• 한국컴퓨터정보학회논문지
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• 제22권7호
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• pp.83-91
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• 2017

In this paper, we propose a thorough solution for adjusting virtual sound field with different kinds of devices and software in preliminary procedure and late stage of music processing. The basic process of music production includes composing, arranging and recording at pre-production stage as well as sound mixing and mastering at post-production stage. At the initial stage of music creation, it should be checked whether the design of virtual sound field, the choice of the tone and the instrument used in the arrangement match the virtual sound field required for the final work. In later recording, mixing and mastering, elaborate adjustments should be done to the virtual sound field. This study also analyzed how to apply the parameter of the effectors to the design and adjustment of the virtual sound field, making it the source of our creation.

• 한국음향학회지
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• 제18권3호
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• pp.79-87
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• 1999

본 논문에서는 입체 음상 제어 알고리즘을 실시간 처리에 적합하도록 재구성하였으며, 범용 DSP 칩인 TI사의 TMS320C40 기반의 디지털 신호처리 보드를 이용하여 실시간 구현하였다. 구현된 실시간 입체 음상 제어 시스템의 성능은 청취 실험을 통하여 평가되었다. 청취실험 결과, 정위된 허음상은 헤드폰과 스피커를 통하여 지각이 가능하였으며. 헤드폰을 통해 재생된 음상이 스피커를 통해 재생된 음상보다 지각이 잘되는 것으로 나타났다. 고도각 지각 실험에서는 전후방향보다 좌우방향의 지각이 우수하게 나타났으며, 정지음보다 이동음의 지각이 우수하게 나타났다.

• Journal of Advanced Marine Engineering and Technology
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• 제38권9호
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• pp.1125-1130
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• 2014

본 논문에서는 디지털 신호처리 기술에 기반하여 주변잡음에 강건한 선박용 음향수신장치를 제안한다. 원하지 않는 주변 잡음을 억제하기 위하여 70Hz-820Hz의 통과대역을 갖는 디지털 대역통과 필터를 적용한다. 4개의 마이크로폰으로부터 입력된 음향 신호들이 디지털 대역통과 필터를 거친 후 8 방향의 방향 탐지가 가능한 방향 탐지 알고리즘을 제안한다. 또한, DSP 칩과 오디오 코덱을 이용한 프로토타입 시스템을 구현하여 본 알고리즘의 동작을 테스트한다.

• 한국마린엔지니어링학회:학술대회논문집
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• 한국마린엔지니어링학회 2006년도 전기학술대회논문집
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• pp.169-170
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• 2006

The main purpose of this study is to develop a program for sound field visualization system which gets noise signals in microphones array for incoming noise signals and it uses to operate noise signals and to store data in multi-channel FFT and is consisted to visualize noise signals with a image which is got by camera in center of array by using beamforming algorithm of the array signal processing.

• 전자공학회논문지B
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• 제28B권12호
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• pp.9-18
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• 1991

In this Paper, a rule-based method for the phenomenon of Korean vocal sound change is proposed. This method could be used to solve a problem between symbolic(Hangul)and phonetic language(Korean) for the study of Korean speech processing. A rule on the phenomenon of vocal sound rearranged for the rule base with a end-consonents on the authority of standard pronunciation rule. The proposed rule base system is simplified by the implementation for the vocal sound change. Also, it is useful to create the data base with phonetic value for the Korean voice processing by syllable unit.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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• pp.441-444
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• 2004

In the sound insulation measurements, the influence of background (extraneous) noise is often serious problem and how to reduce its effect and to improve the signal-to-noise(S/N) ratio is an important theme. As the background noise, such extraneous noises as road traffic noise and machine noise often disturb the measurement. In laboratory measurements on specimens with high sound insulation performances, even the internal noise of the measurement system can become a problem. To improve the signal-to-noise ratio and to improve the measurement accuracy, various kinds of digital signal processing techniques can be applied. In this paper, four kinds of digital signal processing techniques are applied and their effectiveness is examined through field measurements.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 하계종합학술대회 논문집
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• pp.1107-1110
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• 1999

In speech signal processing, the accurate decision of the voiced/unvoiced sound is important for robust word recognition and analysis and a high coding efficiency. In this paper, we propose the mehod of the voiced/unvoiced decision using the LSP parameter which represents the spectrum characteristics of the speech signal. The voiced sound has many more LSP parameters in low frequency region. To the contrary, the unvoiced sound has many more LSP parameters in high frequency region. That is, the LSP parameter distribution of the voiced sound is different to that of the unvoiced sound. Also, the voiced sound has the minimun value of sequantial intervals of the LSP parameters in low frequency region. The unvoiced sound has it in high frequency region. we decide the voiced/unvoiced sound by using this charateristics. We used the proposed method to some continuous speech and then achieved good performance.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2244-2247
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• 2003

In this paper, we present a method which can minimize distortion from desired signal in thoracic sound signal processing. We firstly chose the proper wavelet mother function to reduce noise components. Secondly, we chose a clean thoracic sound, then added Gaussian noise and 3 step(10, 15, 20db) uniform noise to it. Finally, the various wavelet functions are applied for noise cancellation. To evaluate the efficiency of this study, we computed SNR and RSE value. Then we found the optimal mother wavelet function for thoracic sound.

• 한국정보과학회:학술대회논문집
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• 한국정보과학회 2003년도 가을 학술발표논문집 Vol.30 No.2 (1)
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• pp.4-6
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• 2003

A holistic representation, such as sparse ceding or independent component analysis (ICA), was successfully applied to explain early auditory processing and sound classification. In contrast, Part-based representation is an alternative way of understanding object recognition in brain. In this paper. we employ the non-negative matrix factorization (NMF)［1］which learns parts-based representation for sound classification. Feature extraction methods from spectrogram using NMF are explained. Experimental results show that NMF-based features improve the performance of sound classification over ICA-based features.

• 한국음향학회지
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• 제26권4호
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• pp.144-152
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• 2007

기존의 심음분류를 위한 연구들은 인공신경망을 이용하여 주로 이루어졌다. 그러나 심음신호의 통계적 특성을 분석 한 결과 HMM의 의한 신호모델링이 적합한 것으로 나타났다. 본 연구에서는 다양한 질병을 나타내는 심음신호를 HMM을 이용하여 모델링 하고 인식성능이 심음신호의 클러스터링에 따라서 많이 좌우되는 것을 알 수 있었다. 또한 실제 환경에서의 심음신호는 그 시작과 끝나는 시점이 정해지지 않은 연속신호이다. 따라서 HMM을 이용한 심음분류를 위해서는 연속적인 심음신호로부터 한 사이클의 분할된 심음을 추출할 필요성이 있다. 일반적으로 수동분할은 분할오류를 발생시키며 실시간 심음인식에 적합하지 않으므로 분할과정이 필요치 않는 ergodic형 HMM을 변형하여 사용할 것을 제안하였다. 그리고 제안된 HMM은 연속심음을 이용한 분류실험에서 매우 높은 성능을 보임을 알 수 있었다.