• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.184-191
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• 2022

Underwater acoustic wave in shallow water is propagated through multipath that has a large delay spread causing Inter-Symbol Interference (ISI) and these characteristics deteriorate the performance in the communication system. In order to analyze the communication performance and investigate the correlation with multipath delay spread in a reverberant environment, an underwater acoustic communication experiment using Binary Phase-Shift Keying (BPSK) signals with symbol rates from 100 sym/s to 8000 sym/s was conducted in a 5 × 5 × 5 m³ water tank. The acoustic channels in a well-controlled tank environment had the characteristics of dense multipath delay spread due to multiple reflections from the interfaces and walls within the tank and showed the maximum excess delay of 40 ms or less, and the Root Mean Squared (RMS) delay spread of 8 ms or less. In this paper, the performances of Bit Error Rate (BER) and output Signal-to-Noise Ratio (SNR) were analyzed using four types of communication demodulation techniques. And the parameter, Symbol interval to Delay spread Ratio in reverberant environment (SDR_rev), which is the ratio of symbol interval to RMS delay spread in the reverberant environment is defined. Finally, the SDR_rev was compared to the BER and the output SNR. The results present the reference symbol rate in which high communication performance can be guaranteed.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.548-556
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• 2009

This paper proposes an efficient speaker array system to spotlight a sound into a target position in a reverberant environment. The proposed method introduces a criterion of maximizing the energy of system response from each channel to the target position. Simulation results with a sixteen channel speaker array system prove that the performance of the proposed method improves that of the conventional method that uses a broadband beamformer in a reverberant environment.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.8
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• pp.823-831
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• 2008

Since it needs the light computational load and small database, sound source localization method using time delay of arrival(TDOA method) is applied at many research fields such as a robot auditory system, teleconferencing and so on. Researches for time delay estimation, which is the most important thing of TDOA method, had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Silverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time. Also, we define the performance index(PI) to show a similar tendency to R/D ratio, and propose the method to improve the performance of time delay estimation with PI.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.867-872
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• 2002

A high intensity acoustic test facility is constructed at Korea Aerospace Research Institute (KARI) by 2003. The reverberant chamber of the facility has a volume of 1,228 cubic meters and shall provide an acoustic environment of 152 dB over the frequency range of 25 Hz to 10,000 Hz. The facility consists of a large scaled reverberant chamber, acoustic power generation systems, gases nitrogen supply systems, and acoustic control systems. This paper describes how the basic parameters of a chamber and power generation systems are controlled to meet the requirement of the test. The volume of a reverberant chamber is controlled by the size of test objects and the reverberant characteristics of a chamber. The capacity of acoustic power generation systems is determined by the energy absorption of a chamber and the efficiency of acoustic modulators. Simple math is employed to calculate the required power of acoustic modulators. Moreover, the paper explains how the distribution of sound pressure level at low frequency is checked by analytical and numerical methods.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.05a
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• pp.96-102
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• 2001

High intensity vibro-acoustic testing is the appropriate method for flight qualification testing of space flight vehicle which must ensure the acoustic environment of launch. Growing demand for satellites and launch vehicles in korea has resulted in a recent increase in the demand for high intensity vibro-acoustic test facility. The test facility is designed to provide an acoustic environment of 152 ㏈( re 20 ${\mu}$Pa) overall sound pressure level over the band width of 30 Hz to 10,000 Hz in the reverberant chamber. The reverberant chamber has a volume of 1,000 ㎥ with interior dimensions of 8.7m${\times}$l0m${\times}$12m, which can accommodate not only satellites but also launch vehicle payload fairing. Korea Aerospace Research Institute and Korean industries have been carrying out the development of the reverberant chamber and auxiliary devices, such as automatic control system, monitoring/safety device, and jet nozzle, etc. This paper presents the detailed description of High Intensity Acoustic Chamber of KARI, which will be the first and unique testing facility in Korea.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.38 no.3
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• pp.258-263
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• 2010

Sound localization algorithm usually adapts three step process: sampling sound signals, estimating time difference of arrival between microphones, estimate location of sound source. To apply this process in indoor environment, sound localization algorithm must be strong enough against reverberant and noisy condition. Additionally, calculation efficiency must be considered in implementing real-time sound localization system. To implement real-time robust sound localization system we adapt four low cost condenser microphones which reduce the cost and total calculation load. And to get TDOA(Time Differences of Arrival) of microphones we adapt GCC-PHAT(Generalized Cross Correlation-Phase Transform) which is robust algorithm to the reverberant and noise environment. The position of sound source was calculated by using iterative least square algorithm which produce highly accurate position data.

• The Journal of the Acoustical Society of Korea
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• v.31 no.8
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• pp.487-506
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• 2012

This paper proposes a speech enhancement algorithm to improve the speech intelligibility by suppressing both reverberation and background noise. The algorithm adopts a non-causal single-channel minimum variance distortionless response (MVDR) filter to exploit an additional information that is included in the noisy-reverberant signals in subsequent frames. The noisy-reverberant signals are decomposed into the parts of the desired signal and the interference that is not correlated to the desired signal. Then, the filter equation is derived based on the MVDR criterion to minimize the residual interference without bringing speech distortion. The estimation of the correlation parameter, which plays an important role to determine the overall performance of the system, is mathematically derived based on the general statistical reverberation model. Furthermore, the practical implementation methods to estimate sub-parameters required to estimate the correlation parameter are developed. The efficiency of the proposed enhancement algorithm is verified by performance evaluation. From the results, the proposed algorithm achieves significant performance improvement in all studied conditions and shows the superiority especially for the severely noisy and strongly reverberant environment.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.909-915
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• 2008

Researches for time delay estimation had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Siverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1845-1851
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• 2007

Background noise and room reverberation are two causes of degradation in speech in listening situations. Many algorithms developed to enhance reverberant speech. In this paper we propose a dereverberation method for enhancement of speech using modified the linear prediction(LP) residual in reverberant room condition. The proposed dereberberation method based on the fact that the signification excitation of the vocal tract system takes place at the instant of glottal closure in voiced speech. Our method used delay information form each sensor, and we need reverberant signals from 3 sensors. We obtain a new LP residual signal using modified IP residual combination which derived form weighting of the LP residual and the Hilbert transform of LP residual. The nature of the coherently added Hilbert envelop has several large amplitude spikes because of the effects of noise and reverberation. This residual of the clean speech is used to excite the time-varying all-pole filter to obtain the enhanced speech. We achieved simulation of proposed algorithm for performance analysis in reverberation environment. The proposed algorithm improves substantially the quality of reverberant speech.