• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.5
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• pp.907-912
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• 2006

Pre-emphasizing the speech compensates for falloff at high frequencies. The most common form of pre-emphasis is y(n)=s(n)-A${\cdot}$s(n-1), where A typically lies between 0.9 and 1.0 in voiced signal. And, this value reflects the degree of pre-emphasis and equals R(1)/R(0) in conventional method. This paper proposes a new flattening method to compensate the weaked high frequency components that occur by vocal cord characteristic. We used QMF(Quardrature Mirror Filter) to minimize the output signal distortion. After using the QMF to compensate high frequency components, flattening process is followed by R(1)/R(0) at each frame. Experimental results show that the proposed method flattened the weaked high frequency components effectively than auto correlation method. Therefore, the flattening algorithm will apply in speech signal processing like speech recognition, speech analysis and synthesis.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3558-3562
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• 2010

The exact fundamental frequency (pitch) extraction is important in speech signal processing. However the exact pitch extraction from speech signal is very difficult due to the effect of formant and transitional amplitude. So in this paper, the pitch is detected after flattening the spectrum in frequency region by proposed algorithm for minimized harmonics variance value. Experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum. Also, the results show the proposed method is better than conventional method.

• Journal of Navigation and Port Research
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• v.31 no.10
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• pp.833-837
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• 2007

In this study, we have fabricated a gain flattened erbium-doped optical fiber amplifier. Gain flattening filters were realized by the strain-induced long period fiber gratings, which are made of periodically arrayed metal wires. Using the filter of $550{\mu}m$ period, spontaneous emission amplified at C-band wavelength by a 980nm pumping laser was flattened within 1dB of gain ripple. The performance of the simultaneous multi channel amplification was measured using a fabry-perot laser diode. Amplification ratio was above 20dB. This amplifier can be applied to the long distance transmission system based on a wavelength division multiplexing for boosting an attenuated signal.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.805-808
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• 2001

음성인식, 합성 및 분석과 같은 음성신호처리 분야에 있어서 기본주파수 즉, 피치를 정확히 검출하는 것은 중요하나 만일 음성신호의 기존주파수를 정확히 검출할 수 있다면 음성인식에 있어서 화자에 따른 영향을 줄일 수 있기 때문에 인식의 정확도를 높일 수 있고, 음성합성 시에 자연성과 개성을 쉽게 변경하거나 유지할 수 있다. 또한 분석시 피치에 동기 시켜 분석하면 성문의 영향이 제거된 정확한 성도 파라미터를 얻을 수 있다. 따라서 제안한 논문에서는 시간영역처리에서 시간영역파형에서 먼저 성도성분이 제거된 성문특성 즉 피치주기 성분을 강조하기 위해 음성신호의 기울기를 이용한 Positive 센터 클리핑를 수행하고 주파수 영역에서는 원신호의 스펙트럼과 센터클립된 신호의 스펙트럼과의 Peak-Fiting을 수행하고 선형 인터폴래이션(스무딩)을 통해 평탄화된 스펙트럼을 얻었다. 결과적으로 유성음 구간과 음성이 변하는 전이구간에서 G-peak가 강조된 더욱 정확한 Pitch를 검출할 수 있었다.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.273-276
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• 2010

음성신호를 관찰하였을 때 성문특성으로 인해서 고주파쪽 특성이 약화되는 경향이 있다. 약화된 고주파 특성을 보상하기 위하여 프리 엠퍼시스 필터를 통해 보상하고 있다. 프리 엠퍼시스 필터를 간단한 수식으로 표현하면 y(n)=s(n)-As(n-1)와 같이 차분 방정식으로 나타낼 수 있다. 여기서 A값은 보통 0.9에서 1사이의 값을 사용한다. 본 논문에서는 QMF 필터를 이용하여 입력신호를 고주파와 저주파의 2개의 대역으로 분할하여 각 밴드에 프리 엠퍼시스 필터를 적용하여 약화되어진 특성을 정확히 보상하는 방법을 제안한다.

