• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1C
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• pp.96-102
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• 2010

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.75-81
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• 2001

In this paper we analyzed the localization error of near-field detection algorithm in the sea. The near-field detection algorithms using triangulation and wavefront curvature basically assume a signal in two dimension of bearing and range. But the assumption causes localization error because there is three dimension of bearing, range, and depth in the sea. Even through three dimensional effect is considered, the localization error is occurred if multipath propagation in the sea is ignored. To analyze the localization error in the sea, we simulate the near-field localization using acoustic propagation model and focused beamforming considering wavefront curvature. The simulation results indicate that localization error always occurs in the sea and the error varied with sound velocity profile, water depth, bottom slope, source range, etc.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3
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• pp.27-34
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• 1999

This paper proposes an algorithm to design the non-linear array so as to form efficiently the one-shot beam with relatively less sensors for acoustic measurement. In this algorithm, according to the spatial sampling theory the part for high frequency(HF) band has equispaced sensor array and the sensor distances below the HF band are decided as a function of number of HF sensors. As the results of the simulations, the mean and variances of directivity index(DI) of non-linear array which has less sensors are similar to those of linear array. and the DI variation for beam steering angle is very small. And the beam width at -2dB point is 6.8°. Thus it is confirmed that the design algorithm for non-linear array which is proposed to have less sensors can be efficiently used in acoustic measurement.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2E
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• pp.78-86
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• 2003

We analyzed the measured acoustic field to explore the characteristics of a horizontal coherence in shallow water. Signal spatial coherence data were obtained in the continental shelf off the east coast of Korea using a horizontal line array. The array was deployed on the bottom of 130 m water depth and a sound source was towed at 26 m depth in the source-receiver ranges of 1-13 ㎞. The source transmitted 200 ㎐ pure tone. Topography and temperature profiles along the source track were measured to investigate the relationship between the horizontal coherence and environment variations. The beam bearing disturbance and array signal gain degradation is examined as parameters of horizontal coherence. The results show that the bearing disturbance is about ± 8° and seems to be affected by temporal variations of temperature caused by internal waves. The array signal gains show degradation more than 5㏈ by the temporal and spatial variations of temperature and by the down-sloped topography.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.89-90
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• 2008

We present an implementation and it's real-sea test of an underwater acoustic communication system, which allows the system to reduce complexity and increase robustness in time variant underwater environments. For easy adaptation to complicated and time-varying environments of the ocean, all-digital transmitter and receiver systems were implemented. For frame synchronization the CAZAC sequence was used, and QPSK modulation/demodulation method with carrier frequency of 25kHz and a bandwidth of 5kHz were applied to generate 10kbps transmission rate including overhead. To improve transmission quality, we used several techniques and algorithms such as adaptive beamforming, adaptive equalizer, and convolution coding/Viterbi decoding. For the verification of the system performance, measurement of BER has been done in a very shallow water with depth of 20m at JangMok, Geoje. During the experiment, image data were successfully transmitted up to about 9.6km.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.606-613
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• 2002

In this paper we applied a transmit beamforming technique to the underwater acoustic communication system for high rate data transmission. A prototype transmit system was designed and implemented with the general purpose DSP processor and multiple digital-to-analog converters. The performances of the implemented system were evaluated by the experiment in water tank. In order to simplify the procedure the channel coding and equalizer were omitted. And the simplest OOK (On-Off Keying) technique in digital communication methods was applied. The experimental result shows that the transmission data rate is higher about 3 times in the case of 5 hydrophone transmitting may than 1 hydrophone transmitter at bit error rate 10/sup -2/. We verified that the maximum data rate was 400 bps for speech signal transmission in water tank.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.86-92
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• 2001

Various array processing techniques to identify the noise source position or bearing have been developed. Typical array processing techniques which are based on time delay between received signals at two sensors, are classified as conventional beamforming, correlation function and NAH (Near-Field Acoustic Holography) techniques which have their own characteristics with respect to application field and signal processing method. In this study, correlation function technique which could be applied for broadband noise source detection, is adopted and the effective detection technique is proposed considering the effects of source bandwidth and measurement noise correlation of noise sources. The validity of the Proposed technique is evaluated using the 3-dimensional nonlinear any which does not give 3-dimensional Position or bearing ambiguity

• The Journal of the Acoustical Society of Korea
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• v.23 no.7
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• pp.514-520
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• 2004

There are many techniques for underwater source localization. These are the methods based on TDOA (Time Difference Of Arrival) estimation. beamforming techniques and high resolution techniques, etc. In this Paper we estimate the underwater source position using MCPSP (Modified Cross Power Spectrum Phase) function that is calculated on frequency domain using sensors of small number. However, the performances of the localizing method based on MCPSP function drops greatly in the case of CW (Continuous Wave) signal . In this Paper we proposed the TDOA estimation method for pulsed CW signal. In the Proposed method we composed of new segment including a edge of ping. This segment was computed by short-time energy detection. With theoretical representation the performances of the proposed method were analyzed under various environment.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.754-757
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• 2007

In this paper, we present an implementation and it's real-sea test of an underwater acoustic data communication system, which allows the system to reduce complexity and increase robustness in time variant underwater environments. For easy adaptation to complicated and time-varying environments of the ocean, all-digital transmitter and receiver systems were implemented. For frame synchronization the CAZAC sequence was used, and QPSK modulation/ demodulation method with carrier frequency of 25kHz and a bandwidth of 5kHz were applied to generate 10kbps transmission rate including overhead. To improve transmission quality, we used several techniques and algorithms such as adaptive beamforming, adaptive equalizer, and convolution coding/Viterbi decoding. for the verification of the system performance, measurement of BER has been done in a very shallow water with depth of 8m at JangMok, Geoje. During the experiment, image data were successfully transmitted up to about 7.4km.

• Journal of KSNVE
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• v.6 no.6
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• pp.835-842
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• 1996

To identify the locations and strengths of acoustic sources, one may use a microphone line array. Apparent advantage of the source identification method utilizing a line array is that it requires less measurement points than intensity method and holography. This method is based on the information of magnitude and phase difference between pressure signals at each microphone. Since those differences are dependent on the source model, we have to assume them such as plane, monopole, etc. In this paper the conventional source identification methods such as beamforming method and MUSIC method are briefly reviewed by modeling a source as plane and spherical wave, then a modified method is introduced. This can be applied to sound field which may by either coherent or incoherent. Typical simulations and experiment are performed to confirm this identification method.