• Journal of Platform Technology
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• 제8권4호
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• pp.20-28
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• 2020

본 논문에서는 시각장애인의 라이프 사이클을 사전에 학습하여 시각장애인의 자립생활을 돕는 적정기술로 음성인식 기반 스마트 웨어러블 디바이스, 스마트 기기 및 웹 AI서버를 포함하는 음성, 사물 및 문자 인식 플랫폼을 제안하였다. 시각장애인용 웨어러블 기기는 착용편의성과 사물인식기능 효율을 높이기 위해 리버스 넥밴드 구조로 설계하여 제작하였으며, 웨어러블 기기에 부착된 고감도 소형 마이크와 스피커는 웨어러블 기기와 연동된 스마트기기의 앱으로 구성된 음성인식 인터페이스 기능을 지원하도록 구성하였다. 음성, 사물 및 광학문자 인식 서비스는 웹 AI 서버에서 오픈소스 및 구글 API를 활용하였고, 서비스 플랫폼의 음성, 사물 및 광학문자 인식 정밀도는 실험을 통하여 평균 90%이상 달성하였음을 확인하였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(3)
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• pp.83-86
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• 2002

This paper intends to provide convenience to the disabled, losing the use of their limbs, through voice recognition technology. The voice recognition part of this system recognizes voice by DTW (Dynamic Time Warping) Which is most Widely used in Speaker dependent system. Specially, S/N rate was improved through Wiener filter in the pre-treatment phase while considering real environmental conditions; the result values of 12th order feature pattern per frame are extracted by DTW algorithm using LPC and Cepsturm in feature extraction process. Furthermore, miniaturization is pursued using TMS320C32, 71's the floating-point DSP, for the hardware part. Currently, 90% of hardware porting has been completed, but we can confirm that the recognition rate was 96% as a result of performing the DTW algorithm in PC.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제4권2호
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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.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.1027-1032
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• 1994

This paper introduces a interword modeling and a Viterbi search method for continuous speech recognition. We also describe a development of a real-time voice dialing system which can recognize around one hundred words and continuous digits in speaker independent mode. For continuous digit recognition, between-word units have been proposed to provide a more precise representation of word junctures. The best path in HMM is found by the Viterbi search algorithm, from which digit sequences are recognized. The simulation results show that a interword modeling using the context-dependent between-word units provide better recognition rates than a pause modeling using the context-independent pause unit. The voice dialing system is implemented on a DSP board with a telephone interface plugged in an IBM PC AT/486.

• 한국산업융합학회 논문집
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• 제21권1호
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• pp.17-27
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• 2018

We propose a new approach to control a biped robot motion based on iterative learning of voice command for the implementation of smart factory. The real-time processing of speech signal is very important for high-speed and precise automatic voice recognition technology. Recently, voice recognition is being used for intelligent robot control, artificial life, wireless communication and IoT application. In order to extract valuable information from the speech signal, make decisions on the process, and obtain results, the data needs to be manipulated and analyzed. Basic method used for extracting the features of the voice signal is to find the Mel frequency cepstral coefficients. Mel-frequency cepstral coefficients are the coefficients that collectively represent the short-term power spectrum of a sound, based on a linear cosine transform of a log power spectrum on a nonlinear mel scale of frequency. The reliability of voice command to control of the biped robot's motion is illustrated by computer simulation and experiment for biped walking robot with 24 joint.

