• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.60-67
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• 2014

The purpose of this study is to suggest a new concept of the absolute sound level system in order to basically solve all the irrational aspects from the mastering stage to the output stage during the process of producing a sound source. Through this absolute sound level system, all the input absolute sound levels of sound sources become standardized based on 60 dB(S), and then sound source producers can implement a sound level they intend by using the differential sound level tag. Besides, by matching the output absolute sound level with the input absolute sound level, a target sound level can be implemented as a listener intends. With this system introduced, sound source producers are able to focus on the completion of a sound source itself without an unnecessary competition for sound levels, and only by inputting the differential reduction tag, they can implement a sound level balance as intended. At the same time, listeners are able to appreciate all kinds of sound sources fit for the standard of absolute s ound levels they want to listen to, without a process of trial and error, no matter what kinds of listening environments and sound systems they have.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.8-15
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• 2008

In this paper, in order to analyze property of frequency response in optical microphone, system was implemented. The capacitance microphone and fiber-optic transmission path type fiber-optic microphone (FOM) have weaknesses in directivity, size, weight, and price. However suggested optical microphone can be constituted by cheap devices, so it has many benefits like small size, light weight, high directivity, etc. Head part of optical microphone which is suggested in this paper is movable back and forth by sound pressure with the attached reflection plate. Operating point is determined by measuring the respond characteristics and choosing the point on which has maximum linearity and sensitivity while changing the distance between optical head and vibrating plate. We measured the output of the O/E transformed signal of the optical microphone while frequency of sound signal is changed using sound measurement/analysis program, Smaart Live and USBPre, which are based on PC, and compared the result from an existing capacitance microphone. The measured Optical microphone showed almost similar output characteristics as those of the compared condenser microphone, and its bandwidth performance was about 300[Hz]-3[kHz] at up to 3 [dB].

• The Journal of the Acoustical Society of Korea
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• v.13 no.4
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• pp.60-68
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• 1994

In this paper we propose a new Hybrid learning rule applied to multilayer feedforward neural networks, which is constructed by combining Hebbian learning rule that is a good feature extractor and Back-Propagation(BP) learning rule that is an excellent classifier. Unlike the BP rule used in multi-layer perceptron(MLP), the proposed Hybrid learning rule is used for uptate of all connection weights except for output connection weigths becase the Hebbian learning in output layer does not guarantee learning convergence. To evaluate the performance, the proposed hybrid rule is applied to classifier problems in two dimensional space and shows better performance than the one applied only by the BP rule. In terms of learning speed the proposed rule converges faster than the conventional BP. For example, the learning of the proposed Hybrid can be done in 2/10 of the iterations that are required for BP, while the recognition rate of the proposed Hybrid is improved by about $0.778\%$ at the peak.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.64-72
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• 1997

In this paper, study and experiments are performed for finding recognition unit fit which can be used in large vocabulary recognition system. Specifically, a phoneme that is currently used as recognition unit and a syllable in which Korean is well characterized are selected. From comparisons of recognition experiments, the study is performed whether a syllable can be considered as recognition unit of Korean recognition system. For report of an objective result of the comparison experiment, we collected speech data of a male speaker and processed them by hand-segmentation for phoneme boundary and labeling to construct speech database. And for training and recognition based on HMM, we used HTK (HMM Tool Kit) 2.0 of commercial tool from Entropic Co. to experiment in same condition. We applied two HMM model topologies, 3 emitting state of 5 state and 6 emitting state of 8 state, in Continuous HMM on training of each recognition unit. We also used 3 sets of PBW (Phonetically Balanced Words) and 1 set of POW(Phonetically Optimized Words) for training and another 1 set of PBW for recognition, that is "Speaker Dependent Medium Vocabulary Size Recognition." Experiments result reports that recognition rate is 95.65% in phoneme unit, 94.41% in syllable unit and decoding time of recognition in syllable unit is faster by 25% than in phoneme.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.121-124
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• 2001

