• The Journal of the Acoustical Society of Korea
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• v.39 no.1
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• pp.38-46
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• 2020

LOw Frequency Analysis Recording (LOFAR) and Demodulation of Envelop Modulation On Noise (DEMON) grams are bearing-time-frequency plots of underwater acoustic signals, to visualize features for passive sonar. Those grams are characterized by tonal components, for which conventional data coding methods are not suitable. In this work, a novel LOFAR/DEMON gram compression algorithm based on binary map and prediction methods is proposed. We first generate a binary map, from which prediction for each frequency bin is determined, and then divide a frame into several macro blocks. For each macro block, we apply intra and inter prediction modes and compute residuals. Then, we perform the prediction of available bins in the binary map and quantize residuals for entropy coding. By transmitting the binary map and prediction modes, the decoder can reconstructs grams using the same process. Simulation results show that the proposed algorithm provides significantly better compression performance on LOFAR and DEMON grams than conventional data coding methods.

• Composites Research
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• v.16 no.4
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• pp.74-79
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• 2003

Comparison of interfacial properties and microfailure mechanisms of oxygen-plasma treated poly(p-phenylene-2,6-benzobisoxazole(PBO. Zylon) and poly(p-phenylene terephthalamide)(PPTA, Kevlar) fibers/ epoxy composites were investigated using micromechanical technique and nondestructive acoustic emission(AE). Interfacial shear strength(IFSS) and work of adhesion, Wa of PBO or Kevlar fibers/epoxy composites increased by oxygen-plasma treatment. Plasma-treated Kevlar fiber shooed the maximum critical surface tension and polar term, whereas the untreated PBO fiber showed the minimum value. Microfibril fracture pattern of plasma-treated Kevlar fiber appeared obviously. Based on the propagation of microfibril failure toward core region. the number of AE events for plasma-treated PBO and Kevlar fibers increased significantly. The results oi nondestructive AE were consistent well with microfailure modes by optical observation in microdroplet and two-fiber composites tests.

• Journal of Advanced Navigation Technology
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• v.24 no.5
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• pp.393-407
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• 2020

In order to reduce the threat by illegal unmanned aerial vehicles, a tracking system based on sound was implemented. There are three main points to the drone acoustic tracking method. First, it scans the space through variable beam formation to find a sound source and records the sound using a microphone array. Second, it classifies it into a hidden Markov model (HMM) to find out whether the sound source exists or not, and finally, the sound source is In the case of a drone, a sound source recorded and stored as a tracking reference signal based on an adaptive beam pattern is used. The simulation was performed in both the ideal condition without background noise and interference sound and the non-ideal condition with background noise and interference sound, and evaluated the tracking performance of illegal drones. The drone tracking system designed the criteria for determining the presence or absence of a drone according to the improvement of the search distance performance according to the microphone array performance and the degree of sound pattern matching, and reflected in the design of the speech reading circuit.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.42 no.4 s.304
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• pp.45-50
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• 2005

In digital hearing aid system, one of the main problems is acoustic feedback which is known as howling because of miniaturization md high-gain amplification. In this paper, we proposed a feedback cancellation algorithm for hearing aid using time delay and time-varying decorrelation filter. The proposed algorithm has a kind of adaptive filter structure, which is combined with time delay and time-varying decorrelation filter to improve feedback cancellation. An all pass filter was implemented as the time-varying decorrelation filter using low frequency modulator. From the result of computer simulation, it is verified that the proposed algorithm has good ability to cancel feedback.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.11-20
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• 2005

In this paper, we deal with the research about a S/PDIF (Sony Philips Digital Interface) receiver which can operate without PLL (Phase Locked Loop) circuits. Although a S/PDIF receiver is used in most audio devices and audio processors in these days. yet there are only few domestic researches about S/PDIF. Currently used commercial DACs (Digital-to-Analog Converters) which can decode S/PDIF signals, have a PLL circuit inside them. The PLL makes it possible to extract clock information from S/PDIF digital signal and to synchronize a clock signal with input signals. But the PLL circuit makes many diffculties in designing the SOC (System On Chips) of VLSIs (Vew Large Scale Integrated Ciruits) because it is an "analog circuit". We proposed a S/PDIF receiver which doesn't have PLL circuits and only has Pure digital circuits. The key idea of the proposed S/PDIF receiver. is to use the ratio between a 16 MHz basic input clock and S/PDIF signals. After having decoded hundreds thousands S/PDIF inputs, it went to prove that a S/PDIF receiver can be designed with pure digital circuits and without any analog circuits such as PLL circuits. We have confidence that the proposed S/PDIF receiver can be used as an IP (Intellectual Property) for the SOC design of the digital circuits.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.12-15
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• 1996

