• Title, Summary, Keyword: Voice Recognition

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A Voice Controlled Service Robot Using Support Vector Machine

  • Kim, Seong-Rock;Park, Jae-Suk;Park, Ju-Hyun;Lee, Suk-Gyu
    • 제어로봇시스템학회:학술대회논문집
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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.

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A study on Autonomous Travelling Control of Mobile Robot (이동로봇의 자율주행제어에 관한 연구)

  • Lee, Woo-Song;Shim, Hyun-Seok;Ha, Eun-Tae;Kim, Jong-Soo
    • Journal of the Korean Society of Industry Convergence
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    • v.18 no.1
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    • pp.10-17
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    • 2015
  • We describe a research about remote control of mobile robot based on voice command in this paper. Through real-time remote control and wireless network capabilities of an unmanned remote-control experiments and Home Security / exercise with an unmanned robot, remote control and voice recognition and voice transmission are possible to transmit on a PC using a microphone to control a robot to pinpoint of the source. Speech recognition can be controlled robot by using a remote control. In this research, speech recognition speed and direction of self-driving robot were controlled by a wireless remote control in order to verify the performance of mobile robot with two drives.

Characteristics of Cow´s Voices in Time and Frequency domains for Recognition

  • Ikeda, Yoshio;Ishii, Y.
    • Agricultural and Biosystems Engineering
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    • v.2 no.1
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    • pp.15-23
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    • 2001
  • On the assumption that the voices of the cows are produced by the linear prediction filter, we characterized the cows’voices. The order of this filter was determined by examining the voice characteristics both in time and frequency domains. The proposed order of the linear prediction filter is 15 for modeling voice production of the cow. The characteristics of the amplitude envelope of the voice signal was investigated by analyzing the sequence of the short time variance both in time and frequency domains, and the new parameters were defined. One of the coefficients o the linear prediction filter generating the voice signal, the fundamental frequency, the slope of the straight line regressed from the log-log spectra of the short time variance and the coefficients of the linear prediction filter generating the sequence of the short time variance of the voice signal can differentiate the two cows.

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An ASIC implementation of a Dual Channel Acoustic Beamforming for MEMS microphone in 0.18㎛ CMOS technology (0.18㎛ CMOS 공정을 이용한 MEMS 마이크로폰용 이중 채널 음성 빔포밍 ASIC 설계)

  • Jang, Young-Jong;Lee, Jea-Hack;Kim, Dong-Sun;Hwang, Tae-ho
    • 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㏈.

Discrimination of Emotional States In Voice and Facial Expression

  • Kim, Sung-Ill;Yasunari Yoshitomi;Chung, Hyun-Yeol
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.2E
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    • pp.98-104
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    • 2002
  • The present study describes a combination method to recognize the human affective states such as anger, happiness, sadness, or surprise. For this, we extracted emotional features from voice signals and facial expressions, and then trained them to recognize emotional states using hidden Markov model (HMM) and neural network (NN). For voices, we used prosodic parameters such as pitch signals, energy, and their derivatives, which were then trained by HMM for recognition. For facial expressions, on the other hands, we used feature parameters extracted from thermal and visible images, and these feature parameters were then trained by NN for recognition. The recognition rates for the combined parameters obtained from voice and facial expressions showed better performance than any of two isolated sets of parameters. The simulation results were also compared with human questionnaire results.

The Study for Advancing the Performance of Speaker Verification Algorithm Using Individual Voice Information (개별 음향 정보를 이용한 화자 확인 알고리즘 성능향상 연구)

  • Lee, Je-Young;Kang, Sun-Mee
    • Speech Sciences
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    • v.9 no.4
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    • pp.253-263
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    • 2002
  • In this paper, we propose new algorithm of speaker recognition which identifies the speaker using the information obtained by the intensive speech feature analysis such as pitch, intensity, duration, and formant, which are crucial parameters of individual voice, for candidates of high percentage of wrong recognition in the existing speaker recognition algorithm. For testing the power of discrimination of individual parameter, DTW (Dynamic Time Warping) is used. We newly set the range of threshold which affects the power of discrimination in speech verification such that the candidates in the new range of threshold are finally discriminated in the next stage of sound parameter analysis. In the speaker verification test by using voice DB which consists of secret words of 25 males and 25 females of 8 kHz 16 bit, the algorithm we propose shows about 1% of performance improvement to the existing algorithm.

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Development of a Real-time Voice Recognition Dialing System; (실시간 음성인식 다이얼링 시스템 개발)

  • 이세웅;최승호;이미숙;김흥국;오광철;김기철;이황수
    • 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.

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Implementation of speech interface for windows 95 (Windows95 환경에서의 음성 인터페이스 구현)

  • 한영원;배건성
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.34S no.5
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    • pp.86-93
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    • 1997
  • With recent development of speech recognition technology and multimedia computer systems, more potential applications of voice will become a reality. In this paper, we implement speech interface on the windows95 environment for practical use fo multimedia computers with voice. Speech interface is made up of three modules, that is, speech input and detection module, speech recognition module, and application module. The speech input and etection module handles th elow-level audio service of win32 API to input speech data on real time. The recognition module processes the incoming speech data, and then recognizes the spoken command. DTW pattern matching method is used for speech recognition. The application module executes the voice command properly on PC. Each module of the speech interface is designed and examined on windows95 environments. Implemented speech interface and experimental results are explained and discussed.

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Performance Improvement of Voice Dialing System using Post-Processing (후처리를 이용한 음성 다이얼링 시스템의 성능향상)

  • 김원구
    • 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.

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Smart Mirror of Personal Environment using Voice Recognition (음성인식을 이용한 개인환경의 스마트 미러)

  • Yeo, Un-Chan;Park, Sin-Hoo;Moon, Jin-Wan;An, Seong-Won;Han, Yeong-Oh
    • The Journal of the Korea institute of electronic communication sciences
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    • v.14 no.1
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    • pp.199-204
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    • 2019
  • This paper introduces smart mirror that provides the contents needed for an individual's daily life. When a command that is designated as voice recognition is entered, Smart Mirror is produced that outputs desired contents from a display. The contents of the current smart mirror include time, weather, subway information, schedule and photography. Smart mirror sold for commercial private households is difficult to distribute due to high prices, but the smart mirror production presented in this paper can lower the manufacturing cost and can be more easily used by voice recognition.