customer center can be transformed to a hub of customer knowledge and the embodiment of business intelligence in the front line of business. This article is a case study on how the customer center of the K life insurance company regarding customer center operation collects and analyzes customer information and how it has established its voice recognition/analysis system based on big data to improve customer experience management. Factors affecting the successful introduction and implementation of voice recognition/analysis system to a firm, are examined.

• Journal of Sensor Science and Technology
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• v.13 no.1
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• pp.20-26
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• 2004

For the disabled persons who can't use their limbs, motorized wheelchair that is activated by voice recognition module employing speaker independent method, was implemented. The wireless voice transfer device was designed and employed for the user convenience. And the wheelchair was designed to operate using voice and keypad by selection of the user because they can manipulate it using keypad if necessary. The speaker independent method was used as the voice recognition module in order that anyone can manipulate the wheelchair in case of assistance. Using the implemented wheelchair, performance and motion of the system was examined and it has over than 97% of voice recognition rate and proper movements.

• Proceedings of the KIEE Conference
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• 2004.05a
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• pp.147-149
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• 2004

Although most of the people have studied English for a long time, their English conversation capability is low. When we provide them portable conversational-type English learners by the application of computer and information process technology, such portable learners can be used to enhance their English conversation capability by their conventional conversation exercises. The core technology to develop such learner is the development of a voice recognition and synthesis module under an embedded environment. This paper deals with voice recognition and synthesis, prototype of the learner module using a DSP(Digital Signal Processing) chip for voice processing, voice playback function, flash memory file system, PC download function using USB ports, English conversation text function by the use of SMC(Smart Media Card) flash memory, LCD display function, MP3 music listening function, etc. Application areas of the prototype equipped with such various functions are vast, i.e. portable language learners, amusement devices, kids toy, control by voice, security by the use of voice, etc.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.28B no.7
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• pp.548-556
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• 1991

In this paper, Korean spoken continuous digits are ercognized using DHMM(Dynamic Hidden Markov Model) and lexical analysis to provide the base of developing voice dialing system. After segmentation by phoneme unit, it is recognized. This system can be divided into the segmentation section, the design of standard speech section, the recognition section, and the lexical analysis section. In the segmentation section, it is segmented using the ZCR, O order LPC cepstrum, and Ai, parameter of voice speech dectaction, which is changed according to time. In the standard speech design section, 19 phonemes or syllables are trained by DHMM and designed as a standard speech. In the recognition section, phomeme stream are recognized by the Viterbi algorithm.In the lexical decoder section, finally recognized continuous digits are outputed. This experiment shiwed the recognition rate of 85.1% using data spoken 7 times of 21 classes of 7 continuous digits which are combinated all of the occurence, spoken by 10 man.

• Information and Communications Magazine
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• v.10 no.10
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• pp.22-29
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• 1993

This paper describes development of a real-time voice recognition dialing system which can recognize around one hundred word vocabularies in speaker independent mode. The voice recognition algorithm is implemented on a DSP board with a telephone interface plugged in an IBM PC AT/486. In the DSP board, procedures for feature extraction, vector quantization(VQ), and end-point detection are performed simultaneously in every 10msec frame interval to satisfy real-time constraints after the word starting point detection. In addition, we optimize the VQ codebook size and the end-point detection procedure to reduce recognition time and memory requirement. The demonstration system is being displayed in MOBILAB of Korea Mobile Telecom at the Taejon EXPO '93.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.949-958
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• 2018

A voice recognition control system is a system for controlling a peripheral device by recognizing a voice. Recently, a voice recognition control system have been applied not only to smart devices but also to various environments ranging from IoT(: Internet of Things), robots, and vehicles. In such a voice recognition control system, the recognition rate is lowered due to the ambient noise in addition to the voice of the user. In this paper, we propose a dual channel acoustic beamforming hardware architecture for MEMS(: Microelectromechanical Systems) microphones to eliminate ambient noise in addition to user's voice. And the proposed hardware architecture is designed as ASIC(: Application-Specific Integrated Circuit) using TowerJazz $0.18{\mu}m$ CMOS(: Complementary Metal-Oxide Semiconductor) technology. The designed dual channel acoustic beamforming ASIC has a die size of $48mm^2$, and the directivity index of the user's voice were measured to be 4.233㏈.

• The KIPS Transactions:PartB
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• v.14B no.3 s.113
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• pp.183-190
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• 2007

VoiceXML is new markup language which is designed for web resource navigation via voice based on XML. An application using VoiceXML is classified into mutual-controlled and machine-controlled form dialog structure. Such dialog structures can't construct service which provide free navigation of web resource by user because a scenario is decided by application developer. In this paper, we propose VoiceXML application structure using user-controlled form dialog system which decide service scenario according to user's intention. The proposed application automatically detects recognition candidates from requested information by user, and then system uses recognition candidate as voice-anchor. Also, system connects each voice-anchor with new voice-node. An example of proposed system, we implement news service with IT term dictionary, and we confirm detection and registration of voice-anchor and make an estimate of hit rate about measurement of an successive offer from information according to user's intention and response speed. As the experiment result, we confirmed possibility which is more freely navigation of web resource than existing VoiceXML form dialog systems.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1413-1415
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• 2004

This paper proposes a SVM(Support Vector Machine) training algorithm to control a service robot with voice command. The service robot with a stereo vision system and dual manipulators of four degrees of freedom implements a User-Dependent Voice Control System. The training of SVM algorithm that is one of the statistical learning theories leads to a QP(quadratic programming) problem. In this paper, we present an efficient SVM speech recognition scheme especially based on less learning data comparing with conventional approaches. SVM discriminator decides rejection or acceptance of user's extracted voice features by the MFCC(Mel Frequency Cepstrum Coefficient). Among several SVM kernels, the exponential RBF function gives the best classification and the accurate user recognition. The numerical simulation and the experiment verified the usefulness of the proposed algorithm.