• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.119-122
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• 2000

Recently, many complex DSP (Digital Signal Processing) algorithms have being realized on RISC CPU due to good compilation, low power consumption and large memory space. But, real-time implementation of multiple DSP algorithms on RISC requires the minimum and efficient memory usage and the lower occupancy of CPU. In this thesis, the original floating-point code of MPEG-1 audio decoder is converted to the fixed-point code and then optimized to the efficient assembly code in time-consuming function in accord with RISC feature. Finally, compared with floating-point and fixed-point, about 30 and 3 times speed enhancements are achieved respectively. And 3~4 times memory spaces are spared.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.503-504
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• 2004

휴대용 개인 단말기기(Personal digital assistants)의 높은 보급률에 따라 초소형 스피커(Micro-loudspeaker)의 수요가 급격히 증가하면서, 초소형 스피커의 성능 개선이 절실히 요구되고 있다. 본 연구에서는 초소형 스피커의 진동판 소재로 사용되는 폴리에틸렌 나프탈레이트(PEN)와 폴리에테르 이미르(PEI)의 수지 계열 진동판과 니켈(Ni) 진동판의 방사 효율을 비교 분석하여, 진동계의 관점에서 금속 진동판의 활용 가능성에 대해 고찰하였다.

• Journal of Korea Multimedia Society
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• v.18 no.3
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• pp.323-329
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• 2015

In this paper we applied pattern recognition approach to detect audio forgery. Classification of the microphone types and models can help determining the authenticity of the recordings. Canonical correlation analysis was applied to extract feature for microphone classification. We utilized the linear dependence between two near-silence regions. To utilize the advantage of multi-feature based canonical correlation analysis, we selected three commonly used features to capture the temporal and spectral characteristics. Using three different microphones, we tested the usefulness of multi-feature based characteristics of canonical correlation analysis and compared the results with single feature based method. The performance of classification rate was carried out using the backpropagation neural network. Experimental results show the promise of canonical correlation features for microphone classification.

• Journal of Advanced Navigation Technology
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• v.10 no.4
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• pp.406-412
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• 2006

In this paper, we analyzed the performance of digital watermarking by using audio signal as well as logo image watermark. By utilizing the OFDM/QPSK system under AWGN channel environment, watermarked image are transmitted and detected. Experimental results showed that audio signal-based watermark embedding scheme is superior to that of logo image-based, which is able to restore a signal at SNR=3[dB].

• The Journal of the Convergence on Culture Technology
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• v.8 no.6
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• pp.807-812
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• 2022

Recently, as the number of voice recording files submitted as court evidence increases, the number of cases claiming forgery is also increasing. If the audio recording file structure and metadata, which are objective grounds, are completely forged, it is actually impossible to detect forgery of the sophisticated audio recording file. It is extremely rare for the court to reject the file structure and metadata analysis performed with the forged audio recording file. The purpose of this study is to prove that forgery of voice recording file structure and metadata is easily possible. To this end, in this study, it was introduced that forgery detection is impossible when the 'mixed paste' function, which enables sophisticated editing based on the typification of the editing method of voice recording files, is applied. Moreover, it has been proven through experiments that forgery of file structure and metadata is possible. Therefore, a stricter standard for judging the admissibility of evidence is required when the audio recording file is adopted as digital evidence. This study will not only contribute to the standard of integrity in the adoption of digital evidence by judges, but will also contribute to the method of constructing a dataset for artificial intelligence in detecting forgery of recorded files that is expected to be developed in the future.

• Journal of information and communication convergence engineering
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• v.6 no.3
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• pp.266-269
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• 2008

Layered structure of the Korean terrestrial Digital Multimedia Broadcasting (TDMB) transmission system for multimedia broadcasting service is presented in this paper. We made a switching module which is called the Ensemble Remultiplexer. It is designed to remultiplex the Ensemble Transport Interface (ETI) for T-DMB service. This paper describes the remultiplexing process of the Ensemble Remultiplexer.

• Proceedings of the KIPE Conference
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• 1998.10a
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• pp.206-211
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• 1998

In audio techniques calculation of digital filters is accelerated considerably by use of wave variables instead of voltages and currents. The suitability of wave variables for digital simulation of power converters was investigated and the results are reported in this paper. The original method is described briefly, modelling of switches and diode rectifiers is presented, examples are given and the features of the method are discussed.

• The Journal of the Acoustical Society of Korea
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• v.21 no.8
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• pp.719-726
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• 2002

In this paper, we present a layered watermarking technique for digital audio data that is capable of detecting timing change and adapting complexity in detection. The proposed watermarking uses echo hiding as the first layer, which enables the detector to estimate linear speed change. The spread spectrum watermark is then inserted in the second layer which includes additional information like copyright data. We use two kinds of sequences in the second layer, one of which is for synchronization and the other is for data. The results of previous layer are used to make estimate of timing change in the next layer. The detector in the presented method can select detecting range form the first layer to the first layer, second pre-layer, or second main-layer due to the required system specification. Experimental results show that the proposed watermarking technique is robust to several processing attacks including timing change.

• Journal of Broadcast Engineering
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• v.8 no.3
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• pp.228-237
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• 2003

Eureka-147 DAB(Digital Audio Broadcasting) system on which DMB(Digital Multimedia Broadcasting) system is based, was designed for the requirements of CD qualify audio with ${10}^{-4}$ bit error rate. Audio program may be primary service in DAB system, but multimedia program can be primary service in DMB system. Therefore, the bit error rate required must be below ${10}^{-7}$∼${10}^{-8}$ to transmit multimedia data via DMB channel. In order to meet the requirements and keep backward compatibility of DAB system we propose an outer channel coding scheme using Reed-Solomon coding and convolutional interleaving. This paper shows the simulation results for DMB channel performance based on mobile channel model. Also, it describes the needs and the effects of the outer channel coding.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.3 s.357
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• pp.6-11
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• 2007

This paper proposes CMOS optical receivers to reduce effective area and pulse width distortion (PWD) in high definition digital audio interfaces. To mitigate effective area and PWD, proposed receivers include a frans-impedance amplifier (TIA) with dual output and a level shifter with threshold convergence, respectively. Proposed circuits are fabricated using $0.25{\mu}m$ CMOS process and measured result demonstrated the effective area of $270\times120{\mu}m^2$ and PWD of ${\pm}3%$ for the receiver with a dual output TIA, and the effective area of $410\times140{\mu}m^2$ and PWD of ${\pm}2%$ for the receiver with a threshold convergence level shifter.