• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.

• Transactions of the Korean Society of Mechanical Engineers B
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• v.36 no.6
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• pp.655-662
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• 2012

Gas leakage in an oil refinery causes damage to the environment and unsafe conditions. Therefore, it is necessary to develop a technique that is able to detect the location of the leakage and to filter abnormal gas-leakage signals from normal background noise. In this study, the adaptation filter of the finite impulse response (FIR) least mean squares (LMS) algorithm and a cross-correlation function were used to develop a leakage-predicting program based on LABVIEW. Nitrogen gas at a high pressure of 120 kg/$cm^2$ and the assembled equipment were used to perform experiments in a reverberant chamber. Analysis of the data from the experiments performed with various hole sizes, pressures, distances, and frequencies indicated that the background noise occurred primarily at less than 1 kHz and that the leakage signal appeared in a high-frequency region of around 16 kHz. Measurement of the noise sources in an actual oil refinery revealed that the noise frequencies of pumps and compressors, which are two typical background noise sources in a petrochemical plant, were 2 kHz and 4.5 kHz, respectively. The fact that these two signals were separated clearly made it possible to distinguish leakage signals from background noises and, in addition, to detect the location of the leakage.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.375-382
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• 2003

Lip-reading technique based on bimodal is to enhance speech recognition rate in noisy environment. It is most important to detect the correct lip-image. But it is hard to estimate stable performance in dynamic environment, because of many factors to deteriorate Lip-reading's performance. There are illumination change, speaker's pronunciation habit, versatility of lips shape and rotation or size change of lips etc. In this paper, we propose the IIR filtering in time-domain for the stable performance. It is very proper to remove the noise of speech, to enhance performance of recognition by digital filtering in time domain. While the lip-reading technique in whole lip image makes data massive, the Principal Component Analysis of pre-process allows to reduce the data quantify by detection of feature without loss of image information. For the observation performance of speech recognition using only image information, we made an experiment on recognition after choosing 22 words in available car service. We used Hidden Markov Model by speech recognition algorithm to compare this words' recognition performance. As a result, while the recognition rate of lip-reading using PCA is 64%, Time-domain filter applied to lip-reading enhances recognition rate of 72.4%.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.402-409
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• 2005

In this paper. we propose a coding method using a matching pursuit algorithm in a strongly periodic highband signal. Also. we propose an efficient quantizer for the estimated parameters : spectral magnitude and phase. Based on the error concealment principle and sinusoidal model. the MP algorithm requires the high-precision pitch period estimation. To estimate more accurate pitch period. the refined pitch obtained from lowband speech is used. which increases the efficiency of bit allocation. The spectral magnitude parameters are quantized by the method which is combined with MDCT (Modified Discrete Cosine Transform) and multi-stage structure. The spectral phase quantizer uses the $2{\pi}$ modular characteristic of phases and the weighted function by spectral magnitudes. To evaluate the efficiency of the proposed method. we applied it to analysis-by-synthesis system. Furthermore we suggest the possibillity of scalable wideband speech codecs based on band-split structure.

• Journal of Korean Society of Disaster and Security
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• v.14 no.2
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• pp.43-50
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• 2021

In Korea, the frequency of soil disasters has soared recently due to increased torrential rains caused by abnormal weather conditions. In particular, soil generated from mountainous areas is flowing into small rivers along valleys, depositing rivers and adding to flood damage. In order to prevent damage from such soil disasters, it is important to predict sediments and to quantitatively identify bed load. In this work, we conducted an experiment to indirectly measure acoustic sensor-based bed load collision sounds using pipe hydrophones, and compared them with raw data by applying denoising methods to improve the reliability of the measured data. As a result, we derive results in a more clear analysis of bed load estimation by correcting noise when the denoising method is applied to raw data.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.987-994
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• 2020

In this paper, we studied the effect of kernel size of CNN layer on performance in acoustic noise attenuators. This system uses a deep learning algorithm using a neural network adaptive prediction filter instead of using the existing adaptive filter. Speech is estimated from a single input speech signal containing noise using a 100-neuron, 16-filter CNN filter and an error back propagation algorithm. This is to use the quasi-periodic property in the voiced sound section of the voice signal. In this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed to verify the performance of the noise attenuator for the kernel size. As a result of the simulation, when the kernel size is about 16, the MSE and MAE values are the smallest, and when the size is smaller or larger than 16, the MSE and MAE values increase. It can be seen that in the case of an speech signal, the features can be best captured when the kernel size is about 16.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.331-339
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• 2016

This paper suggests an estimation method of the source signal and the channel impulse response (CIR) using ray-based blind deconvolution (RBD) in the underwater acoustic channel environment where Doppler effect exists by the relative motion between source and receiver. It is difficult to estimate the CIR on Doppler effect by the matched filter with a highly Doppler-sensitive waveform such as the m-sequence signal because Doppler shift can severely degrade the correlation between the received signal corrupted by Doppler effect and the original source signal. In this study, the Doppler-shifted source-signal's phase is estimated using the RBD, and the received signal is compensated by it to obtain the Doppler-corrected CIR. It is verified that using the matched filter with the received signal from the experimental data fails to estimate the CIR while the obtained CIR by the suggested method has the similarity to the propagation path of the ray model. Also, the results show that the reconstructed source signal using the RBD has the better Doppler shift compensation than the Doppler-shifted source signal derived from scattering function.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1991.04a
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• pp.17-22
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• 1991