음성 합성기의 합성음의 자연감을 높이기 위해 자연음에 내재하는 정확한 운율 법칙을 구하여 음성합성 시스템에서 이를 구현해 주어야 한다 무제한 어휘 음성합성 시스템의 문-음성 합성기에서 필요한 운율 법칙은 언어학적 정보를 이용해 구하거나, 자연음에서 추출하고 있다 그러나 추출한 운율 법칙이 자연음에 내재하는 모든 운율 법칙을 반영하지 못했거나, 잘못 구현되는 경우에는 합성음의 자연성이 떨어지게 된다. 이런 점을 고려하여 본 논문에서는 한국어 자연음을 분석하여 추출한 운율 정보를 인공 신경망이 학습하도록 하고 훈련을 마친 인공 신경망에 문장을 입력하고, 출력으로 나오는 운율 정보와 자연음의 운율 정보를 비교한 결과 제안한 인공 신경망이 자연음에 내재하고 있는 운율을 학습할 수 있음을 알 수 있었다. 운율의 3대 요소는 피치 , 지속시간, 크기의 변화이다. 제안한 인공 신경망이 한국어 문장의 음소 열을 입력으로 받아들이고, 각 음소의 지속시간에 따른 피치변화와 크기 변화를 출력으로 내보내면 자연음을 분석해 구한 각 음소의 운율 정보인 목표 패턴과 출력 패턴 의 오차를 최소화하도록 인공 신경망의 가중치를 조절할 수 있도록 설계하였다. 지속시간에 따른 각 음소의 피치와 크기 변화를 학습시키기 위해 피치 및 크기 인공 신경망을 구성하였다. 이들 인공 신경망을 훈련시키기 위해 먼저 음소 균형 문장 군을 구축하여야 하고, 이들 언어 자료를 특정 화자가 일정 환경에서 읽고 이를 녹음하여 , 분석하여 구한운율 정보를 운율 데이터베이스로 구축하였다. 문장 내의 각 음소에 대해 지속 시간과 피치 변화 그리고 크기 변화를 구하고, 곡선 적응 방법을 이용하여 각 변화 곡선에 대한 다항식 계수와 초기 값을 구해 운율 데이터베이스를 구축한다. 이 운율 데이터베이스의 일부는 인공 신경망을 훈련시키는데 이용하고, 나머지로 인공 신경망의 성능을 평가하여 인공 신경망이 운율 법칙을 학습할 수 있었다. 언어 자료의 문장 수를 늘리고 발음 횟수를 늘려 운율 데이터베이스를 확장하면 인공 신경망의 성능을 높일 수 있고, 문장 내의 음소의 수를 감안하여 인공 신경망의 입력 단자의 수는 계산량과 초분절 요인을 감안하여 결정해야 할 것이다

• The Journal of the Acoustical Society of Korea
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• v.39 no.5
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• pp.476-482
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• 2020

Sequence-to-sequence deep neural networks with attention mechanisms have shown superior performance across various domains, where the sizes of the input and the output sequences may differ. However, if the input sequences are much longer than the output sequences, and the characteristic of the input sequence changes within a single output token, the conventional attention mechanisms are inappropriate, because only a single context vector is used for each output token. In this paper, we propose a double-attention mechanism to handle this problem by using two context vectors that cover the left and the right parts of the input focus separately. The effectiveness of the proposed method is evaluated using speech recognition experiments on the TIMIT corpus.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.509-517
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• 2013

In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.48 no.2
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• pp.127-134
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• 2020

High supersonic aircraft are exposed to high temperature environments by aerodynamic heating during supersonic flight. Thermal protection system structures such as double-panel structures are used on the skin of the fuselage and wings to prevent the transfer of high heat into the interior of an aircraft. The thin-walled double-panel skin can be exposed to acoustic loads by supersonic aircraft's high power engine noise and jet flow noise, which can cause sonic fatigue damage. Therefore, it is necessary to examine the behavior of supersonic aircraft skin structure under thermal-acoustic load and to predict fatigue life. In this paper, we designed and fabricated thermal-acoustic test equipment to simulate thermal-acoustic load. Thermal-acoustic testing of the titanium specimen under thermal-acoustic load was performed. The analytical model was verified by comparing the thermal-acoustic test results with the finite element analysis results.

• Journal of the Korean Society of Safety
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• v.12 no.4
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• pp.214-219
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• 1997

The mechanical seals, which are installed in rotating machines like pump and compressor, are generally used as sealing devices in the many fields of industries. The failure of mechanical seals such as leakage, crack, breakage, fast and severe wear, excessive torque, and squeaking results in big problems. To identify abnormal phenomena on mechanical seals and to propose the proper monitoring parameter for the failure of mechanical seals, sliding wear experiments were conducted. Acoustic emission, torque, and temperature were measured during experiments. Optical microstructure was observed for the wear processing after every 10 minute sliding at rotation speed of 1750 rpm and scanning electron microscopy was also observed. Except for the initial part of every experiment, the variation of acoustic emission was well coincided with torque variation during the experiments. This study concludes that acoustic emission and torque are proper monitoring parameters for the failure of mechanical seals. The intensity of acoustic emission signals is measured in root mean square voltage. Temperature of sealing face will be used as a parallel parameter for increasing the reliability of monitoring system.

• The Journal of the Acoustical Society of Korea
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• v.41 no.6
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• pp.610-620
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• 2022

An acoustic vector sensor can simultaneously receive vector quantities, such as particle velocity and acceleration, as well as acoustic pressure at one location, and thus it can be used as a single input multiple output receiver in underwater acoustic communication systems. On the other hand, vector signals received by a single vector sensor have different channel characteristics due to the azimuth angle between the source and receiver and the difference in propagation angle of multipath in each component, producing different communication performances. In this paper, we propose a channel parameter-based weighting method to improve the performance of an acoustic communication system using a single vector sensor. To verify the proposed method, we used communication data collected from the experiment conducted during the KOREX-17 (Korea Reverberation Experiment). For communication demodulation, block-based time reversal technique which is robust against time-varying channels were utilized. Finally, the communication results showed that the effectiveness of the channel parameter-based weighting method for the underwater communication system using a single vector sensor was verified.