• The Journal of the Acoustical Society of Korea
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• v.41 no.1
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• pp.16-29
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• 2022

This paper presents a Doppler shift estimation method and signal processing schemes for Direct Sequence Spread Spectrum (DSSS) transmission to overcome the Doppler shift due to the moving of the underwater communication unit. The proposed method estimates a Doppler shift via 2 step procedures using the preamble with the two 64-length Frank sequences which has a good self-correlation characteristic and is insensitive to the Doppler shift. Furthermore, a packet of DSSS underwater mobile communication and a RAKE receiver are designed using the proposed Doppler shift estimation method. Due to the modulation scheme of the designed DSSS underwater mobile communication using Differential-Quadrature Phase Shift Keying (DQPSK) for the data symbol transmission, the RAKE receiver dose not need a phase tracking and easily makes coherent signals among the combining RAKE branches. The designed RAKE receiving scheme including the proposed Doppler shift estimation method successfully decides information data using the DSSS signal transmitted from the pseudo-moving transmitter with velocity upto about 17.5 m/s.

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.131-142
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• 2022

In order to examine the possibility of horizontal simulations of acoustic waves on the environments of big water depth variations, this study introduces a 3-dimensional model based on the pababolic equation. The model gives approximated solutions by separating the cross- and non cross-terms in the equation. Assuming artificial bathymetry (25 km × 4 km) with a source frequency 75 Hz, the simulations give clear horizontal refractions on the transmission loss distributions. The degree of refractions shows non-linear increase along the propagating range and proportional increase with water depth along the cross range. Another simulations with the real bathymetry (25 km × 8 km) also give clear horizontal refractions. The horizontal distributions present little difference with the depth resolution variations of the same data source because the model gives interpolations over the depth data before simulations. Meanwhile, the horizontal distributions show big difference with those of different data sources.

• The Journal of the Acoustical Society of Korea
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• v.25 no.2
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• pp.61-71
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• 2006

An MP3 encoder chip has been designed and fabricated with the hardware and software co-design concepts. In the aspect of the software. the calculation cycles of the distortion control loop. which requires most of the calculation cycles in MP3 encoding procedure. have been reduced to $67\%$ of the original algorithm through the 'scale factor Pre-calculation'. By using a floating Point 32 bit DSP core and designing the FFT block with the hardware. we can get the additional reduction of the calculation cycles in addition to the software optimization. The designed chip has been verified using HW emulation and fabricated via 0.25um CMOS technology The fabricated chip has the size of $6.2{\time}6.2mm^2$ and operates normally on the test board in the qualitative and quantitative aspect.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.102-108
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• 2007

A high-Performance scalable speech codec generally requires a very low-rate first layer and a fine granule second layer, and this codec can be implemented with the harmonic codec and the MDCT-based transform codec for each layer. In this structure, however. each codec requires independent frequency transform and the time delay of each codec is accumulated. resulting in long time delay for the overall codec. In this paper, new MDCT structure in the second layer is Proposed. where MDCT is forced to share the look-ahead region of the first layer in order to prevent the time delay accumulation and the resulting functional error of MDCT is analyzed and removed after IMDCT The Proposed delayless MDCT requires no additional bits and Provides the equivalent coding performance with the reduced time delay, yielding a meaningful enhancement of the overall codec.

• The Journal of the Acoustical Society of Korea
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• v.28 no.5
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• pp.441-446
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• 2009

This paper proposes an automatic indexing algorithm of golf video using audio information. In the proposed algorithm, the input audio stream is demultiplexed into the stream of video and audio. By means of Adaboost-cascade classifier, the continuous audio stream is classified into announcer's speech segment recorded in studio, music segment accompanied with players' names on TV screen, reaction segment of audience according to the play, reporter's speech segment with field background, filed noise segment like wind or waves. And golf swing sound including drive shot, iron shot, and putting shot is detected by the method of impulse onset detection and modulation spectrum verification. The detected swing and applause are used effectively to index action or highlight unit. Compared with video based semantic analysis, main advantage of the proposed system is its small computation requirement so that it facilitates to apply the technology to embedded consumer electronic devices for fast browsing.