• 인터넷정보학회논문지
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• 제24권2호
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• pp.11-18
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• 2023

최근 음성 감정 인식(Speech Emotion Recognition, SER)분야는 음성 특징과 모델링을 활용하여 인식률을 개선하기 위한 많은 연구가 진행되고 있다. 기존 음성 감정 인식의 정확도를 높이기 위한 모델링 연구 이외에도 음성 특징을 다양한 방법으로 활용하는 연구들이 진행되고 있다. 본 논문에서는 음성 감정이 시간 흐름과 연관이 있음을 착안하여 시계열 방식으로 음성파일을 시간 구간별로 분리한다. 파일 분리 이후, 음성 특징인 Mel, Chroma, zero-crossing rate (ZCR), root mean square (RMS), mel-frequency cepastral coefficients (MFCC)를 추출하여서 순차적 데이터 처리에 사용하는 순환형 신경망 모델에 적용하여 음성 데이터에서 감정을 분류하는 모델을 제안한다. 제안한 모델은 librosa를 사용하여 음성 특징들을 모든 파일에서 추출하여, 신경망 모델에 적용하였다. 시뮬레이션은 영어 데이터 셋인 Interactive Emotional Dyadic Motion Capture (IEMOCAP)을 이용하여 recurrent neural network (RNN), long short-term memory (LSTM) and gated recurrent unit(GRU)의 모델들의 성능을 비교 및 분석하였다.

• 대한음성학회지:말소리
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• 제44호
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• pp.115-130
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• 2002

Voice input is often required in many new application environments, but the low rate of speech recognition makes it difficult to extend its application. Previous approaches were to raise the accuracy of the recognition by post-processing of the recognition results, which were all lexical-oriented. We suggest a new semantic-oriented approach in speech recognition error correction. Through experiments using a speech-driven in-vehicle telematics information application, we show the excellent performance of our approach and some advantages it has as a semantic-oriented approach over a pure lexical-oriented approach.

• 음성과학
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• 제8권1호
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• pp.159-165
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• 2001

In this paper, we make an experiment on speech recognition systems with barging-in and non-barging-in utterances. Barging-in capability, with which we can say voice commands while voice announcement is coming out, is one of the important elements for practical speech recognition systems. Barging-in capability can be realized by echo cancellation techniques based on the LMS (least-mean-square) algorithm. We use three kinds of telephone interface boards with barging-in capability, which are respectively made by Dialogic Company, Natural MicroSystems Company and Korea Telecom. Speech database was made using these three kinds of boards. We make a comparative recognition experiment with this speech database.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2001년도 춘계학술발표논문집 (하)
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• pp.1177-1180
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• 2001

전화를 통해 인터넷 정보를 들을 수 있는 보이스 포탈(Voice Portal) 서비스가 인기를 얻고 있다. Voice Portal 서비스란 알고자 하는 정보를 Speech Recognition System에 음성으로 명령하면 전화를 통해 음성으로 원하는 정보를 듣는 서비스이다. Authentication의 절차를 수행하는 SMS (Short Message Service) 서버 Module, PSTN과 Database 서버사이의 Interface를 제공하는 CTI (Computer Telephony Integration) Module, CTI 서버와 WWW (World Wide Web) 사이의 Voice XML Module, 정보를 검색하기 위한 Searching Module들이 필요하다. 본 논문은 Speech Recognition technology를 기반으로 한 CTI Module 설계를 구현하였다. 또한 인정 방식으로 Random한 일회용 password를 기반으로 한 SMS Authentication을 택하므로 더욱 더 안정된 서비스 제공을 목적으로 하였다.

• 산업경영시스템학회지
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• 제28권3호
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• pp.133-141
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• 2005

The previous research has been mainly focused on individual technology of voice recognition, voice synthesis, translation, and bone transmission technical. Recently, commercial models have been produced using aforementioned technologies. In this research, a new automated translation support system concept has been proposed by combining established technology of bone transmission and wireless system. The proposed system has following three major components. First, the hybrid system consist of headset, bone transmission and other technologies will recognize user's voice. Second, computer recognized voice (using small server attached to the user) of the user will be converted into digital signal. Then it will be translated into other user's language by translation algorithm. Third, the translated language will be wirelessly transmitted to the other party. The transmitted signal will be converted into voice in the other party's computer using the hybrid system. This hybrid system will transmit the clear message regardless of the noise level in the environment or user's hearing ability. By using the network technology, communication between users can also be clearly transmitted despite the distance.