최근 기계장치로부터 발생하는 소음을 감소시키는 새로운 방법으로서, 능동 적으로 소음을 제어하는 방법에 대한 연구가 활발히 진행되고 있다. 이것은 원하지 않는 소음을 그 신호의 역위상을 갖는 부가음을 이용하여 능동적으 로 감쇠시키는 방법으로서, 저주파수 대역에서 비효율적인 수동적인 방법인 소음기둥에 대한 대안으로 많은 학자들의 관심의 대상이 되어 왔다. 초기에 는 소음을 줄이기 위해 요구되는 여러가지 음향요소의 전달함수를 제어하는 데 대한 불가능성으로 인해 능동 소음제어에 대한 실질적인 발전이 지연되 어 왔으나 최근 마이크로 컴퓨터를 비롯한 전자공학의 발전으로 인해 적응 신호처리 분야가 등장하게 되었으며, 음향계의 소음을 원하는 수준까지 제어 하는 능동 소음제어의 실시간 구현이 가능하게 되었다. 그 중에서도 음이 1 차원적으로 전파한다고 볼 수 있는 길이가 긴 덕트구조물에서의 능동 소음 제어는 가장 기본적이며 현실적으로 자동차 배기계나 냉동.공조설비에 있어 서 실용적으로 적용할 수 있는 문제임 만큼 많은 연구가 이루어지고 있다. 이러한 능동 소음제어 방법을 음향계에 적용하였을 때, 부가적인 음을 발생 하는 제어용 스피커로 인해 입력마이크로폰으로의 음향궤환이 존재하고 이 에 따라 제어계가 불안정해질 수 있으며, 또한 변환기의 사용으로 인한 부가 적인 전달함수가 존재하므로 이에 대한 중요한 의미를 갖고 고려하여야 한 다. 본 연구에서는 적응 필터링 이론에 의한 소음원의 입력신호에 대한 최적 한 예측으로써 부가음을 발생시키고, 입력신호 및 제어된 출력신호간의 차인 오차를 최소화 시키도록 하는 오차적응제어법을 이용한 능동소음 제어 방법 을 제시하였다. 이와 아울러 제어계의 환경변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해야 하는 적응 소음제어 시스템에서, 음향궤환과 함 께 필히 고려해야 하는 부가적인 전달함수의 영향을 고려한 능동 소음제어 에 대해 연구하였다. 경량화 추세에 따라 지반이나 케이싱이 경량이거나 유연하여 회전축과 동적으로 연성된 경우 회전축-베어링-지반으로 이루어진 2중구조의 회전축 계 동특성을 해석할 수 있는 프로그램을 개발하므로서 회전 기계류의 진동 전반에 걸친 문제점에 대한 그 원인과 현상을 명확히 분석하여 국내의 전기 계류의 보다 신뢰성있는 설계 및 제작자료를 확보하는데 기여할 수 있게 하 였다.존의 small molecular Gd-chelate에 비해 매우 큼을 알 수 있었다. MnPC는 간세포에 흡수된 후 담도계로 배출되는 간특이성 조영제임을 확인하였다. 장비 내에서 반복 시행한 평균값의 차이는 대체적으로 유의한 차이가 없었으나, 다른 장비에서 반복 시행한 장비간의 사이에는 유의한 차이가 있는 경우가 더 많았다. 따라서 , MRS 검사를 소뇌나 뇌교의 어떤 절환에 적용하기 전에 각 장비 마다 정상 기준치를 반드시 얻은 후에 이상여부를 판 정하는 것이 필수적이라고 생각된다.EX> 이상이 적절한 진단기준으로 생각되었다. $0.4{\;}\textrm{cm}^3$ 이상의 좌우 부피차를 보이는 모든 증례에서 육안적으로도 해마위축이 뚜렷이 나타났다. 결론 : MR영상을 이용한 해마의 부피측정은 해마경화증 환자의 진단에 있어 육안적인 MR 진단이 어려운 제한된 경우에만 실제적 도움을 줄 수 있는 보조적인 방법으로 생각된다.ofile whereas relaxivity at high field is not affected by τS. On the other hand, the change in τV does not affect low field profile but strongly in fluences on both inflection fie이 and the maximum relaxivity value. The results shows a fluences on both inflection field

• The Journal of the Acoustical Society of Korea
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• v.8 no.1
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• pp.5-12
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• 1989

This paper is concerned with the characteristics on the statistical parameters of vibration signal from bearing with changing its operating conditions as well as the spreading of faults. The rms, Kurtosis, crest factor, probability of exceedance and probability density function have been chose as the statistical parameters. To characterize of each, vibration signals have been recorded from four ball tester at different loads, operation speeds and time. The values of the statistical parameters for each frequency band have been calculated after A/D conversion and digital filtering of the recorded signals. It has been found that unlike rms values the statistical parameters such as Kurtosis etc. are almost unchanging with the change of the operating conditions such as load and speed. This suggests that the statistical parameters may be used for determining the development of faults independent of the operating conditions. In fact, the statistical parameters deviate considerably from their respective normal values when the faults developed under load conditions in the samples, conforming the suggestion.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.665-670
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• 2002

The most common form of pre-emphasis is y(n)=s(n)-As(n-1), where A typically lies between 0.9 and 1.0 in voiced signal. Also, 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 interval information of LSP to estimate formant frequency, After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed method flattened the weaked high frequency components effectively. That is, we could improve the flattening characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.