An array often has faulty elements in real operation. The faulty elements, producing no output or highly reduced gain than other normal elements, cause an elevated sidelobe level and fail to reject the interference signals in an adaptive beamformer. In this paper we have presented the beamforming algorithm for arrays with faulty elements. In the ideal case, an autocorrelation matrix computed from array output data is the toeplitz. However, the inverse of the autocorrelation matrix computed from array with faulty elements can not be obtained due to deficient values of matrix. To overcome this problem, an adaptive beamforming algorithm using the average values of the diagonal terms of matrix is proposed. The computer simulations have been performed to study the performance of the presented method. We have been able to solve the degrees-of-freedom problem that is the drawback of the previous subaperture processing technique.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.56-62
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• 2001

In this paper, we adopt a missing data theory to speech recognition. It can be used in order to maintain high performance of speech recognizer when the missing data occurs. In general, hidden Markov model (HMM) is used as a stochastic classifier for speech recognition task. Acoustic events are represented by continuous probability density function in continuous density HMM(CDHMM). The missing data theory has an advantage that can be easily applicable to this CDHMM. A marginalization method is used for processing missing data because it has small complexity and is easy to apply to automatic speech recognition (ASR). Also, a spectral subtraction is used for detecting missing data. If the difference between the energy of speech and that of background noise is below given threshold value, we determine that missing has occurred. We propose a new method that examines the reliability of detected missing data using voicing probability. The voicing probability is used to find voiced frames. It is used to process the missing data in voiced region that has more redundant information than consonants. The experimental results showed that our method improves performance than baseline system that uses spectral subtraction method only. In 452 words isolated word recognition experiment, the proposed method using the voicing probability reduced the average word error rate by 12％ in a typical noise situation.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.555-558
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• 2006

감정을 포함하고 있는 음성은 청자로 하여금 화자의 심리상태를 파악할 수 있게 하는 요소 중에 하나이다. 음성신호에 포함되어 있는 감정을 인식하여 사람과 로봇과의 원활한 감성적 상호작용을 위하여 특징을 추출하고 감정을 분류한 방법을 제시한다. 음성신호로부터 음향정보 및 운율정보인 기본 특징들을 추출하고 이로부터 계산된 통계치를 갖는 특징벡터를 입력으로 support vector machine (SVM) 기반의 패턴분류기를 사용하여 6가지의 감정- 화남(angry), 지루함(bored), 기쁨(happy), 중립(neutral), 슬픔(sad) 그리고 놀람(surprised)으로 분류한다. SVM에 의한 인식실험을 한 경우 51.4%의 인식률을 보였고 사람의 판단에 의한 경우는 60.4%의 인식률을 보였다. 또한 화자가 판단한 감정 데이터베이스의 감정들을 다수의 청자가 판단한 감정 상태로 변경한 입력을 SVM에 의해서 감정을 분류한 결과가 51.2% 정확도로 감정인식하기 위해 사용한 기본 특징들이 유효함을 알 수 있다.

• Proceedings of the KSME Conference
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• 2004.11a
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• pp.601-606
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• 2004

Vibration spectrums of the test loop according to flow conditions were analyzed in order to identify the sources of vibration at peak frequencies. While a flow condition of the sweep test was changed by varying pump rotational speed from 450 rpm to 1500 rpm by the step 150 rpm, midspan acceleration of the test section in width-direction and dynamic pressure perturbation in the test section were measured. Other sources of vibration due to the flow structure interactions, such as acoustic resonance, blade pulsing frequency and bellows wrinkles, were investigated. Pressure perturbation in the section and acoustic resonance due to branch pipe give major effects to the vibration of the test section in high frequency range of 1.5 kHz to 2.8 kHz.

• Journal of the Korean Society of Manufacturing Technology Engineers
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• v.6 no.4
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• pp.124-129
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• 1997

The experimental investigation is mainly focused to detect tool wear by cutting signal measurements in multi-insert face milling SS 41 and STS 304. This research have investigated the effects on the insert number, which has relationship with mean-cutting force. AE(acoustic emission) signal, tool life and surface roughness in machining SS41 and STS 304. The cutting force and AE signal are monitored to analyse the cutting process, The surface roughness of the specimens machined by TiN coated tool with the various insert numbers measured at various cutting speeds, feed rates and depths of cut, The width of flank wear is also observed.