• Journal of the Korea Society for Simulation
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• v.9 no.3
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• pp.1-11
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• 2000

This paper proposes the practical echo-signal synthesis models - UTAHID (Underwater TArget by Highlight Distribution) & M-UTAHID(Modified UTAHID) - of underwater target for active sonar engineering At high frequencies all the echo components that are the specular reflected waves and various elastic scattering wave scan be regarded the summation of individual echo from some equivalent scattering centers, so the underwater target is characterized by highlights distributed in spatial target structure. Proposed models are compared with characteristics of random distributed model & equivalent interval highlight model, and analyzed target strength, echo-elongation effect, target time spread loss and so on. Thus these can be efficiently used in various real systems related to underwater target echo-signal synthesis on active sonar and acoustic countermeasure.

• Journal of Institute of Control, Robotics and Systems
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• v.17 no.5
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• pp.499-504
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• 2011

In this paper, we suggest a method for discretized frequency modulations of ultrasonic signals. A continuous sweep of frequency modulation signals can be modelled with fine levels of discretization. If the ultrasound signals are modulated with monotonically decreasing frequencies, then the cross-correlation between an emitted signal and received signal can be used to identify the distance of multiple target objects. For the discretized frequency synthesis, CF ultrasounds with different frequencies are serially ordered. The auto-correlation test with the signal shows effective results for distance estimation. The discretized frequency syntheses have better distance resolution than CF ultrasound signals and the resolution depends on the number of the combined ultrasound frequencies.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.3
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• pp.454-468
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• 1997

This paper presents a bitwidth optimization algorithm for efficient hardware sharing in digital signal processing system. The proposed algorithm determines the fixed-point representation for each signal through bitwidth optimization to generate the hardware requiring less area. To reduce the operator area, the algorithm partitions the abstract operations in the design description into several groups, such that the operations in the same group can share an operator. The partitioning result are fed to a high-level synthesis system to generate the pipelined fixed-point datapaths. The proposed algorithm has been implemented in SODAS-DSP an automatic synthesis system for fixed-point DSP hardware. Accepting the models of DSP algorithms in schematics, the system automatically generates the fixed-point datapath and controller satisfying the design constraints in area, speed, and SNR(Signal-to-Noise Ratio). Experimental results show that the efficiency of the proposed algorithm by generates the area-efficient DSP hardwares satisfying performance constraints.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.2
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• pp.141-145
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• 2009

A C-to-SystemC synthesizer which processes the input behavior according to high-level synthesis, and then transforms the synthesis result into SystemC module code is implemented in this paper. In the synthesis process, the input behavioral description in C source code is scheduled using list scheduling algorithm and register allocation is performed using left-edge algorithm on the result of scheduling. In the SystemC process, the output from high-level synthesis process is transformed into SystemC module code by combining it with SystemC features such as channels and ports. The operation of the implemented C-to-SystemC synthesizer is validated through simulating the synthesis of elliptic wave filter in SystemC code. C-to-SystemC synthesizer can be used as a part of tool-chain which helps to implement SystemC design methodology covering from modeling to synthesis.

• Phonetics and Speech Sciences
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• v.8 no.4
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• pp.89-95
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• 2016

The purpose of this work is to analyze and synthesize the pseudo-periodicity of voice using a source model. A speech signal has periodic characteristics; however, it is not completely periodic. While periodicity contributes significantly to the production of prosody, emotional status, etc., pseudo-periodicity contributes to the distinctions between normal and abnormal status, the naturalness of normal speech, etc. Measurement of pseudo-periodicity is typically performed through parameters such as jitter and shimmer. For studying the pseudo-periodic nature of voice in a controlled environment, through collected natural voice, we can only observe the distributions of the parameters, which are limited by the size of collected data. If we can generate voice samples in a controlled manner, experiments that are more diverse can be conducted. In this study, the probability distributions of vowel pitch variation are obtained from the speech signal. Based on the probability distribution of vocal folds, pulses with a designated jitter value are synthesized. Then, the target and re-analyzed jitter values are compared to check the validity of the method. It was found that the jitter synthesis method is useful for normal voice synthesis.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.17 no.10 s.113
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• pp.1007-1019
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• 2006

In this paper, the synthesis method of multibeam radiation pattern with multi-directional beams and nulls is proposed. Genetic algorithm is applied for the optimization of amplitude and phase of the feeding signal into each element that is structured in planar array of single patch antenna with an arbitrary field distribution. To verify the validity of the proposed method, $2{\times}2$ planar array antenna is used for the measurement and the measured far field radiation pattern is compared with the simulation result. As a result, the synthesis of multibeam radiation pattern is implemented by the optimization of amplitude and phase of the feeding signal into each element of $8{\times}8$ planar array antenna system.

• The Journal of the Korea Contents Association
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• v.5 no.3
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• pp.167-173
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• 2005

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) extraction method by using pitch pulses and Zero Crossing Rate in order to unexistent with a voiced and unvoiced consonants in a frame. And this paper present a TSIUVC approximate-synthesis method by using frequency band division. As a result, this method obtains a high quality approximation-synthesis waveform within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. And the TSIUVC extraction rate was $91\%$ for female voice and $96.2\%$ for male voice respectively This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis, and speech synthesis.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.15-22
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• 2000

This paper proposes a distributed highlight algorithm to generate efficiently echo signal for underwater target. In this algorithm, echo signal is synthesized by discontinuity highlights that are varied to incident angle, by equivalent and specified position highlights according to the spatial target structure. Completed UTAHID(Underwater TArget by Highlight Distribution) model is confrimed that PTS, ETS, echo elongation effect, target time spreading loss and envelope fluctuation are satisfied to expected values by various simulations. Thus it can be efficiently used in all sorts of real systems related to underwater target echo signal synthesis in active sonar.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.28-33
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• 2000

For efficient quantization of speech representation, it is common to incorporate Perceptual characteristics of human hearing. However, the focus has been confined only to the magnitude information of speech, and little attention has been paid to phase information. This paper presents a novel approach, termed perceptually irrelevant phase elimination (PIPE), to find out irrelevant phase information of acoustic signals in terms of perception. The proposed method, which is based on the observation that the relative phase relationship within a critical band is perceptually important, is derived not only for stationary Fourier signal but also for harmonic signal. The proposed method is incorporated into the analysis/synthesis system based on harmonic representation of speech, and subjective test results demonstrate the effectiveness of proposed method.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.763-767
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• 2008

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.