• The Journal of the Acoustical Society of Korea
• /
• v.33 no.1
• /
• pp.48-53
• /
• 2014

The underwater acoustic communication (UAC) is characterized by doubly spread channel. It is included in the time-variant doppler shift and delay-time spreads due to multiple paths. To compensate such distorted signals, various techniques including time-reversal processing, spatial diversity, phase estimator, and equalizer are being applied. In this paper, a spatial filter based on the beamforming is proposed as a method to mitigate such inter-symbol interferences that are generated in time-varying multipath channels. The proposed technique realizes coherent communications by steering the direction of the desired signals and improves the performance of UAC by increasing the signal-to-interference plus noise ratio using the array gain.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.39 no.5
• /
• pp.556-563
• /
• 2002

In this paper, a novel method for designing an M-channel, causal, stable IIR cosine-modulated filter bank (CMFB) with near PR property is proposed. The IIR prototype filter is designed with a simple constraint using lattice stucture with 1st order allpass filter components. The IIR prototype filter which is designed by the proposed method has higher stopband attenuation and sharper roll-off characteristic than the one which is designed by the previously proposed method with similar complexity. The proposed M-channel IIR CMFB which is designed from this IIR prototype filter is applied to subband acoustic echo canceller (AEC). We obtained about 15dB higher ERLE using this subband AEC than when M-channel FIR subband AEC with similar complexity.

• The Journal of the Acoustical Society of Korea
• /
• v.11 no.2
• /
• pp.59-67
• /
• 1992

Recently, the problem which actively control the unwanted noise propagated from the technical structure by the generated secondary sound has become considerable topic from the environmental preservation point of view. In most of these studies, active noise control deals with a plane wave propagation at low frequency using adaptive filtering techniques. On the other hand, in real acoustic systems are mostly short due to the limitation of geometric configuration. In this case, the acoustic properties such as reflections and resonances inside the acoustic system should be considered. In this paper, the acoustic modeling method for short length duct was introduced using the transfer matrix method, and the active noise control problem was investigated with \implementation of FIR filter for the transfer function of control system derived from this modeling method. The identification methods for the acoustic model of actual control system was proposed by numerical computation technique based on the estimation of optimal FIR filter coefficients. The acceptable attenuation on the real acoustic system and stability of the controller are predicted in this computational simulation.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.4C
• /
• pp.343-349
• /
• 2010

In this paper, we study real time AEC (acoustic echo canceller) based on NLMS adaptive filter. Proposed method improves conversation quality by enhancing the performance of AEC during double talk section and reduces the power consumption by controling the adaption operation of NLMS adaptive filter. Proposed method examines the convergence of the NLMS adaptive filter, stores the estimated echo path and chooses operation of NLMS adaptive filter. Furthermore if double talk is detected, the proposed AEC utilizes the stored echo path optionally considering missed double talk time. When the proposed AEC is used, the performance of the AEC is enhanced although the simple double talk detector is used and the power consumption of the AEC is reduced.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.1
• /
• pp.13-20
• /
• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.7
• /
• pp.573-578
• /
• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.5
• /
• pp.353-360
• /
• 2017

The single sensor based acoustic reflection control system separates the incident and reflected signals from the single sensor output, and reduces the reflected signal by generating an out-of-phase signal from the incident signal component. In this paper, we propose an optimal filter design method for a single sensor based reflection control system. In the proposed method, it is shown that the optimum control filter design is possible by using the measured impulse responses of the reflection and control paths. The reflection control algorithm based on the proposed optimal filter achieves better performance than the conventional adaptive filter-based algorithm and effectively controls the reflection without the initial convergence time. We performed computer simulations using the signals obtained in a 1-dimensional acoustic duct environment, and from the simulation results, it was confirmed that the proposed optimal filter has robust performance even in noisy environment.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.1
• /
• pp.42-46
• /
• 2002

In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.

• Korean Journal of Optics and Photonics
• /
• v.11 no.3
• /
• pp.193-197
• /
• 2000

A polarized mode convertible, wavelength tunable optical filters with acoustic barriers and acousto-optic effect have been produced in LiNb03 substrate utilizing the Ti double diffusion technique. Polarization conversion in excess of 81 % and a spectral width of -200 kHz (-1.83 nm) were achieved at a wavelength of 1551.6 nm and RF frequencies of 173.07 kHz and 173.05 kHz for both transverse electric (TE) and transverse magnetic (lM) input polarizations, respectively. The electrical driving power was 10.97 mW and reduced to about 10% of one for an optical filter without an acoustic barrier. A linear tuning rate of 8.2 nmlMHz and sidelobe intensity of -4 dB was demonstrated. rated.

• Journal of the Korean Institute of Navigation
• /
• v.17 no.1
• /
• pp.99-105
• /
• 1993

Underwater acoustic positioning systems have been extensively used not only in surface position fixing but also in underwater position fixing. Recently, these systems have been applied in the field of installation and underwater inspection offshore platforms etc. But in these systems are included the fixing errors as results of a signal with noise and irregular motion of vessel by ocean waves. In this paper to improve the accuracy of the position fixing a Kalman filter is applied to the short baseline(SBL) acoustic positioning system. The optimal position obtained by the Kalman filter is compared with the raw position and it is confirmed that the former is more accurate than the latter.