• Title, Summary, Keyword: Voice Recognition

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Variation of the Verification Error Rate of Automatic Speaker Recognition System With Voice Conditions (다양한 음성을 이용한 자동화자식별 시스템 성능 확인에 관한 연구)

  • Hong Soo Ki
    • MALSORI
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    • no.43
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    • pp.45-55
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    • 2002
  • High reliability of automatic speaker recognition regardless of voice conditions is necessary for forensic application. Audio recordings in real cases are not consistent in voice conditions, such as duration, time interval of recording, given text or conversational speech, transmission channel, etc. In this study the variation of verification error rate of ASR system with the voice conditions was investigated. As a result in order to decrease both false rejection rate and false acception rate, the various voices should be used for training and the duration of train voices should be longer than the test voices.

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A Study on Intelligent Control Algorithm Development for Cooperation Working of Human and Robot (인간과 로봇 협력작업을 위한 로봇 지능제어알고리즘 개발에 관한 연구)

  • Lee, Woo-Song;Jung, Yang-Guen;Park, In-Man;Jung, Jong-Gyu;Kim, Hui-Jin;Kim, Min-Seong;Han, Sung-Hyun
    • Journal of the Korean Society of Industry Convergence
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    • v.20 no.4
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    • pp.285-297
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    • 2017
  • This study proposed a new approach to develop an Intelligent control algorithm for cooperative working of human and robot based on voice recognition. In general case of speaker verification, Gaussian Mixture Model is used to model the feature vectors of reference speech signals. On the other hand, Dynamic Time Warping based template matching techniques were presented for the voice recognition about several years ago. We converge these two different concepts in a single method and then implement in a real time voice recognition enough to make reference model to satisfy 95% of recognition performance. In this paper it was illustrated the reliability of voice recognition by simulation and experiments for humanoid robot with 18 joints.

Design And Implementation of a Speech Recognition Interview Model based-on Opinion Mining Algorithm (오피니언 마이닝 알고리즘 기반 음성인식 인터뷰 모델의 설계 및 구현)

  • Kim, Kyu-Ho;Kim, Hee-Min;Lee, Ki-Young;Lim, Myung-Jae;Kim, Jeong-Lae
    • The Journal of The Institute of Internet, Broadcasting and Communication
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    • v.12 no.1
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    • pp.225-230
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    • 2012
  • The opinion mining is that to use the existing data mining technology also uploaded blog to web, to use product comment, the opinion mining can extract the author's opinion therefore it not judge text's subject, only judge subject's emotion. In this paper, published opinion mining algorithms and the text using speech recognition API for non-voice data to judge the emotions suggested. The system is open and the Subject associated with Google Voice Recognition API sunwihwa algorithm, the algorithm determines the polarity through improved design, based on this interview, speech recognition, which implements the model.

The research on the MEMS device improvement which is necessary for the noise environment in the speech recognition rate improvement (잡음 환경에서 음성 인식률 향상에 필요한 MEMS 장치 개발에 관한 연구)

  • Yang, Ki-Woong;Lee, Hyung-keun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.22 no.12
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    • pp.1659-1666
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    • 2018
  • When the input sound is mixed voice and sound, it can be seen that the voice recognition rate is lowered due to the noise, and the speech recognition rate is improved by improving the MEMS device which is the H / W device in order to overcome the S/W processing limit. The MEMS microphone device is a device for inputting voice and is implemented in various shapes and used. Conventional MEMS microphones generally exhibit excellent performance, but in a special environment such as noise, there is a problem that the processing performance is deteriorated due to a mixture of voice and sound. To overcome these problems, we developed a newly designed MEMS device that can detect the voice characteristics of the initial input device.

The University Gusdance System using the Alexa (Alexa를 이용한 대학안내 시스템)

  • Kim, Tae Jin;Kim, Dong Hyun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.21 no.11
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    • pp.2061-2066
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    • 2017
  • The voice recognition technology is to recognize the voice of an user and execute the command. Recently, the voice recognition is evolving to the artificial intelligence voice recognition by adding the scheme of the natural language processing. The AI voice recognition is exploited to control the IoT devices or provide the information, such as the news or the wether. The University Information which is one of fields serviced by the information provider is mainly presented on the web. However, since too much information are presented on the web, it is difficult for an user to find efficiently the specific information which the user want to know. In this paper, we design and implement the university guidance system to recognize the user voice searching the information and provide the result using the voice. To do this, we classify the university data and design the lambda function to provide the data.