• Proceedings of the Optical Society of Korea Conference
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• 2000.08a
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• pp.140-141
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• 2000

금속으로 코팅한 광섬유는 폴리머로 코팅한 광섬유에 비해 여러 가지 장점을 가지고 있다. 금속 코팅 광섬유는 열적, 화학적 안정성이 높고 고온에서 사용할 수 있으며 수분 침투에 강하고 높은 굽힘 강도를 가진다. 또 납땜을 할 수가 있어서 광섬유 소자 패키징에 있어서 매우 유리하다$^{(1)}$ . 장주기 광섬유 격자는 WDM 용 필터, 온도 및 변형도 센서, 광섬유 증폭기의 이득 평탄화 필터에 많은 유용성을 가지고 있다. 광섬유 격자를 소자로 사용하기 위해서는 재 코팅이 필수적인데 금속은 좋은 대안이 될 것이다. 금속을 재 코팅 재료로 사용하기 위해서는 금속 코팅이 격자에 미치는 영향에 대한 연구가 요구된다. 광섬유 격자 쌍은 단일 장주기 격자에 비해 분해능이 좋아 센서 및 필터 등에 유용하여 본 연구에서는 광섬유 격자쌍 위에 은을 코팅하여 투과 스펙트럼에 미치는 영향을 알아보았다. (중략)

• Computational Structural Engineering
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• v.9 no.4
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• pp.229-237
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• 1996

In some current procedures, ground motions from different sources have been scaled by their peak ground accelerations and combined to obtain smoothed response spectra for specific regions. As consideration of the inelastic deformation capacity of structure, inelastic deformations are permitted under seismic ground excitation in all codes. In the ATC(Applied Technology Council) and UBC(Uniform Building Code), the inelastic design spectrum is obtained by reducing the elastic design spectrum by a factor that is independent of structural period. In this study, the average of nonlinear response spectra calculated from a sample of 20 records for each event are constructed to obtain the smoothed response spectra. These response spectra are used to examine the effects of structural strength factors such as the yield strength ratio and damping value. Through the regression analysis of nonlinear response of system for a given damping value and yield strength ratio, the required yield strength for seismic design can be estimated for a certain earthquake event. And a response modification coefficient depending on the natural period for current seismic design specifications are proposed.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.159-162
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• 1998

Speech synthesis coding is classified into three categories: waveform coding, source coding and hybrid coding. To obtain the synthetic speech with high quality, the synthesis by waveform coding is desired. However, it is difficult to apply waveform coding to synthesis by syllable or phoneme unit, because it does not divide the speech into excitation and formant component. Thus it is required to alter the excitation in waveform coding for applying waveform coding to synthesis by rule. In this paper we propose a new pitch alteration method that minimizes the spectrum distortion by using the behavior of cepstrum. This method splits the spectrum of speech signal into excitation spectrum and formant spectrum and transforms the excitation spectrum into cepstrum domain. The pitch of excitation cepstrum is altered by zero insertion or zero deletion and the pitch altered spectrum is reconstructed in spectrum domain. As a result of performance test, the average spectrum distortion was below 2.29%.

• MALSORI
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• no.56
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• pp.225-235
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• 2005

Speech unit concatenation with a large database is presently the most popular method for speech synthesis. In this approach, the mismatches at the unit boundaries are unavoidable and become one of the reasons for quality degradation. This paper proposes an algorithm to reduce undesired discontinuities between the subsequent units. Optimal matching points are calculated in two steps. Firstly, the fullback-Leibler distance measurement is utilized for the spectral matching, then the unit sliding and the overlap windowing are used for the waveform matching. The proposed algorithm is implemented for the corpus-based unit concatenating Korean text-to-speech system that has an automatically labeled database. Experimental results show that our algorithm is fairly better than the raw concatenation or the overlap smoothing method.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.381-384
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• 2002

In speech signal processing, it Is very important to detect the pitch exactly in speech recognition, synthesis and analysis. but, it is very difficult to pitch detection from speech signal because of formant and transition amplitude affect. therefore, in this paper, we proposed a pitch detection using the spectrum flattening techniques. Spectrum flattening is to eliminate the formant and transition amplitude affect. In time domain, positive center clipping is process in order to emphasize pitch period with a glottal component of removed vocal tract characteristic. And rough formant envelope is computed through peak-fitting spectrum of original speech signal in frequency domain. As a results, well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. After all, we obtain residual signal which is removed vocal tract element The performance was compared with LPC and Cepstrum, ACF 0wing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.