• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2008년도 춘계종합학술대회 A
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• pp.421-424
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• 2008

본 논문에서는 다양한 배경잡음에 의해 열화된 음성을 강조하기 위하여 청각모델에 기초로 한 잡음환경에 적응된 잡음억압 시스템을 제안한다. 제안한 시스템은 먼저 유성음과 무성음의 구간을 검출한 후, 각 입력 프레임에서 적응적인 청각기강의 처리를 한다. 마지막으로 진폭성분과 위상성분이 포함된 신경회로망을 사용하여 잡음신호를 제거한 후에 음성을 강조하는 처리를 한다. 본 시스템은 신호대잡음비의 평가방법을 통하여 다양한 잡음에 의해서 열화된 음성신호에 대해서 유효하다는 것을 실험으로 확인한다.

• 전자통신동향분석
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• 제8권2호
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• pp.69-91
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• 1993

In the conventional speech processing, the technique of FFT(Fast Fourier Transform) is usually applied to the finite number of samples within the window of specified length using the fixed sampling rate. In this case, the temporal resolution is dependent upon the length of window while the spectral resolution is dependent upon the number of samples within the window. Thus, once the temporal resolution is determined the spectral resolution is also determined or vice versa. To resolve this type of dilemma, a new type of bank-filter similar to the characteristics of cochlear model needs to be considered. Furthermore, wide dynamic range of cochlea certainly helps the stable extraction of speech features. In the paper, the human auditory system will be briefly introduced and previous works on auditory preprocessors based on cochlear model will be reviewed. As a conclusion, the design considerations of auditory preprocessors based on cochlear model will be addressed.

• 한국음향학회지
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• 제20권8호
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• pp.81-87
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• 2001

The human auditory system is appropriate for the "constant Q"system. The STFT (Short Time Fourier Transform) is not suitable for the auditory perception model since it has constant bandwidth. In this paper, the CWT (continuous wavelet transform) is employed for the auditory filter model. In the CWT, the frequency resolution can be adjusted for auditory sensation models. The proposed CWT is applied to the modeling of the JNVF. In addition, other signal processing methods such as STFT, VER-FFT and VFR-STFT are discussed. Among these methods, the model of JNVF (Just Noticeable Variation in Frequency) by using the CWT fits in with the JNVF of auditory model although it requires quite a long time.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -3
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• pp.1717-1720
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• 2002

Objective quality measurement schemes that in- corporate properties of the human auditory system. The basilar membrane(BM) acts as a spectrum analyzer, spatially decomposing the signal into frequency components. Each filterbank is an implementation of the ERB, gam-machirp function. This filterbank is level-dependent asymmetric compensation filters. And for the validation of the auditory model, we calculate the CPD. Quality measurement is obtained from the result.

• 대한인간공학회지
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• 제25권3호
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• pp.67-75
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• 2006

HVIMs consists of simulated driver models implemented with series of mathematical functions and computerized vehicle dynamic models. To effectively model the perception process, as a part of driver models, psychophysical nonlinearity should be considered not only for the single-modal stimulus but for the stimulus of multiple modalities and interactions among them. A series of human factors experiments were conducted using the primary sensory of visual and auditory modalities to find out the effects of auditory cues in visual velocity estimation tasks. The variations of auditory cues were found to enhance/reduce the perceived intensity of velocity as the level changed. These results indicate that the conventional psychophysical power functions could not applied for the perception process of the HVIMs with multi-modal stimuli. 'Ruled surfaces' in a 3-D coordinate system(with the intensities of both kinds of stimuli and the ratio of enhancement, respectively for each coordinate) were suggested to model the realistic perception process of multi-modal HVIMs.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1998년도 하계학술대회 논문집 G
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• pp.2259-2261
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• 1998

Auditory Models that are capable of achieving human performance would provide a basis for realizing effective speech processing systems. Perceptual invariance to adverse signal conditions (noise, microphone and channel distortions, room reverberations) may provide a basis for robust speech recognition and speech coder with high efficiency. Auditory model that simulates the part of auditory periphery up through the auditory nerve level and new distance measure that is defined as angle between vectors are described.

• 한국정보통신학회논문지
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• 제13권4호
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• pp.652-657
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• 2009

본 논문에서는 다양한 배경 잡음에 의해 열화된 음성을 강조하기 위하여 청각모델에 기초로 한 잡음환경에 적응된 잡음억제 시스템을 제안한다. 제안한 시스템은 먼저 유성음, 무성음 및 묵음의 구간을 검출한 후, 각 입력 프레임에서 적응적인 청각기강의 처리를 한다. 마지막으로 진폭성분과 위상성분이 포함된 신경회로망을 사용하여 잡음신호를 제거 한 후에 음성을 강조하는 처리를 한다. 본 시스템은 신호대잡음비의 평가방법을 통하여 다양한 잡음에 의해서 열화 된 음성신호에 대해서 유효하다는 것을 실험으로 확인한다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2144-2147
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• 2003

Objective quality measurement schemes that incorporate properties of the human auditory system. The basilar membrane (BM) acts as a spectrum analyzer, spatially decomposing the signal into frequency components. Filterbanks were used to complementing the linearity of BM. Each filterbank is an implementation or the Equivalent rectangular Bandwidth (ERB), gammachirp function. This filterbank is level-dependent asymmetric compensation filters. And for the validation of the auditory model, we calculate the calculated perceived difference(CPD).

• 한국공작기계학회:학술대회논문집
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• 한국공작기계학회 2000년도 추계학술대회논문집 - 한국공작기계학회
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• pp.495-499
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• 2000

Auditory system is separated to Outer Ear, Middle Ear and Inner Ear, Middle Ear plays an important role as the sound transfer on amplitude. With analysing of Middle Ear, we can understand disease and compare unformal auditory systems. However, the investigation of mechanical modeling and analysis have been reported in a few paper. In this paper, a three dimensional Eardrum model of human ear was developed and analysed applying the general purpose Finite-Element program (Nastran). Vibration patterns of the eardrum obtained from FEM analysis are in agreements with the experimental results using stroboscope.

• The Journal of the Acoustical Society of Korea
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• 제16권1E호
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• pp.15-23
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• 1997

Zero-Crossings with Peak Amplitudes(ZCPA) model motivated by human auditory periphery was proposed to extract reliable features speech signals even in noisy environments for robust speech recognition. In this paper, some practical considerations for digital hardware implementations of the ZCPA model are addressed and evaluated for recognition of speech corrupted by several real world noises as well as white Gaussian noise. Infinite impulse response(IIR) filters which constitute the cochliar filterbank of the ZCPA are replaced by hamming bandpass filters of which frequency responses are less similar to biological neural tuning curves. Experimental results demonstrate that the detailed frequency response of the cochlear filters are not critical to performance. Also, the sensitivity of the model output to the variations in microphone gain is investigated, and results in good reliability of the ZCPA model.