• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.7
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• pp.3117-3130
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• 2016

High peak-to-average power ratio (PAPR) of transmitted signals is a major drawback in Multicarrier modulation (MCM) systems. Companding transform is a well-known method to reduce the PAPR without restrictions on system parameters such as the number of subcarriers, frame format and constellation type. In this paper, a novel adaptive companding scheme, mainly focuses on compressing the large signals into the desirable distribution, is proposed to reduce the PAPR with low implementation complexity. In addition, formulas to calculate its PAPR and bit error rate (BER) performance are also derived. Simulation results confirm that the proposed scheme can achieve an effective tradeoff between PAPR reduction and BER performance by carefully choosing the companding parameter.

• The Journal of the Acoustical Society of Korea
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• v.10 no.2
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• pp.44-51
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• 1991

In this paper, we have studied the FM-extended broadcasting which makes the enhancement of stereo signal-to-noise ration by utilizing the compression technique and quadrature moduation method of standard subcarrier signal in transmitter station, and corresponding demodulation method and expansion technique in receiver system. The proposed system is completely compatible with the conventional FM stereo system. We have confirmed the validity of the proposed method by implementation of transmitter/receiver system including quadrature modulation/demodulation and companding system.

• Journal of Biomedical Engineering Research
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• v.28 no.6
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• pp.732-743
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• 2007

Background noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes the results of two experiments in which speech recognition performance was determined for listeners with normal hearing and sensorineural hearing loss in noise environment. First, we compared speech enhancement algorithms by evaluation speech recognition ability in various speech-to-noise ratios and types of noise. Next, speech enhancement algorithms by reducing background noise were presented and evaluated to improve speech intelligibility for sensorineural hearing impairment listeners. We tested three noise reduction methods using single-microphone, such as spectrum subtraction and companding, Wiener filter method, and maximum likelihood envelop estimation. Their responses in background noise were investigated and compared with those by the speech enhancement algorithm that presented in this paper. The methods improved speech recognition test score for the sensorineural hearing impaired listeners, but not for normal listeners. The results suggest the speech enhancement algorithm with the loudness compression can improve speech intelligibility for listeners with sensorineural hearing loss.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.18 no.2
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• pp.39-47
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• 2008

Among previous Side Channel Analysis (SCA) methods, Differential Power Analysis (DPA) based on the statistical characteristics of collected signals has been known as an efficient attack for uncovering secret key of cryptosystems. However, the attack performance of this method is affected very much by the temporal misalignment and noise of collected side channel signals. In this paper, we propose a new method to surmount the noise problem in DPA. The performance of the proposed method is then evaluated while analyzing the power consumption signals of Micro-controller chips during a DES operation. Its performance is then compared to that of the original DPA in the time and frequency domains. When we compare the experimental results with respect to the needed number of traces to uncover the secret key, our proposed method shows the performance enhancement 33% in the time domain and 50% in the frequency domain.

• The KIPS Transactions:PartC
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• v.18C no.4
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• pp.213-216
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• 2011

Differential Power Analysis is fully affected by various noises including temporal misalignment. Recently, Ryoo et al have introduced an efficient preprocessor method leading to improvements in DPA by removing the noise signals. This paper experimentally proves that the existing preprocessor method is not applied to all processor. To overcome this defect, we propose a Differential Trace Model(DTM). Also, we theoretically prove and experimentally confirm that the proposed DTM suites DPA.

• Journal of Broadcast Engineering
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• v.24 no.4
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• pp.670-675
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• 2019

AM radio broadcasting has the advantage that the sound quality is lower than that of FM radio but the service area is wide and at night it can be transmitted even to overseas due to ionospheric reflection. AM radio broadcasts usually transmit large amounts of several tens to several hundreds of kilowatts (KW), requiring a lot of power, which is a financial burden on broadcasters. Recently, it is required to introduce a way to reduce the power of AM radio broadcasting in Korea. Therefore, it is possible to save a certain amount of power compared to the conventional AM radio broadcasting, and it is possible to maintain the existing license service area and sound quality, and what policies are needed to introduce the technology. In this paper, we predict the reduction of transmission power when a high efficiency transmission system is applied to existing AM radio broadcasting transmission system. As a result of the research, AMC adopting the high efficiency transmission method instead of the conventional AM radio transmission method, and when the modulation ratio of AM radio is 70%, if -3 dB companding is applied, the annual transmission power charge is decreased from 5,185,838,160Won to 3,528,900,086Won with saving of 32.0%, When -6 dB companding is applied, it is expected to be reduced to 2,608,378,934Won, which is 49.7% savings.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.7 no.2
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• pp.47-54
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• 1982

In this paper, a improved per-channel PCM Coder with 1-bit interpolation is proposed. The coder converts a telephone signal to 15-segments u-law PCM signal of a large dynamic range. The A/D conversion technique of the proposed converter requires a feedback loop around a quantizer operates at high speed, and a accumulater for accumulating the quantized values to provide PCM outputs. To obtain both linear and compressed PCM signals a improved table look-up method is presented. The operations of the proposed converter are certified through the experiments to be good. The experimental circuit comprises TTL logic gates, a resistive D/Z converter and a simple differential amplifier. From the results of the experiments, it is known that the proposed converter has many advantage to be adopted economically for per-channel onverter used in rural area service.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.84-89
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• 2002

In this paper, we present the design method of noise reduction circuit in telephonic system. The circuit consists of compressor, expander and a filter. The basic idea of a proposed method compresses the audible signal in order to mask the channel noise during transmission and then expand at the reverse rate the transmitted signal to naturally recover the original signal. Of course, there should be no distortion or other degradation of the audio itself in passing through companding(compress/expand) cycle. In the compressing process, the gain of compressor is automatically controlled by the envelope level of input signal in order to increase the effective dynamic range of input signal and to improve the signal to noise ratio. The compressed rate is the root time of a audible signal. The compressed signal should be expanded at the square time of the signal to recover a original signal. Simulation shows the proposed method improves the performance of the noise reduction of a channel noise as well as stability. 

• Journal of the Korea Computer Industry Society
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• v.5 no.5
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• pp.737-746
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• 2004

Two-channel multiwavelet system is investigated for image compression application in this paper. Generally, multiwavelets are known for their superb capability of compressing non-stationary signals like voice. However, multivavelet system have a critical problem in processing and compressing image data due to mesh-grid visual artifacts. In our two-channel multiwavelet system we have investigated incorporation of pre and post filtering to the multiwavelet transform and compression system for alleviating those ingerent visual artifacts due to multiwavelet effect. In addition, to quantify the image data compression performance of proposed multiwavelet system, computer simulations have been performed using various image data. For bit allocation and quantization, the Lagrange multiplier technique considering data rate vs. distortion rate along with a nonlinear companding method are applied equallly to all systems considered, here. The simulation results have yielded 1 ~ 2 dB compression enhancement over the scalar savelet systems. If the more advanced compression methods like SPIHT and run-length channel coding were adopted for the proposed multiwavelet system, a much higher compression gain could be obtained.