• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.343-346
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• 2005

In this study, a basic concept of wavelet encoding and its advantages over Fourier based phase encoding application. Wavelet encoding has been proposed as an alternative way to Fourier based phase encoding in magnetic resonance imaging. In wavelet encoding, the RF pulse is designed to generate wavelet-shaped excitation profile of spins. From the resulting echo signals, the wavelet transform coefficients of spin distribution are acquired and an original spin density is reconstructed from wavelet expansion. Wavelet encoding has several advantages over phase encoding. By minimizing redundancy of the data acquisition in a dynamic series of images, we can avoid some encoding steps without serious loss of quality in reconstructed image. This strategy may be regarded as data compression during imaging. Although there are some limitations in wavelet encoding, it is a promising scheme in a dynamic imaging.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.533-536
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• 2006

본 논문에서는 음성 신호로부터 감정의 특징을 나타내는 요소를 찾아내는 것을 목표로 하고 있다. 일반적으로 감정을 인식할 수 있는 요소는 단어, 톤, 음성신호의 피치, 포만트, 그리고 발음 속도 및 음질 등이 있다. 음성을 기반으로 감정을 익히는 방법 중에서 현재 가장 많이 접근하고 있는 방법은 피치에 의한 방법이 있다. 사람의 경우는 주파수 같은 분석 요소보다는 톤과 단어, 빠르기, 음질로 감정을 받아들이게 되는 것이 자연스러운 방법이므로 이러한 요소들이 감정을 분류하는데 중요한 요소로 쓰일 수 있다. 따라서, 본 논문에서는 감정에 따른 음성의 특징을 추출하기 위해 사람의 감정 중에서 비교적 자주 쓰이는 평상, 기쁨, 화남, 슬픔에 관련된 4가지 감정을 비교 분석하였으며, 인간의 감정에 대한 음성의 특성을 분석한 결과, 강도와 스펙트럼에서 각각의 일관된 결과를 추출할 수 있었고, 이러한 결과에 대한 실험 과정과 최종 결과 및 근거를 제시하였다. 끝으로 실험에 의해 제안한 방법의 유용성을 입증하고자 한다.

• Proceedings of the KSPS conference
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• 2002.11a
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• pp.119-122
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• 2002

This paper focuses on building up a database of commercial stocks using XML syntax and looks into a way of building up a system with the combination of XML and XSLT that provides connectivity to client-server databases through vocal means. The use of XSLT has several advantages. Most importantly, it can transform a type of data into different formats. A vocal interface minimizes some space and time limits imposed on users outside premises when they need an instant connection to their database. In this fashion, the users can check information on stock lists without being pressurized by certain limits. PC, PDAs and cellular phones are some examples of mobile connection. The use of VoiceXML creates vocal applications. In VoiceXML servies, users can gain immediate access to data upon the input of their voices and the DTMF signals of the telephone.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.8 no.3
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• pp.113-119
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• 1983

In this paper, a method of the step size adaption is studied on the delta modulation coding of speech signals. Exponential adaption processes are reserched by a new circuit model. It is presented a shorten error recovery in decoder step size. Practical considerations favor one algorithm, and its digital implementation has been adapted for the illustration of above method, using the rate multipliers and the validity is verified by laboratory experiment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.9 no.3
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• pp.120-126
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• 1984

In this paper, there is presented the novel encoder circuit design method in the realization of exponential adaption process on the delta modulation coding of speech signals. The digital implementation has been adapted for the illustration of above, especially using a rate multiplier end a double integration circuit. The use of a double integration of the local decoder included in the ADM encoder in prove the undesirable characteristics which the low switching speed of the ratemultiplier couses the SQNR to decreuse, and the SQNR of the decoding signal by above realization is relatively uniformed in wide range of signal levels. The validity of the above design is verified by laboratory experiments.

• International Journal of Control, Automation, and Systems
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• v.5 no.1
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• pp.61-69
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• 2007

In this paper, we developed not only a reliable sound localization system including a VAD(Voice Activity Detection) component using three microphones but also a face tracking system using a vision camera. Moreover, we proposed a way to integrate three systems in the human-robot interaction to compensate errors in the localization of a speaker and to reject unnecessary speech or noise signals entering from undesired directions effectively. For the purpose of verifying our system's performances, we installed the proposed audio-visual system in a prototype robot, called IROBAA(Intelligent ROBot for Active Audition), and demonstrated how to integrate the audio-visual system.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1E
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• pp.23-29
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• 1992

음성파형은 인근 표본값들 사이에 높은 상관관계를 나타낸다. 음성신호의 상관관계를 증가시키 기 위한 한 방법으로는 부호화하기 전에 입력신호를 단순히 적분시키는 방법이다. 이 적분된 rqkt들은 수신기에서 일반 미분기에 의해 제거될 수 있다. 이렇게 하면 음성신호의 저역주파수가 강조되고 인근 표본값의 자기 상관관계가 증가된다. 이런 과정을 시그마-델타 기법이라 한다. 이 논문에서는 그러한 시 그마-델타의 특성을 사용하는 예측기를 새로이 제안한다. 즉, 부호화하기 전에 입력신호를 적분하고 인 근한 과거 및 미래의 두 표본을 사용하여 적분된 현재표본을 예측한다. 제안된 예측기는 CCITT-권고 형 ADPCM의 평균 예측이득보다 8.65db 높게 얻어졌다.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1E
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• pp.42-49
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• 1992

시간 영역에서 음성의 피치 정보를 추출하는 새로운 알고리즘을 제안한다. 이 알고리즘은, 위상 이 일치하는 고조파 성분의 합은 위상이 일치하지 않는 고조파 성분의 합의 경우보다 주기 정보를 분명 히 나타낸다는 사실을 이용한 것이다. 즉, 음성 신호의 위상 성분을 0으로 되도록 하여 실질적으로 기본 파와 모든 고조파 성분의 위상을 일치시킨다. 이 알고리즘은 잡음이 없는 음성의 경우 0.18%의 조오류 를 보이며, 0dB 눅의 경우에도 3.63%의 조오류를 보임으로써 잡음에 강건한 성질이 있음을 알 수 있다. 또한 시간 영역에서의 결정 논리를 사용하므로 피치 해상도가 우수하다. 전반적인 실험결과는 제안된 알고리즘이 피치 검출에 상당히 효율적임을 나타낸다.

• Proceedings of the KIEE Conference
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• 1989.11a
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• pp.481-483
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• 1989

This paper is concentrated on the recognition of Korean Digits. The speech signals of each of digits are fed into computer through the 18 bandpass filters, AD converter. Spectrum input data are analyzed and used. BASIC program language is used for recognition performance and the result of recognition is outputed to computer screen and printer. In this paper, the strength and weakness of filter-bank analysis method is described and the technique of real-time recognition is argued. In this experiment, Ratio of recognition for speaker dependent recognition was about 97% and recognition time was also satisfied. Therefore, A way of speaker independent recognition will be presented and using for special communication in the future.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.10
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• pp.141-150
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• 1997

In this paepr, we propose an intelligent double talk detector that can enhance the performance of the acoustic echo cancellation system. The conventional double talk detection methods often misunderstand the echo path changes as double talk. Although there exist several detection methods that can distinguish the echo path changes from the double-talks, they show poor tracking performance because of the excessive decision delay for the discrimination and can only be used after the adaptive digital filter converges. A new and more effective ditetion algorithm has been proposed, where the detection mechanism is performed by observing the change rate of the reflection coefficients of the two lattice predictors that re placed on the near-end and far-end terminals. The excellence of the proposed method is verified by extensive computer simulations using real speech signals.