• The Journal of the Acoustical Society of Korea
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• v.33 no.6
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• pp.400-406
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• 2014

This paper describes a robust watermark decoding model using a SVM(Support Vector Machine). First, the embedding process is performed inversely for a watermarked signal. And then the watermark is extracted using the proposed model. For SVM training of the proposed model, data are generated that are watermarks extracted from sounds containing watermarks by four different embedding schemes. BER(Bit Error Rate) values of the data are utilized to determine a threshold value employed to create training set. To evaluate the robustness, 14 attacks selected in StirMark, SMDI and STEP2000 benchmarking are applied. Consequently, the proposed model outperformed previous method in PSNR(Peak Signal to Noise Ratio) and BER. It is noticeable that the proposed method achieves BER 1% below in the case of PSNR greater than 10 dB.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.439-451
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• 2021

We propose a deep learning-based beamformer combined with spectral subtraction for continuous speech recognition operating in noisy environments. Conventional beamforming systems were mostly evaluated by using pre-segmented audio signals which were typically generated by mixing speech and noise continuously on a computer. However, since speech utterances are sparsely uttered along the time axis in real environments, conventional beamforming systems degrade in case when noise-only signals without speech are input. To alleviate this drawback, we combine online beamforming algorithm and spectral subtraction. We construct a Continuous Speech Enhancement (CSE) evaluation set to evaluate the online beamforming algorithm in noisy environments. The evaluation set is built by mixing sparsely-occurring speech utterances of the CHiME3 evaluation set and continuously-played CHiME3 background noise and background music of MUSDB. Using a Kaldi-based toolkit and Google web speech recognizer as a speech recognition back-end, we confirm that the proposed online beamforming algorithm with spectral subtraction shows better performance than the baseline online algorithm.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.192-201
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• 2016

Condition Monitoring System (CMS) has been used to detect unexpected faults of wind turbine caused by the abrupt change of circumstances or the aging of its mechanical part. In fact, it is a very hard work to do regular inspection for its maintenance because wind turbine is located on the mountaintop or sea. The purpose of this study is to find out distribution patterns of vibration signals measured from the main mechanical parts of wind turbine according to its operation condition. To this end, acceleration signals of main bearing, gearbox, generator, wind speed, rotational speed, etc were measured through the long period more than 2 years and trend analyses on each signal were conducted as a function of the rotational speed. In addition, correlation analysis among the signals was done to grasp the relation between mechanical parts. As a result, the vibrations were dependent on the rotational speed of main shaft and whether power was generated or not, and their distributions at a specific rotational speed could be approximated to Weibull distribution. It was also investigated that the vibration at main bearing was correlated with vibration at gearbox each other, whereas vibration at generator should be dealt with individually because of generating mechanism. These results can be used for improving performance of CMS that early detects the mechanical abnormality of wind turbine.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.3
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• pp.78-86
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• 2017

In this paper, we propose hidden object detection system using parametric array based on acoustic signal that is harmless to human body. A transmit signal of the proposed detection system uses a high directive chirp signal generated from parametric array phenomenon, which uses technique to improve a signal to noise (SNR) of a received signal and a distance resolution trough the dechirp processing. The transmit sensor array is constructed as $8{\times}2$ and has a horizontal beam width of $7^{\circ}$ and vertical beam width of $26^{\circ}$. To verify the detection and visualization of the proposed system, a 2-axis driving control system based on linear stage was constructed, and A-scan, B-scan, and C-scan experiments was addressed for hidden object. From experimental results, we detected and visualized the hidden bronze plate and pipe by cloth and the visualized shapes was confirmed. Especially, the obtained errors was $0.015m^2$ for bronze plate, and $0.046m^2$ for pipe.

• Journal of Advanced Navigation Technology
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• v.14 no.4
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• pp.571-577
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• 2010

Driving aircraft requires extremely complicated and detailed information processing. Pilots perform their tasks by selecting the information relevant to them. In this processing, spatial information presented simultaneously through crossmodal link is advantageous over the one provided in singular sensory mode. In this paper, probability to apply providing visual spatial information along with auditory information to enemy tracking system in aircraft navigation is empirically investigated. The result shows that auditory spatial information, which is virtually created through HRTF is advantageous to visual spatial information alone in attention processing. The findings suggest auditory spatial information along with visual one can be presented through crossmodal link by utilizing stereophonic sound such as HRTF. which is available in the existing simple stereo system.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.135-144
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• 2003

