• The Journal of the Acoustical Society of Korea
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• v.38 no.1
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• pp.82-95
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• 2019

In this study, we analyzed the beam time series of ocean reverberation which was conducted in the eastsouthern region of East Sea, Korea during the August, 2015. The reverberation data was gathered by moving research vessel towing LFM (Linear Frequency Modulation) source and triplet receiver array. After signal processing, we analyzed the variation of ocean reverberation level according to the seafloor bathymetry, source/receiver depth and sound speed profile. In addition, we used the normalized data by using cell averaging algorithm and identified the statistical characteristics of seafloor scatterer by using moment estimation method and estimated shape parameter. Also, we analyzed the coincidence of data with Rayleigh and K-distribution probability by Kolmogorov-Smirnov test. The results show that there is range dependency of reverberation according to the bathymetry and also that the time delay and the intensity level change depend on the depths of source and receiver. In addition, we observed that statistical characteristics of similar Rayleigh probability distribution in the ocean reverberation.

• The Journal of the Acoustical Society of Korea
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• v.41 no.3
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• pp.255-267
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• 2022

In this paper, noise characteristics of bubbles created by an orifice-type bubble generator are studied. In order to understand the overall bubble noise characteristics, the bubble noise spectra proposed by Strasberg and Blake, respectively, are examined, and an air injection experiment was performed in the large cavitation tunnel of KRISO to measure the bubble noise. The experiments were performed under a quiescent condition and flow conditions using 5 types of air bubble generator. From the measurement results, the characteristics of the bubble noise spectrum according to the experimental conditions are observed, and the effect of each parameter on bubble noise is analyzed by regression analysis. Finally, empirical models based on the regression analysis for bubble noise are presented, and it is confirmed that the estimated bubble noise is in good agreement with the measured results.

• The Journal of the Acoustical Society of Korea
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• v.42 no.4
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• pp.357-363
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• 2023

In underwater signal processing, separating individual signals from mixed signals has long been a challenge due to low signal quality. The common method using Short-time Fourier transform for spectrogram analysis has faced criticism for its complex parameter optimization and loss of phase data. We propose a Triple-path Recurrent Neural Network, based on the Dual-path Recurrent Neural Network's success in long time series signal processing, to handle three-dimensional tensors from multi-channel sensor input signals. By dividing input signals into short chunks and creating a 3D tensor, the method accounts for relationships within and between chunks and channels, enabling local and global feature learning. The proposed technique demonstrates improved Root Mean Square Error and Scale Invariant Signal to Noise Ratio compared to the existing method.

• The Journal of the Acoustical Society of Korea
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• v.25 no.5
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• pp.207-212
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• 2006

The purpose of this study was to compare voice parameters in auditory conditions (normal/noise/music) between a teacher group and a control group. Results of statistical analysis showed that the teacher group had higher jitter (%) and shimmer (%) values than the control group. It indicated that the teacher group had larger variations in pitch and dynamic of their voice. In the teacher group, the voice under noisy condition showed a higher value of fundamental frequency than that under normal condition. though its fundamental frequency did not show any significant difference between the noisy condition and the musical condition. In the control group, however, although the voice under noisy condition also showed a higher value of fundamental frequency than that under normal condition, its fundamental frequency was significantly different between the noisy condition and the musical condition.

• The Journal of the Acoustical Society of Korea
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• v.26 no.8
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• pp.375-380
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• 2007

A reverberation chamber should be designed and constructed so as to satisfy its purposes and available space. However, it is somewhat difficult to meet the intended design requirements due to various errors from construction process. So, the post-construction measurement of its volume and surface areas is very essential to check the actual volume and volume uncertainty of a reverberation chamber These values should be carefully calculated and accurately estimated since they are used not only to evaluate the acoustic characteristics of building materials but also to calculate uncertainties for other acoustic characteristics. In this work, the method for the calculation and uncertainty estimation of the volume of a reverberation chamber is presented. To this end, the coordinates of all corners was measured with Total Station after construction. The results showed that the calculated volume of the measured reverberation chamber differs by 5 % from the design specification. The expanded volume uncertainty was also estimated to be about 2 % of the total calculated volume.

