• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.65-70
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• 2005

This paper deals with the problem of double-talk (DT) detection in anacoustic echo canceller (AEC). In the DT detection algorithm with correlation coefficient, detection errors occasionally occur because it is hard to set the threshold to distinguish DT from echo path change (EPC). Adaptive filter falls into the situation that it stops updating its filter coefficients when EPC is erroneously considered as DT at the starting-point of EPC. In addition, in case of echo path changing during the DT period, the end-point detection of DT period fails so that the AEC cannot update its filter coefficients for a while even after the DT period ends. To solve these problems, in this paper, we propose a novel AEC that employs an auxiliary filter. We use the idea that though the error signal cannot be estimated using reference signal in case or DT situation but it can be in case or EPC situation. The experimental result verifies that the proposed method could solve the problems caused by DT detection error or echo path change during the DT period.

• The Journal of the Acoustical Society of Korea
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• v.33 no.4
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• pp.232-237
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• 2014

A passive sonar of warship is composed of several directional or omni-directional sensors. In order to model the acoustic signal received into a warship sonar, the wave propagation modeling is usually required from arbitrary noise source to all sensors equipped to the sonar. However, the full calculation for all sensors is time-consuming and the performance of sonar simulator deteriorates. In this study, we suggest an asymptotic method to estimate the sonar signal arrived to sensors adjacent to the reference sensor, where it is assumed that all information of eigenrays is known. This method is developed using Taylor series for the time delay of eigenray and similar to Fraunhofer and Fresnel approximation for sonar aperture. To validate the proposed method, some numerical experiments are performed for the passive sonar. The approximation when the second-order term is kept is vastly superior. In addition, the error criterion for each approximation is provided with a practical example.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.559-567
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• 2020

This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm's limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.

• Journal of the Korean Society of Marine Environment & Safety
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• v.26 no.3
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• pp.297-307
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• 2020

In the study, energy flow analysis is performed to predict the performance of silencers. To date, deterministic approaches such as finite element method have been widely used for silencer analysis. However, they have limitations in analyzing large structures and mid-high frequency ranges due to unreasonable computational costs and errors. However, silencers used for ships and off-shore plants are much larger than those used in other engineering fields. Hence, energy governing equation, which is significantly efficient for systems with high modal density, is solved for silencers in ships and off-shore plants. The silencer is divided into two different acoustic media, air and absorption materials. The discontinuity of energy density at interfaces is solved via hypersingular integrals for the 3-D modified Helmholtz equation to analyze multi-domain problems with the energy flow boundary element method. The method is verified by comparing the measurements and analysis results for ship silencers over mid-high frequency ranges. The comparisons confirm good agreement between the measurement and analysis results. We confirm that the applied analysis method is useful for large silencers in mid-high frequency ranges. With the proven procedures, energy flow analysis can be performed for various types of silencer used in ships and off-shore plants in the first stage of the design.

• 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.

• 한국전산유체공학회:학술대회논문집
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• 2009.11a
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• pp.13-18
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• 2009

Flight vehicles such as wheel wells and bomb bays have many cavities. The flow around a cavity is characterized as an unsteady flow because of the formation and dissipation of vortices brought about by the interaction between the free stream shear layer and the internal flow of the cavity. The resonance phenomena can damage the structures around the cavity and negatively affect the aerodynamic performance and stability of the vehicle. In this study, a numerical analysis was performed for the cavity flows using the unsteady compressible three-dimensional Reynolds-Averaged Navier-Stokes (RANS) equation with Wilcox's turbulence model. The Message Passing Interface (MPI) parallelized code was used for the calculations by PC-cluster. The cavity has aspect ratios (L/D) of 2.5 ~ 7.5 with width ratios (W/D) of 2 ~ 4. The Mach and Reynolds numbers are 0.4 ~ 0.6 and $1.6{\times}106$, respectively. The occurrence of oscillation is observed in the "shear layer and transient mode" with a feedback mechanism. Based on the Sound Pressure Level (SPL) analysis of the pressure variation at the cavity trailing edge, the dominant frequencies are analyzed and compared with the results of Rossiter's formula. The dominant frequencies are very similar to the result of Rossiter's formula and other experimental data in the low aspect ratio cavity (L/D = ~ 4.5). In the large aspect ratio cavity, however, there are other low dominant frequencies due to the leading edge shear layer with the dominant frequencies of the feedback mechanism. The characteristics of the acoustic wave propagation are analyzed using the Correlation of Pressure Distribution (CPD).