Environments of Hoarseness in Children (소아애성에 영향을 주는 환경에 대한 연구)

  • 안철민;박상준;이건영
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.8 no.2
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    • pp.173-177
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    • 1997
  • The speech movements are acquired activity, not determined by instincts or by biologic inheritance either. The child listens to the sound from the surrounding persons, observes the speech movement of the people and tried to imitate them. Then the child acquires their specific phonation pattern. We guessed that the parents influences to the child are very important in the developing of the speech movements. Because the parents are first contact person to the baby. The recognition of parents about the voice changes in the child will be important too. And social environments such as kindergarden, school, friends contact with, can influence to the voice of the child. We investigated the state of the voice, parents influence and social environmental factor. In the bases of this study, we knew that the parents recognition about the voice changes of child, faulty vocal habits of child, social environmental factors influenced to the voice of child. And we thought we have to do our best for the early detection of voice changes and proper treatment.

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Speech Intelligibility of Alaryngeal Voices and Pre/Post Operative Evaluation of Voice Quality using the Speech Recognition Program(HUVOIS) (음성인식프로그램을 이용한 무후두 음성의 말 명료도와 병적 음성의 수술 전후 개선도 측정)

  • Kim, Han-Su;Choi, Seong-Hee;Kim, Jae-In;Lee, Jae-Yol;Choi, Hong-Shik
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.15 no.2
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    • pp.92-97
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    • 2004
  • Background and Objectives : The purpose of this study was to examine objectively pre and post operative voice quality evaluation and intelligibility of alaryngeal voice using speech recognition program, HUVOIS. Materials and Methods : 2 laryngologists and 1 speech pathologist were evaluated 'G', 'R', 'B' in the GRBAS sclae and speech intelligibility using NTID rating scale from standard paragraph. And also acoustic estimates such as jitter, shimmer, HNR were obtained from Lx Speech Studio. Results : Speech recognition rate was not significantly different between pre and post operation for pathological vocie samples though voice quality(G, B) and acoustic values(Jitter, HNR) were significantly improved after post operation. In Alaryngeal voices, reed type electrolarynx 'Moksori' was the highest both speech intelligibility and speech recognition rate, whereas esophageal speech was the lowest. Coefficient correlation of speech intelligibility and speech recognition rate was found in alaryngeal voices, but not in pathological voices. Conclusion : Current study was not proved speech recognition program, HUVOIS during telephone program was not objective and efficient method for assisting subjective GRBAS scale.

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Real-time Phoneme Recognition System Using Max Flow Matching (최대 흐름 정합을 이용한 실시간 음소인식 시스템 구현)

  • Lee, Sang-Yeob;Park, Seong-Won
    • Journal of Korea Game Society
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    • v.12 no.1
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    • pp.123-132
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    • 2012
  • There are many of games using smart devices. Voice recognition is can be useful way for input. In the game, voice have to be quickly recognized, at the same time it have to be manipulated promptly as well. In this study, we developed the optimized real-time phoneme recognition using max flow matching that it can be efficiently used in the game field. Firstly, voice wavelength is transformed to FFT, secondly, transformed value is made by a graph in Z plane, thirdly, data is extracted in specific area, and then data is saved in database. After all the value is recognized using weighted bipartite max flow matching. This way would be useful method in game or robot field when researchers hope to recognize the fast voice recognition.

Voice Portal based on SMS Authentication at CTI Module Implementation by Speech Recognition (SMS 인증 기반의 보이스포탈에서의 음성인식을 위한 CTI 모듈 구현)

  • Oh, Se-Il;Kim, Bong-Hyun;Koh, Jin-Hwan;Park, Won-Tea
    • Proceedings of the Korea Information Processing Society Conference
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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.

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The Voice Dialing System Using Dynamic Hidden Markov Models and Lexical Analysis (DHMM과 어휘해석을 이용한 Voice dialing 시스템)

  • 최성호;이강성;김순협
    • 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.

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