• Journal of Biomedical Engineering Research
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• v.15 no.4
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• pp.383-388
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• 1994

Chaotic analysis of EEG signal responding to auditory stimulus with various carrier frequency and constant triggering frequency is given in this paper. The EEG signal is obtained from the digital 12channel EEG system made in our laboratory. The carrier frequency is varied from 1 kHz to 3 kHz by 0.5 kHz step. Chaos analysis such as pseudo phase space portrait, Lyapunov exponent, and so on is done on the auditory stimulated evoked potential. This result is found to be quite consistent with the well known results from the psychological perception theory.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.57-62
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• 2010

Recently in the mobile market, the communication technology which bases on the sense of sight, sound, and touch has been developed. However, human beings uses all five - vision, auditory, palatory, olfactory, and tactile - senses to communicate. Therefore, the current paper presents a technology which enables individuals to be aware of other people's emotions through a machinery device. This is achieved by the machine perceiving the tone of the voice, body temperature, pulse, and other biometric signals to recognize the emotion the dispatching individual is experiencing. Once the emotion is recognized, a scent is emitted to the receiving individual. A system which coordinates the emission of scent according to emotional changes is proposed.

• Investigative Magnetic Resonance Imaging
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• v.15 no.2
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• pp.130-138
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• 2011

Purpose : To compare fMRIs of visual and auditory word generation tasks, and to evaluate the difference of its activated areas and lateralization according to the mode of stimuli. Materials and Methods : Eight male normal volunteers were included and all were right handed. Functional maps were obtained during auditory and visual word generation tasks in all. Normalized group analysis were performed in each task and the threshold for significance was set at p<0.05. Activated areas in each task were compared visually and statistically. Results : In both tasks, left dominant activations were demonstrated and were more lateralized in visual task. Both frontal lobes (Broca's area, premotor area, and SMA) and left posterior middle temporal gyrus were activated in both tasks. Extensive bilateral temporal activations were noted in auditory task. Both occipital and parietal activations were demonstrated in visual task. Conclusion : Modality independent areas could be interpreted as a core area of language function. Modality specific areas may be associated with processing of stimuli. Visual task induced more lateralized activation and could be a more useful in language study than auditory task.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.13-20
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• 2001

In this paper, we propose a new digital audio watermarking scheme In the cepstrum domain. We insert a digital watermark signal Into the cepstral components of the audio signal using a technique analogous to spread spectrum Communications, hiding a narrow band signal in a wade band channel. In our proposed method, we use pseudo-random sequences to watermark the audio signal. The watermark Is then weighted in the cepstrum domain according to the distribution of cepstral coefficients and the frequency masking characteristics of the human auditory system. The proposed watermark embedding scheme minimizes audibility of the watermark signal. and the embedded watermark is robust to mu1tip1e watermarks, MPEG audio ceding and additive noose.

• The Journal of the Acoustical Society of Korea
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• v.6 no.1
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• pp.58-65
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• 1987

The audible tones of switching system employed in the country at present do not only disagree with CCITT Recommendation Q35, but also are different in the specifications and the sense of hearing. Such a discord with the specifications and the sense of hearing causes a subscriber and an operator to confuse. In this paper characteristics of the sense of hearing of them and the accuracy of tone specifications are studied and novel unified audible specifications are proposed. And it is showed that the digital tone generation of digital switching systems can be implemented readily, using a ROM.

• Journal of Korea Multimedia Society
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• v.8 no.11
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• pp.1432-1438
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• 2005

A psychoacoustic model based noise shaping method is proposed. The method shapes the noise in the frequency domain such that its presence with a host signal will not be perceptually noticeable. The derivation of imperceptible noise levels from the masking thresholds of the signal involves deconvolution associated with the spreading function in the psychoacoustic model. It has been known as an ill-conditioned Problem. In this paper, a constrained optimization is applied such that the noise excitation level conforms to the masking thresholds of the signal. Thus, the noises embedded in the signal will not be perceived by human ear, and its performance is demonstrated experimentally.

• 한국HCI학회:학술대회논문집
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• 2007.02b
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• pp.478-483
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• 2007

본 연구는 가전제품 사용자 인터페이스에서 음성 신호와 청각 신호의 최적 할당 방법을 기술하였다. 가정에서 수시로 접하는 가전제품에서 음성 유저 인터페이스(Voice User Interface, 이하 VUI) 는 음성을 매개로 일어나는 인간과 기계 간 인터페이스를 뜻한다. 음성 유저 인터페이스의 단독적 적용보다는 소리 신호와 함께 사용하여 사용자들의 인터페이스를 향상시킬 수 있다. 본 연구에서는 주부 사용자들을 대상으로 F.G.I, 실험, Depth Interview 를 수행하여 가전제품의 음성 생성 및 표현 인터페이스에서 음성과 소리 신호의 배치에 대한 사용자들의 니즈 조사 및 실험 결과를 기반으로 최적의 할당 방법을 제시하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.217-220
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• 2002

본 연구는 폐쇄된 임의의 공간상에서 2개의 마이크로폰 어레이를 이용하여 마이크로폰에 수신된 신호들의 도착 시간차를 추정하는 새로운 알고리즘을 제안한다. 제안된 알고리즘은 입력 음성신호를 Discrete wavelet transform을 이용하여 인간의 청각 특성과 가장 유사한 주파수 해상도를 갖도록 대역 분할한 후 각 주파수 대역에서 신호 대 잡음비를 구하여 신호 대 잡음비가 가장 높은 대역만 선택적으로 취하고 해당 대역에서만 최종적인 시간 지연 값을 추정하게 된다. 최종 시간 지연 측정에 사용된 알고리즘은 기존의 CPSP에 해당 대역의 주파수 SNR을 가중치로 주어 구하게된다. 이러한 대역 분할 가중방식은 다양한 형태의 동적인 잡음 환경 하에서 안정적인 성능을 가질 수 있다. 제안된 알고리즘은 저주파와 고주파 각각의 모의 잡음환경 하에서 컴퓨터 실험을 통해 성능을 입증하도록 한다.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2613-2615
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• 2001

청각시스템의 저전력 및 가격의 저렴화를 위해 달팽이관의 BM(Basilar Membrain)모델을 아날로그 VLSI 마이크로 파워 공정으로 구현하고 있다. 본 논문에서는 소리의 주파수 정보 추출기능을 하는 직렬 연결된 트리구조(TSBF : Tree-structured Cascaded Bandpass Filter)의 16채널의 아날로그 중간대역통과 필터회로를 CMOS VLSI 공정을 이용하여 설계하였다. 특히 큰 입력 신호에 대해서도 파형왜곡 없이 선형적인 특성을 가지는 트랜스 컨턱터를 이용하여 필터를 구현하였다. 필터는 저대역통과필터와 출력이득의 감쇄를 줄이기 위해서 중간대역통과필터를 이용하여 전체 시스템을 설계했다. 본 논문에서 기존의 150mVp-p 입력신호 범위의 트랜스 컨턱터를 Substrate 입력을 가지는 트랜스 컨턱터를 이용하여 입력신호 범위를 1Vp-p 까지 늘였다.

• Proceedings of the KOSOMBE Conference
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• v.1991 no.05
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• pp.39-42
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• 1991

Two types of speech signal processores (SSP) for cochlea prosthesis are designed. One is designed using cochlear model and the other is designed using Information (formant, pitch, intensity) extraction method. For these, cochlear model and acoustic information extraction method are proposed. The result shows SSP of cochlear model type contain more acoustic cues than that of information extraction type.