• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.300-308
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• 2014

Photoacoustic imaging is a useful tool for the diagnosis of atherosclerosis because it is capable of providing anatomical and pathological information at the same time. A photoacoustic signal detector is a pivotal element to achieve high spatial resolution, so that it should have broadband spectrum with a high center frequency. Since a photoacoustic imaging probe is directly inserted into blood vessel to diagnose atherosclerosis, the total size of the photoacoustic signal detector should be less than 1 mm. The main purpose of this paper is to demonstrate that PVDF can be used as an active material for the photoacoustic signal detector with a high frequency and broadband characteristic. The photoacoustic signal detector developed in this study was a single element ultrasound transducer with an aperture of $0.5{\times}0.5mm$ and the total size of 1 mm. In the design stage, the natural focal depth was adjusted for an effective focal area to cover the region of interest, i.e., 1~5 mm in depth. This was because geometrical focusing could not be used due to the small aperture. Through a pulse-echo test, it was ascertained that the developed photoacoustic signal detector has the -6 dB bandwidth ranging between 40.1 and 112.8 MHz and the center frequency of 76.83 MHz.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.568-575
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• 2020

Ultrasound has been widely used in various industrial fields. One of challenging application areas is cooling microelectronics. Ultrasonic cooling systems can work with air, argon (Ar) and nitrogen (N₂) instead of conventional refrigerant such as freon gas, which can cause global warming. Furthermore, ultrasonic systems do not have moving parts, thus high durability can be obtained. So it is necessary to develop ultrasonic cooling systems due to environmental issues and durability points. In this paper, the design and fabrication processes are explained. When designing the system, a feasibility test was performed with a prototype cooler. Based on the result, finite element analysis with ANSYS software was performed. The predicted anti-resonance frequency for a piezoelectric actuator was 34.8 kHz, which was in good agreement with the experimental result of 34.6 kHz with 0.6% error. In addition, the predicted anti-resonance frequency for the ultrasonic waveguide was 39.4 kHz, which also agreed well with the experimental value of 39.8 kHz with 1.0% error. Based on these results, the developed ultrasonic waveguide might be applicable in microchip cooling.

• The Journal of the Acoustical Society of Korea
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• v.30 no.1
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• pp.1-8
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• 2011

This paper presents a method for improving the performance of ISF coefficients quantizer through compensating the defect of the split structure vector quantization using the ordering property of ISF coefficients. And design the ISF coefficients quantizer for wideband speech codec using proposed method. The wideband speech codec uses split structure vector quantizer which could not use the correlation between ISF coefficients fully to reduce complexity and the size of codebook. The proposed algorithm uses the ordering property of ISF coefficients to overcome the defect. Using the ordering property, the codebook redundancy could be figured out. The codebook redundancy is replaced by the adaptive-extended codebook to improve the performance of the quantizer through using the ordering property, ISF coefficient prediction and interpolation of existing codebook. As a result, the proposed algorithm shows that the adaptive-extended codebook algorithm could get about 2 bit gains in comparison with the existing split structure ISF quantizer of AMR-WB (G.722.2) in the points of spectral distortion.

• The Journal of the Acoustical Society of Korea
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• v.30 no.1
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• pp.9-16
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• 2011

When a hydrofoil ship plies at high speed, there exist possibilities of collision with ocean mammals dwelling near the surface. An active sonar located within the immersed hydrofoil structure that provides the lift for the vessel, can be used for early warning of their presence. The proper functioning of the active sonar system depends on its ability to reject noise and pick up the target signal. In this article, we measured the noise on a hydrofoil of an operating ship with two flush-mounted hydrophones. The measurements were conducted for the purpose of (1) identifying the effect of operating state of machinery likes engine, cooler and generator (2) observing the change of noise depending on the measuring position (3) observing the change of noise with increasing ship speed. To verify our experiment, experiments were performed three times and the measured results are compared with other investigations and they show similarity to each other. The results are analyzed with frequency domain in order to apply to operating active sonar detecting system and focus on high frequency band within sonar's operating frequency region. Through these experiments and analysis, it is expected that we can identify the generated noise around hydrofoil where active sonar is installed and these results lead us to design active sonar that could distinguish target signal from noise more effectively.

• The Journal of the Acoustical Society of Korea
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• v.19 no.2
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• pp.33-39
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• 2000

We have studied to obtain the frequency characteristics of the Surface Acoustic Wave(SAW) bandpass filter, having low shape factor, it's interdigital transducer(IDT) was formed on the 35° Y-cut X-propagation Quartz substrate and was evaporated by Aluminium. And then, we performed computer-simulation by a simulator. And, we can design that the apodization weighted type IDT as an input transducer of the filter and the withdrawal weighted type IDT as an output transducer of the filter from the results of our computer-simulation. Also, we have employed that the number of pairs of the input and output IDT are 2200 pairs and 1000pairs, respectively and used the Kaiser-Bessel window function in order to minimize the effect of ripple. And, while the width and the space of IDT's finger are 6㎛ m and 5.75㎛, respectively and we could obtain the resonable results when the IDT thickness was 6000Å in consideration of the ratio of SAW's wavelength, and IDT aperture is 2mm. Frequency response of the fabricated SAW bandpass filter has the property that the center frequency is about 70MHz, shape factor is less than 1.3, bandwidth at the 1.5dB is probably 1.3MHz, out-band attenuation is almost -45dB, insertion loss is 19dB and ripple in the width of bandpass is 1dB approximately. Therefore, these frequency characteristics of the fabricated SAW bandpass filter are agreed well with the designed values.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.717-726
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• 2003