A method for the automatic segmentation of speech signals is described. The method is dedicated to the construction of a large database for a Text-To-Speech (TTS) synthesis system. The main issue of the work involves the refinement of an initial estimation of phone boundaries which are provided by an alignment, based on a Hidden Market Model(HMM). Multi-layer perceptron (MLP) was used as a phone boundary detector. To increase the performance of segmentation, a technique which individually trains an MLP according to phonetic transition is proposed. The optimum partitioning of the entire phonetic transition space is constructed from the standpoint of minimizing the overall deviation from hand labelling positions. With single speaker stimuli, the experimental results showed that more than 95% of all phone boundaries have a boundary deviation from the reference position smaller than 20 ms, and the refinement of the boundaries reduces the root mean square error by about 25%.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2
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• pp.68-77
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• 2005

The purpose of a spatial audio system is to give a listener an impression as if he were present in a recorded environment when its sound is reproduced. For this purpose a dummy head microphone is generally used. Because of its human-like shape, dummy head microphone can reproduce spatial images through headphone reproduction. However, its shape and size are restriction to public use and it is difficult to convert the output signal of dummy head microphone into a multi-channel signal for multi-channel environment. So, in this paper, we propose a multi-channel 3D microphone technology. The multi-channel 3D microphone acquire a spatial audio using five microphones around a horizontal plane of a rigid sphere and through post processing, it can reproduce various reproduction signals for headphone, stereo, stereo dipole, 4ch and 5ch reproduction environments. Because of complex computation, we implemented H/W based post processing system. To verily the Performance of the multi-channel 3D microphone, localization experiments were Performed. The result shows that a front/back confusion, which is the one of common limitations of conventional dummy head technology, can be reduced dramatically.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.266-269
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• 2010

급속한 정보 통신 기술의 발달로 인해 멀티미디어 재생 개발 기술들은 단순히 수동적으로 보고 듣는 재생 기술에서 벗어나 청취자 감성, 취향 등에 따라 보다 실감 있고 사용자가 능동적으로 재생할 수 있는 기술로 진화 하고 있다. 지금까지의 오디오 서비스는 음원 개발자 중심의 오디오 서비스, 즉 보컬 및 모든 악기가 믹스된 단일음원이기 때문에 사용자는 단순히 오디오 음원 개발자나 음반 제작사가 발매한 단일 음원을 일방적으로 수동적 청취할 수밖에 없다. 하지만 사용자 능동형 오디오 서비스에서는 사용자가 능동적으로 자신이 원하는 음악적 취향에 따라 능동적으로 각각의 객체 기반의 독립 음원을 선택, 감성에 따른 음원 효과 추가, 최적의 음원 청취 위치(Sweet Spot) 변경, 음원 및 스피커 재생 공간 및 위치 변경 재생 등을 할 수가 있다. 본 논문에서는 디지털 음원들을 입력받아 임의의 필터링을 실행하고, 사용자 음원 보정 정보, 출력 유닛의 공간적, 음향적 특성을 상위제어기로부터 입력받아 전신호경로 상에 디지털 신호처리 하여 출력신호를 생성하는 DSP 시스템 플랫폼 및 알고리즘에 관하여 소개한다.

• Korean Journal of Materials Research
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• v.6 no.3
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• pp.245-252
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• 1996

본 연구는 화강암 폐재의 재활용을 목적으로 한 기초적인 연구로서, 분말형태의 화강암폐재를 Ca(OH2)와 혼합하여 수열 hot press 법에 의해 고화시켰다. 아울러 고화체의 기계적 성질을 평가하였으며, 미시적 조직구조의 변화 및 파괴거동을 파악하기 위해 음향방출실험을 실시하였다. 고화체의 기계적 성질은 수열온도의 의존성이 있었으며, 28$0^{\circ}C$에서 최대강도를 보였다. 또한 고화체의 파면은 수열온도에 따라 현저히 다른 양상을 보였으며, 수열실험동안 다양한 화합물이 생성되었다. 그 중에서 cyclowollastonite, tobermorite 및 rankinite 등은 강도를 향상시키는 주된 화합물이었고, crossite 및 xonotlite 등은 강도의 저하를 초래하였다. 한편, 기공이 많이 존재할수록 AE counts는 더많이 발생하였으며, 최대하중에서 AE counts는 최대치를 보였고, 강도가 증가함에 따라 AE신호는 보다 많이 방출되었다.

• 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.