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.22 no.1
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• pp.22-28
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• 2012

While a dash panel component, close to passengers, plays a very important role to protect heat and noise from a power train, it is also a main path that transfers vibration energy and eventually radiates acoustic noise into the cavity. Therefore, it is important to provide optimal design schemes incorporating sound packages such as a dash isolation pad and a floor carpet, as well as structures. The present study is the extension of the previous investigation how design variables affect sound radiation, which was carried out using the simple plate and framed system. A novel FE-SEA hybrid simulation model is used for this study. The system taken into account is a dash panel component of a sedan vehicle, which includes front pillars, front side members, a dash panel and corresponding sound packages. Design variables such as panel thicknesses and sound packages are investigated how they are related to two main NVH indexes, sound radiation power(i.e. structure-borne) and sound transmission loss(i.e. air borne). In the viewpoint of obtaining better NVH performance, it is shown that these two indexes do not always result in same tendencies of improvement, which suggests that they should be dealt with independently and are also dependent on frequency regions.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.21 no.9
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• pp.820-827
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• 2011

Recently, noise has been regarded as one of the most notorious and frequent environmental pollutions which can be often encountered not only in the living space but also in the industrial site. Studies on physiological and psychological effects of long-term noise exposure to human being have commanded the public interest on noise issues. Since environmental noises such as traffic noise and construction noise is mainly flowed through the open window, it is necessary to develop the active noise control system to reduce it inside the building. Although control speakers and microphones for the noise signal measurement in the control region are essential for the conventional active noise control methods, it is impossible to implement them in the control region in the building environment because the control region is the living quarter and they may hinder activities of the residents. Therefore, we proposed the active window system to reduce the exterior noise flowed through the open window with microphones installed outside the window and control speakers installed at the frame of the window. To confirm the performance of the proposed active window, we carried out the simulation and experiment using active window system with 8 control speakers. Simulation results showed the noticeable noise reduction effect inside the control region within the frequency range without the spatial aliasing. Experimental result showed that the total acoustic potential energy inside the room of the scale model is reduced to about 10dB within the interest of frequency range.

• Journal of Korea Multimedia Society
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• v.4 no.3
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• pp.230-237
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• 2001

Adaptive filters are widely used for acoustic echo canceler, adaptive equalizer and adaptive noise canceler. Coefficients of adaptive filters are updated by NLMS algorithm. However, Coefficients are misaligned by ambient noises when they are adapted by NLMS algorithm. In this Paper, a method determined the adaptation constant by low-pass filtered instantaneous gradient vector of LMS algorithm using orthognality principles of optimal filter is proposed. At initial states, instantaneous gradient vector, that is the cross-correlation of input signals and estimation error signals, has large value because input signals are remained in estimation error signals. When an adaptive filter is conversed, the cross-correlation will be close to zero. It isn's affected by ambient noises because ambient noises are uncorrelated with input signals. Determining adaptation constant with the cross-correlation, adaptive filters can be robust to ambient noises and the convergence rate doesn't slower As results of computer simulations, it is shown that the performance of proposed algorithm is betted than that of conventional algorithms.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.1
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• pp.18-25
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• 2018

In this paper, we present a low-power implementation of the multi-channel hearing aid system using a general-purpose DSP chip. The system includes an acoustic amplification algorithm based on Wide Dynamic Range Compression (WDRC), an adaptive howling canceller, and a single-channel noise reduction algorithm. To achieve a low-power implementation, each algorithm is re-constructed in forms of integer program, and the integer program is converted to the assembly program using BelaSigna(R) 250 instructions. Through experiments using the implementation system, the performance of each processing algorithm was confirmed in real-time. Also, the clock of the implementation system was measured, and it was confirmed that the entire signal processing blocks can be performed in real time at about 7.02MHz system clock.