• The Journal of the Acoustical Society of Korea
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• v.27 no.4
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• pp.178-182
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• 2008

In measurements of the vibration of structures underwater with a laser Doppler vibrometer, the surface vibration is measured by means of detecting the phase change of the laser beam due to the structural vibration. The laser beam passes through the sound field radiated from the vibrating structures underwater. It experiences an additional phase change due to the change in refractive index in the radiated sound field. This phase change due to the sound field may cause the error in surface vibration measurements. In this paper, this phase change due to the radiated sound filed has been analyzed. The numerical simulation has been peformed to evaluate the phase change in sound field radiated from an infinite cylindrical structure vibrating underwater.

• The Journal of the Acoustical Society of Korea
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• v.27 no.6
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• pp.309-318
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• 2008

In this paper, we discuss a method for efficiently coding the high-band signal in the split-band coding approach where an input signal is divided into two bands and then each band may be encoded separately. Generally, and especially through the research on the artificial bandwidth extension (BWE), it is well known that there is a correlation between the two bands to some degree. Therefore, some coding gain could be achieved by utilizing the correlation. In the BWE-prediction-based coding approach, using a simple linear BWE function may not yield optimal results because the correlation has a non-linear characteristic. In this paper, we investigate the new coding scheme more in details. A few representative BWE functions including linear and non-linear ones are investigated and compared to find a suitable one for the coding purpose. In addition, it is also discussed whether there are some additional gains in combining the BWE coder with the predictive vector quantizer which exploits the temporal correlation.

• The Journal of the Acoustical Society of Korea
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• v.43 no.2
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• pp.253-259
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• 2024

This paper proposes a new speech enhancement model based on a complex nested U-Net with a dual-branch decoder. The proposed model consists of a complex nested U-Net to simultaneously estimate the magnitude and phase components of the speech signal, and the decoder has a dual-branch decoder structure that performs spectral mapping and time-frequency masking in each branch. At this time, compared to the single-branch decoder structure, the dual-branch decoder structure allows noise to be effectively removed while minimizing the loss of speech information. The experiment was conducted on the VoiceBank + DEMAND database, commonly used for speech enhancement model training, and was evaluated through various objective evaluation metrics. As a result of the experiment, the complex nested U-Net-based speech enhancement model using a dual-branch decoder increased the Perceptual Evaluation of Speech Quality (PESQ) score by about 0.13 compared to the baseline, and showed a higher objective evaluation score than recently proposed speech enhancement models.

• The Journal of the Acoustical Society of Korea
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• v.43 no.1
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• pp.138-144
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• 2024

This paper presents a bistatic to monostatic conversion technique to analyze the bistatic target strength of submarines. The technique involves determining the transmission path length of acoustic waves, which are emitted from a source, scattered off an underwater target, and eventually received by a receiver. By generating a corresponding virtual scattering surface, this method effectively transforms the target strength analysis problem from bistatic to monostatic. The converted monostatic target strength problem can be assessed using a well-established monostatic numerical methods. The bistatic target strength analysis for Benchmark Target Strength Simulation (BeTTSi), a widely used target strength model were performed. The results were compared with those calculated by boundary element methods and Kirchhoff approximation, and confirmed the validity and the practical applicability of the proposed analysis technique for evaluating submarine target strength.

• The Journal of the Acoustical Society of Korea
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• v.43 no.1
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• pp.19-28
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• 2024

This paper proposes a method to improve the performance of ship identification through lofargram analysis of ship noise by applying the Hough Transform to a Convolutional Neural Network (CNN) model. When processing the signals received by a passive sonar, the time-frequency domain representation known as lofargram is generated. The machinery noise radiated by ships appears as tonal signals on the lofargram, and the class of the ship can be specified by analyzing it. However, analyzing lofargram is a specialized and time-consuming task performed by well-trained analysts. Additionally, the analysis for target identification is very challenging because the lofargram also displays various background noises due to the characteristics of the underwater environment. To address this issue, the Hough Transform is applied to the lofargram to add lines, thereby emphasizing the tonal signals. As a result of identification using CNN models on both the original lofargrams and the lofargrams with Hough transform, it is shown that the application of the Hough transform improves lofargram identification performance, as indicated by increased accuracy and macro F1 scores for three different CNN models.