This paper presents the methods that design the Wavelet-based pitch detector for 2,4 kbps Harmonic-CELP Coder, and that achieve the effective waveform interpolation by decision window shape of the transition region, Waveform interpolation coder operates by encoding one pitch-period-sized segment, a prototype segment, of speech for each frame, generate the smooth waveform interpolation between the prototype segments for voiced frame, But, harmonic synthesis of the prototype waveforms between previous frame and current frame occur not only waveform errors but also discontinuity at frame boundary on that case of pitch halving or doubling, In addtion, in transition region since waveform interpolation coder synthesizes the excitation waveform by using overlap-add with triangularity window, therefore, Harmonic-CELP fail to model the instantaneous increasing speech and synthesis waveform linearly increases, First of all, in order to detect the precise pitch period, we use the hybrid 1st pitch detector, and increse the precision by using 2nd ACF-pitch detector, Next, in order to modify excitation window, we detect the onset, offset of frame by GCI, As the result, pitch doubling is removed and pitch error rate is decreased 5.4% in comparison with ACF, and is decreased 2,66% in comparison with wavelet detector, MOS test improve 0.13 at transition region.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.99-105
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• 1997

This paper analyzes DLL(Delay lock loop) under the multipath fading effects. The evaluated performance measures include the steady-state timing error probability density function (PDF) and the mean-time-to-lose-lock (MTLL) under multipath fading effects. The discriminator characteristic S(${\epsilon}$) is shown to be zero at the point of timing error ${\epsilon}_{0}$ that is not zero, and the MTLL decreases as the delayed signal power $g_{2}$ and delayed time ${\tau}_{d}$ increase. We approximate the steady-state timing error PDF linearly with these variables and evaluate the steady-state timing error PDF and MTLL. The severe multipath fading effects result lower MTLL, in this case we make MTLL larger by increasing the early-late discriminator offset ${\Delta}$. First, we calculate the timing error point ${\epsilon}_{0}$, and present the performance of DLL under multipath fading. The timing error PDF, MTLL and the performance of DLL with ${\Delta}$ are also investigated. And we conclude that the larger ${\Delta}$ makes a higher MTLL and a better performance of DLL under multipath fading effects.

• The Journal of the Acoustical Society of Korea
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• v.39 no.2
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• pp.120-130
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• 2020

This paper solves the problem in underdetermined convolutive mixture by improving the disadvantages of the multichannel nonnegative matrix factorization technique widely used in blind source separation. In conventional researches based on Spatial Covariance Matrix (SCM), each element composed of values such as power gain of single channel and correlation tends to degrade the quality of the separated sources due to high variance. In this paper, level and frequency normalization is performed to effectively cluster the estimated sources. Therefore, we propose a novel SCM and an effective distance function for cluster pairs. In this paper, the proposed SCM is used for the initialization of the spatial model and used for hierarchical agglomerative clustering in the bottom-up approach. The proposed algorithm was experimented using the 'Signal Separation Evaluation Campaign 2008 development dataset'. As a result, the improvement in most of the performance indicators was confirmed by utilizing the 'Blind Source Separation Eval toolbox', an objective source separation quality verification tool, and especially the performance superiority of the typical SDR of 1 dB to 3.5 dB was verified.

• The Journal of the Acoustical Society of Korea
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• v.16 no.6
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• pp.5-11
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• 1997

Recently, the search for the whereabout of the huge Bell Imperial-Dragon-Temple becomes a great issue. If it happens to be found out and ringing at the original location of the Bell in Kyungjoo City, the Bell might be a great national treasure and lasting to the eternity with her beautiful sound. The Bell was so huge that the total weight of the raw material put into crucibles was 497,581 Kun (289 tons), the shoulder weight 10.3 Chuk (3.14 m) and the maximum thickness 9 Chon (27.4 cm). The Bell was erected in 754 in Shilla Dynasty and was assumed to be lost during the war time by the 3rd invasion of Mongolians (1235~8). However, the author found out that the huge Bell was recast into a new small Bell (8.1 ton) in 1103 by the people of Koryu Dynasty and then the new small Bell was hung in the same position as in the original huge Bell. 135 years later, the new small Bell was carried out by Mongolian forces as a spoil of war from Kyungjoo to the Bay Tonghaegoo, through the saddle point of Mountain Toham, Yangbuk and Riber Great Bell. At the bay, Mongolian forces wished to bring back the Bell to Mongolia by a ship, but they dropped the Bell into the sea by accident. So, if this was the case, the bell at the seabed may be the new small bell (7.4 ton) but not the original huge Bell (41.0 ton) For the evaluation of missing data of the two bells, the author sets up two equations relating all the dimensions and their weights, which seems to be a useful guide to the design of bells. The results of the evaluation of the Bells are as follows. The huge Bell The new small Bell Weight 41.0 ton 7.4 ton Shoulder ht. 3.14 m 2.07 m Mouth diameter 2.468 m 1.546 m Max. thickness 27.4 cm (9 Chon) 11.9 cm (3.9 Chon)