• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1443-1453
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• 1998

This paper presents the speech synthesis engine that converts the character strings kept in a computer memory into the synthesized speech sounds with enhancing the intelligibility and the naturalness by adapting the waveform processing method. The speech engine using demisyllable speech segments receives command streams for pitch modification, duration and energy control. The command based engine isolates the high level processing of text normalization, letter-to-sound and the lexical analysis and the low level processing of signal filtering and pitch processing. The TTS(Text-to-Speech) system implemented by using the speech synthesis engine has three independent object modules of the Text-Normalizer, the Commander and the said Speech Synthesis Engine those of which are easily replaced by other compatible modules. The architecture separating the high level and the low level processing has the advantage of the expandibility and the portability because of the mix-and-match nature.

• Journal of the Korea Society of Computer and Information
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• v.15 no.2
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• pp.109-117
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• 2010

Digital watermarking has drawn extensive attention for protecting digital contents from unauthorized copying. This paper proposes a new watermarking scheme in frequency domain for copyright protection of digital audio. In our proposed watermarking system, the original audio is segmented into non-overlapping frames. Watermarks are then embedded into the selected prominent peaks in the magnitude spectrum of each frame. Watermarks are extracted by performing the inverse operation of watermark embedding process. Simulation results indicate that the proposed scheme is robust against various kinds of attacks such as noise addition, cropping, resampling, re-quantization, MP3 compression, and low pass filtering. Our proposed watermarking system outperforms Cox's method in terms of imperceptibility, while keeping comparable robustness with the Cox's method. Our proposed system achieves SNR (signal-to-noise ratio) values ranging from 20 dB to 28 dB. This is in contrast to Cox's method which achieves SNR values ranging from only 14 dB to 23 dB.

• The Journal of the Acoustical Society of Korea
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• v.24 no.8
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• pp.431-440
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• 2005

The precise measurement of ultrasonic power is important to the qualify assurance and the safety of medical ultrasonic equipments In the Present work, a brier theory was introduced to determine the ultrasonic Power from the time valving balance-indication due to the radiation force acting on an absorbing target and/or other causes such as buoyancy during the repetition of on/off behavior of ultrasonic irradiation. The developed automated system measuring the ultrasonic power was described in detail with the precise mechanical alignment tool , the electric signal generation network, the control and measurement network and the appropriate procedure. The ultrasonic power measured by the developed system was compared to the reference data calibrated by the other national metrology institute at 1 MHz, 5 MHz, 10 MHz, and 15 MHz over the range 10 mW to 10 W. Their relative differences are within $5\%$.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.367-374
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• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.3A
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• pp.198-204
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• 2011

Acoustic communication system is a transmission technology sending sound and data simultaneously. However, data signal can be audible in this system when data is transmitted with high transmission power. The more transmission power is reduced, the more distance that can transmit data is shortened. Therefore, the study that increase the transmission distance is needed. In this paper, we would like to increase transmission distance by adapting receive diversity in acoustic communication system. We measure received performance of both proposed system and Single Input Sing Output (SISO) system according to distance with same transmission power. When SISO satisfies Bit Error Rate (BER) of $7{\times}10^{-3}$ at about 2m, Selection Combining (SC) technique satisfies 2 meters, and Equal Gain Combining (EGC) technique satisfies 4 meters.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Archives of Plastic Surgery
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• v.45 no.3
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• pp.284-288
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• 2018

Dynamic infrared thermography (DIRT) has been used for the preoperative mapping of cutaneous perforators. This technique has shown a positive correlation with intraoperative findings. Our aim was to evaluate the accuracy of perforator mapping with DIRT and augmented reality using a portable projector. For this purpose, three volunteers had both of their anterolateral thighs assessed for the presence and location of cutaneous perforators using DIRT. The obtained image of these "hotspots" was projected back onto the thigh and the presence of Doppler signals within a 10-cm diameter from the midpoint between the lateral patella and the anterior superior iliac spine was assessed using a handheld Doppler device. Hotspots were identified in all six anterolateral thighs and were successfully projected onto the skin. The median number of perforators identified within the area of interest was 5 (range, 3-8) and the median time needed to identify them was 3.5 minutes (range, 3.3-4.0 minutes). Every hotspot was correlated to a Doppler sound signal. In conclusion, augmented reality can be a reliable method for transferring the location of perforators identified by DIRT onto the thigh, facilitating its assessment and yielding a reliable map of potential perforators for flap raising.

• 한국해양학회지
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• v.24 no.4
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• pp.172-183
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• 1989

Acoustical estimation of vertical distributions of zooplankton was carried out by using the backscattered signals of ADCP which was deployed in the Korea Strait in September, 1978. the backscattered signals of ADCP represents the total backward sound intensity caused by the scatters existing within the insonified water column of 8m deep. Based on these backscattered intensities, the estimated number of individual zooplankton (namely, Copepods) varies with depth such that in the surface layer above the thermocline, the numbers are exceedingly larger than the lower layer. It is also shown that a relatively larger number of individuals exists in the bottom cold water layer in the centeral part of the Strait. The horizontal distribution of the zooplankton in the surface layer across the west channel of the Strait shows that the number is higher in the coastal zone of the Korea and it decreases toward the central and then remains constant up to the vicinity of the Tsushima Island where it increases again. This type of distributions is well fitted to the one obtained at several stations by the conventional method at least in qualitative way. Therefore, it is quite plausible to use the ADCP data for monitoring the spatial and temporal distributions of zooplankton.

• Journal of Navigation and Port Research
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• v.35 no.10
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• pp.811-816
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• 2011

In this paper, we analysis to consider the performance of PSPM (Phase Shift Pulse-position Modulation), the one of the low power communication technique, in near-field underwater sound channel by sea trial. PSPM is a QPSK(Quadrature Phase Shift Keying) modulation combined with PPM(Pulse Position Modulation) for low power communication in WBAN(Wireless Body Area Network). It is known that the bandwidth efficiency of PSPM is lower than conventional PSK but the power efficiency increases. In this paper, we will analyze the BER performance of PSPM using data acquired from the sea trials. The BER of QPSK was $6.04{\times}10^{-2}$, PSPM was $3.5{\times}10^{-1}$. Also, PSNR of QPSK was 9.37 dB and in case of PSPM was 9.11 dB.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.05a
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• pp.535-538
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• 2009

Auto Pilot is the system which move automatically the vessel through locating operation mode to automatic after entering operating course using a electronic chart or plotter. And water jet is the a propulsion system that make a power to push the vessel through spouting the accelerated water which is absolbed by the hole in the bottom of vessel. The water jet receive the effect of the depth of water lowly, it's acceletion efficiency is higher under high speed and have an advantage on vibrating and floating sound, so it's demand is increasing as new propulsion system. However, the signal systems of auto Pilot and water jet are defferant, we need the system to communicate between each system. We propose the interface system which communicate between Auto pilot and water jet efficiently in this journal.