• The Journal of the Acoustical Society of Korea
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• v.38 no.1
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• pp.114-119
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• 2019

The transmitted acoustic signals are severely influenced by multiply reflected signals from boundaries, such as sea surface and bottom in the shallow water. Very large reflection signals from boundaries cause inter-symbol interference so that the performance of the underwater acoustic communication is degraded. Usually, the waveform shaping filters are used to prevent the reflected signals under this kind of acoustic channel. Especially, the raised cosine filter is widely used, which can also be used to restrict the bandwidth of the transmitted signal. In this study, we evaluate the raised cosine filter for image data transmission in the shallow water, and propose a new modified raised cosine filter. The QPSK (Quadrature Phase Shift Keying) system is used for the underwater acoustic communication simulations with different distances and symbol rates. As a result, the bit error rate was reduced from the minimum 1.0 % to the maximum 32 %.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.438-445
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• 2002

In this paper, the necessity of the filter coefficients compensation for the lattice transversal joint (LTJ) adaptive filter was explained in general and with ease by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed using the property that speech signal is stationary during a short time period and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.743-749
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• 2007

Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

• The Journal of the Acoustical Society of Korea
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• v.34 no.3
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• pp.177-183
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• 2015

This paper proposes a robust nonlinear acoustic echo canceller (NAEC) which is effective for modeling the nonlinearity of a speaker module and the long acoustic echo path within a speech communication environment. The proposed NAEC utilizes a sigmoid pre-processor for modeling the speaker nonlinearity and a partitioned block frequnecy-domain adaptive filter for identifying the acoustic echo path with small delay. Simulation results confirmed that the proposed algorithm achieves excellent performance with much lower computational complexity than the previous NAEC.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.5
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• pp.961-968
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• 2007

Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.28-34
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• 2001

In stereo acoustic echo canceller scheme, coefficients of adaptive filter converge very slowly or misconverge to real acoustic echo path in receiving room. This is due to cross-correlation in stereo signals. In this paper, a new preprocess algorithm is proposed to improve the performance of stereo AEC(acoustic echo canceller) without computational burden. The proposed algorithm reduces cross-correlation using Gram-Schmidt orthogonality principles and nonlinear filtering. Computer simulations demonstrate that this algorithm performs well compared to conventional ones. When the acoustic path of transmitting room is changed, stereo AEC using proposed algorithm is well performed.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.5-13
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• 2001

It is well known that there are two experimental errors on acoustic decay measurements. ,One is due to the influence of the band pass filter the other one is that of an averaging device. In this paper the influence of the filter is investigated in detail. To minimize the influence of the filter, the product of the filter bandwidth B (3dB bandwidth) and the reverberation time T/sub 60/ of the room under test should be at least 16. Moreover, if the initial part of an acoustic decay curve is important, the strong requirement, i. e. BT/sub 60/〉64, must be satisfied. In this paper, the wavelet filter bank instead of the band pass filter bank is applied to obtain an acoustic decay curve. As a result, the influence of filter is reduced and then the value of BT/sub 60/ required for obtaining an acceptable decay curve becomes at least 4. The strong requirement for the initial part of a decay curve is also replaced by the BT/sub 60/〉16 instead of BT/sub 60/〉64.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.264-273
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• 2002

Adaptive filters cannot exactly estimate the echo path of the receiving room because of the cross-correlation of stereo signals. In this paper, a new pre-processing method reducing the cross-correlation without degradation of stereophony is proposed to enhance the performance of stereophonic acoustic echo canceller. To reduce the cross-correlation, absolutes of two orthogonal signals derived from each channel signals are added to original channel signals. Assuming that the power of each channel signal is larger than that of the cross-correlation, the computation of pre-processing can be reduced. As results of simulations, it is shown that the performance of stereo acoustic echo canceller with the proposed pre-processing method is better than that of conventional ones.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.153-161
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• 2003

In this paper, we propose a wavelet-based detection method to identify efficiently the time-delay or multipath channel of ocean acoustic signals due to complex ocean medium and boundary layers. Our proposed method employs wavelet packet transform to analyze the received broadband acoustic signals and applies the matched filter to determine the time region of interest. Also, we present numerical testing that results on both the simulated and real data revealed the efficiency of this method in time-delay estimation and moreover its capability in estimating the time-delay of individual path in multipath channel, in which the arrival patterns are too close to be separated by the matched filter method.