• The Journal of the Acoustical Society of Korea
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• v.26 no.8
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• pp.415-425
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• 2007

This paper proposes a method of underwater source localization using the wideband interference patterns matching. By matching two interference patterns in the spectrogram, it is estimated a ratio of the range from source to sensor5, and then this ratio is applied to the Apollonius circle. The Apollonius circle is defined as the locus of all points whose distances from two fixed points are in a constant value so that it is possible to represent the locus of potential source location. The Apollonius circle alone, however still keeps the ambiguity against the correct source location. Therefore another equation is necessary to estimate the unique locus of the source location. By estimating time differences of signal arrivals between source and sensors, the hyperbola equation is used to get the cross point of the two equations, where the point being assumed to be the source position. Simulations are performed to get performances of the proposed algorithm. Also, comparisons with real sea experiment data are made to prove applicability of the algorithm in real environment. The results show that the proposed algorithm successfully estimates the source position within an error bound of 10%.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.290-297
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• 2009

This paper presents an improved adaptive multi-rate wideband (AMR-WB) algorithm for the efficient Text-To-Speech (TTS) database compression. The proposed algorithm includes unnecessary common bit-stream (CBS) removal and parameter delta coding combined with speaker-dependent huffman coding to reduce the required bit-rate without any quality degradation. We also propose lossy coding schemes to produce the maximum bit-rate reduction with negligible quality degradation. The proposed lossless algorithm including CBS removal can reduce bit-rate by 12.40% without quality degradation compared with the 12.65 kbps AMR-WB mode. The proposed lossy algorithm can reduce bit-rate by 20.00% with 0.12 PESQ degradation.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.599-609
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• 2009

An overview of the recently finalized multi-channel audio coding standard MPEG Surround is provided. This audio coding standard downmixes multi-channel signals to mono or stereo signals and, simultaneously, extracts spatial parameters for its encoding process. In its decoding process, it reconstructs multi-channel signals based on the downmix signals and spatial parameters. Since the downmix signals are coded in conventional audio coding format such as AAC and MP3 and the spatial parameters require a small amount of information MPEG Surround guarantees high sound quality multi-channel audio at low bit rates. Besides, it is backward-compatible to conventional audio coding techniques because the downmix signals can be played on portable audio devices ignoring the spatial parameter information. In this paper, Part 1 presents an overview of the basic structure of MPEG Surround and Part 2 describes various modes and tools including the binaural mode which supports the virtual 5.1-channel playback via headphones or earphones. The listening test results by various companies and organizations are also presented.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.815-821
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• 2009

Information processing in variable and noisy environments is usually accomplished by means of adaptive filters. Among various adaptive algorithms, Least Mean Square (LMS) has become the most popular for its robustness, good tracking capabilities and simplicity, both in terms of computational load and easiness of implementation. In practical application of the LMS algorithm, the most important key parameter is the Step Size. As is well known, if the Step Size is large, the convergence rate of the algorithm will be rapid, but the steady state mean square error (MSE) will increase. On the other hand, if the Step Size is small, the steady state MSE will be small, but the convergence rate will be slow. Many researches have been proposed to alleviate this drawback by using a variable Step Size. In this paper, a new variable Step Size LMS(VSSLMS) called Categorized VSSLMS (CVSSLMS) is proposed. CVSSLMS updates the Step Size by categorizing the current status of the gradient, hence significantly improves the convergence rate. The performance of the proposed algorithm was verified from the view point of convergence rate, Excessive Mean Square Error(EMSE), and complexity through experiments.

• The Journal of the Acoustical Society of Korea
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• v.25 no.3
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• pp.113-120
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• 2006

This paper is concerned with a equalization technique for Orthogonal Frequency Division Multiplexing (OFDM) systems based on a block type pilot arrangement over slow fading channels. The bit rates obtained in underwater channels are relatively modest compared to some other communication channels such as cellular phones or indoor wireless systems. Consequently. the Doppler effect is the important parameter in tracking a channel. In case of a coherent demodulation scheme, the residual mean phase errors due to Doppler frequency may be fatal for the performance of the system. The equalizer could not solely handle mean Doppler shift. To account for the common Doppler effect a phase error tracking loop is used with the frequency equalizer. so that the rotation errors are avoided. Furthermore. simulations show that we can reduce the computational load of the tracking loop with negligible effect on performance.