• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.824-826
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• 2015

최근 음성인식 시스템의 성능 향상은 많이 개선되었지만 아직도 잡음과 같은 문제로 인하여 문제점이 나타나고 있다. 음성인식 시스템에 있어서의 잡음 문제를 해결함으로써 인식 성능을 향상할 목적으로 본 논문에서는 단시간 스펙트럼에 기초한 주파수특성을 고려한 위너필터를 사용한 잡음 차감 알고리즘을 제안한다. 제안한 알고리즘은 먼저 각 프레임에서 문턱값을 검출한 후에 비묵음 구간과 묵음 구간을 식별한다. 각 프레임에 대해서 비묵음 구간에서는 위너필터법에 의한 잡음 차감법을 실시하며, 묵음 구간에 대해서는 일반적인 잡음 차감법을 적용한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.6
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• pp.699-712
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• 1989

This paper describes a method of Korean speech synhes is using format VOCODER. The parameters of speech synthes is are a follows, 1) format F1, F2, and F3 by spectrum moment method and F4, F5 using the length of vocal tract. 2) pitch frequencies obtained by optimu, Comb method using AMDF. 3) short time average energy and short time mean amplitude. 4) The decision method of bandwidth reportd by Fant. 5) voicde/unvoiced discrimination using zerocrossing. 6) excitation wave reported by Rosenberg. 7) gaussian white noise. Synthesis results are in fairly good agreement with original speech.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.105-111
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• 1997

Before application of the angular spectrum method to calculate acoustic fields in stratified water, its efficiencies and errors were analyzed by using a virtual boundary in homogeneous water. As the results, it was confirmed that the angular spectrum method was able to calculate an acoustic field rapidly though some errors due to the limitation of reference field size and number of data in FFT ware included. A modified method combined the angular spectrum with Lommel's approximation, which was newly proposed in this paper, was useful to reduce the errors.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.39 no.12
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• pp.1281-1286
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• 2015

Fatigue cracks can be generated in aircraft as a result of the cumulative time spent during flight operations, which can extend for long periods of time and cover a variety of missions. If a crack occurs in an aircraft's main spar, it can generate many problems, including a lift time reduction. To solve this problem, it was necessary to perform an analysis of fatigue crack growth in the fatigue critical locations. Much time and expense is involved in generating the stress needed for a crack propagation analysis over a long period of time to obtain the amount of data required for an actual aircraft. In this paper, an algorithm is developed that can calculate the spectrum of stress over a long period of time for a mission by the Southwest Research Institute, which is based on the short-time load factor data produced using the peak-valley cycle counting method.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.21-28
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• 1995

In this paper an intonation conversion method is presented which provides the basic study on converting the original speech into the artificially intoned one. This method employs the other speaker's excitation signals as intonation information and the original vocal tract spectra, which are warped with the other speaker's ones by using DTW. as vocal features, and intonation converted speech signals are synthesized through short-time inverse Fourier transform(STIFT) of their product. To evaluate the intonation converted speech by this method, we collect Korean single vowels and sentences spoken by 30 males and compare fundamental frequency contours spectrograms, distortion measures and MOS test between the original speech and the converted one. The result shows that this method can convert and speech into the intoned one of the other speaker's.

• Journal of the Korean Society of Manufacturing Technology Engineers
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• v.19 no.3
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• pp.407-413
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• 2010

This paper reports on the research which analyzes acoustic signals acquired in progressive compressing, hole blanking, and burr compacting process. An acoustic sensor was set on the bed of hydraulic press. Acoustic signal is generated from progressive stamping process. First the signal acquired from the unit process; compressing, blanking or compacting, is studied by Fourier Transform and Short Time Fourier Transform. The blanking process emitted ultrasonic signal with more than 20kHz, but the compressing and compacting processes emitted acoustic signals with lower than 10kHz. The combined signals periodically acquired right after the tool grinding were then analyzed. 70-80kHz signals appeared in time-frequency domain, but not in the frequency domain, the magnitude of which was related to the tool wear. Short Time Fourier Transform made up for the Fourier Transform in analyzing the emitted signal for stamping process in the ultrasonic domain.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.1
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• pp.333-342
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• 2012

This paper presents a vowel onset point (VOP) detection method based on the human auditory system. This method maps the "perceptual" frequency scale, i.e. Mel scale onto a linear acoustic frequency, and then establishes a series of Triangular Mel-weighted Filter Bank simulate the function of band pass filtering in human ear. This nonlinear critical-band filter bank helps greatly reduce the data dimensionality, and eliminate the effect of harmonic waves to make the formants more prominent in the nonlinear spaced Mel spectrum. The sum of mel spectrum peaks energy is extracted as feature for each frame, and the instinct at which the energy amplitude starts rising sharply is detected as VOP, by convolving with Gabor window. For the single-word database which contains 12 vowels articulated with different kinds of consonants, the experimental results showed a good average detection rate of 72.73%, higher than other vowel detection methods based on short-time energy and zero-crossing rate.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.3
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• pp.465-470
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• 2021

A robust speech recognition technology is required that does not degrade the performance of speech recognition and the quality of the speech when speech recognition is performed in an actual environment of the speech mixed with noise. With the development of such speech recognition technology, it is necessary to develop an application that achieves stable and high speech recognition rate even in a noisy environment similar to the human speech spectrum. Therefore, this paper proposes a speech enhancement algorithm that processes a noise suppression based on the MMSA-STSA estimation algorithm, which is a short-time spectral amplitude method based on the error of the least mean square. This algorithm is an effective nonlinear speech enhancement algorithm based on a single channel input and has high noise suppression performance. Moreover this algorithm is a technique that reduces the amount of distortion of the speech based on the statistical model of the speech. In this experiment, in order to verify the effectiveness of the MMSA-STSA estimation algorithm, the effectiveness of the proposed algorithm is verified by comparing the input speech waveform and the output speech waveform.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.11
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• pp.2548-2554
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• 2010

This paper describes a design of fast Fourier transform(FFT) processor for EEG(electroencephalogram) signal analysis for health care services. Hamming window function with 1/2 overlapping is adopted to perform short-time FFT(ST-FFT) of a long period EEG signal occurred in real-time. In order to analyze efficiently EEG signals which have frequency characteristics in the range of 0 Hz to 100 Hz, a 256-point FFT processor is designed, which is based on a single-memory bank architecture and the radix-4 algorithm. The designed FFT processor has been verified by FPGA implementation, and has high accuracy with arithmetic error less than 2%.