• Journal of the Korea Institute of Military Science and Technology
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• v.27 no.2
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• pp.127-139
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• 2024

This paper discusses a gravity compensation technique designed to reduce wavefront error caused by gravity during the assembly and alignment of satellite multi-band optical sensor. For this study, the wavefront error caused by gravity was analyzed for the opto-mechanical structure of multi-band optical sensor. Wavefront error, an indicator of optical performance, was computed by using the displacements of optics calculated through structural analysis and optical sensitivity calculated through optical analysis. Since the calculated wavefront error caused by gravity exceeded the allocated budget, the gravity compensation technique was required. This compensation technique reduces wavefront error effectively by applying the compensation load to the appropriate position of the housing tube. This method successfully meets the wavefront error budget for all bands. In the future, a gravity compensation equipment applying this technique will be manufactured and used for assembly and alignment of multi-band optical sensor.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.6
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• pp.1222-1236
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• 2010

In this paper, we propose an improved distributed multi-view video coding method that is robust to illumination changes among different views. The use of view dependency is not effective for multi-view video because each view has different intrinsic and extrinsic camera parameters. In this paper, a modified distributed multi-view coding method is presented that applies illumination compensation when generating side information. The proposed encoder codes DC values of discrete cosine transform (DCT) coefficients separately by entropy coding. The proposed decoder can generate more accurate side information by using the transmitted DC coefficients to compensate for illumination changes. Furthermore, AC coefficients are coded with conventional entropy or channel coders depending on the frequency band. We found that the proposed algorithm is about 0.1~0.5 dB better than conventional algorithms.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.293-296
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• 2004

This paper proposes an architecture of channel estimation for multi-band OFDM UWB systems presented to IEEE 802.15.3a by Multi-band OFDM alliance(MBOA). The multi-band OFDM (MB-OFDM) systems should have channel estimation for compensation of signal distortion by multi-band channel. The moving-averaging estimation algorithm and multi-band equalization architecture for MB-OFDM UWB systems proposed in this paper was verified by the simulation. Simulation results show that MB-OFDM system with the proposed architecture have the performance improved by about 3.4 dB compared to system with no channel estimation in 0.1$\pi$ phase-rotated channel.

• Journal of Biomedical Engineering Research
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• v.26 no.3
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• pp.177-184
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• 2005

In this paper, we proposed a new hearing aid algorithm to improve SNR(signal to noise ratio) of noisy speech signal and speech perception. The proposed hearing aid algorithm is a multi-band loudness compensation based independent component analysis (ICA). The proposed algorithm was compared with a conventional spectral subtraction algorithm on behind-the-ear type hearing aid. The proposed algorithm successfully separated a target speech signal from background noise and from a mixture of the speech signals. The algorithms were compared each other by means of SNR. The average improvement of SNR by ICA based algorithm was 16.64dB, whereas spectral subtraction algorithm was 8.67dB. From the clinical tests, we concluded that our proposed algorithm would help hearing aid user to hear clearly a target speech in noisy conditions.

• ETRI Journal
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• v.39 no.4
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• pp.535-545
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• 2017

This paper presents a novel K-band (18 GHz) 16-quadrature amplitude modulation (16-QAM) orthogonal frequency-division multiplexing (OFDM)-based $2{\times}2$ line-of-sight multi-input multi-output communication system. The system can deliver 356 Mbps on a 56 MHz channel. Alignment mismatches, such as amplitude and/or phase mismatches, between the transmitter and receiver antennas were examined through hardware experiments. Hardware experimental results revealed that amplitude mismatch is related to antenna size, antenna beam width, and link distance. The proposed system employs an alignment mismatch compensation method. The open-loop architecture of the proposed compensation method is simple and enables facile construction of communication systems. In a digital modem, 16-QAM OFDM with a 512-point fast Fourier transform and (255, 239) Reed-Solomon forward error correction codecs is used. Experimental results show that a bit error rate of $10^{-5}$ is achieved at a signal-to-noise ratio of approximately 18.0 dB.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.41 no.1
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• pp.41-54
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• 2004

In this paper, we propose a new digital hearing aid circuit that compensates the impaired threshold level changing nonlinearly using a multi-band compensation technique. In the algorithm the hearing frequency range 8kHz is divided into 64 bands which is 125Hz resolution. Each band is controlled finely to compensate the hearing impaired proportional to personal ROM table. The multi-band is introduced using a FFT/IFFT Processor which makes to control in frequency domain. As a result, the proposed circuit is more efficient $15\%$ than a conventional ones such as FIR filter architecture in terms of the compensation gun and accuracy. The hardware size was reduced $65\%$ than a general FFT by pre-handling of the input data.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5A
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• pp.432-440
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• 2011

For the 4th generation mobile communication, LTE-advanced system needs the broad frequency band up to 100MHz for providing the data rate of maximum 1Gpbs. However, it is very difficult to secure the broad frequency band in the current frequency allocation situation. So, carrier aggregation was proposed as the solution, in which several fragmented frequency bands are used at the same time. Basically, multiple parallel receivers are required to get the information data from the different frequency bands but this conventional multi-chain receiver system is very inefficient. Therefore, in this paper, we like to study the single chain system that is able to receive the multi-band signals in a single receiver based on the time division multiplexing (TDM) reception method. This proposed TDM receiver efficiently manage to receive the multi-band signals in time domain and handle the baseband signals with one DSP board. However, the serious distortion could be generated by the sampling timing offset (STO) in the TDM-based system. Therefore, we like to analyze STO effects in the TDM-based system and propose a compensation method using estimated STO. Finally, it is shown by simulation that the proposed method is appropriate for the single chain receiver and show good compensation performance.

• Journal of Internet Computing and Services
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• v.19 no.6
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• pp.1-7
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• 2018

This paper present an 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) coding method integrated with ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding) and LMS-MPC(Least Mean Square - Multi Pulse Coding) used V/UV/S(Voiced / Unvoiced / Silence) switching, compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. In integrating several methods, it is important to adjust the bit rate of voiced and unvoiced sound source to 8kbps while reducing the distortion of the speech waveform. In adjusting the bit rate of voiced and unvoiced sound source to 8 kbps, the speech waveform can be synthesized efficiently by restoring the individual pitch intervals using multi pulse in the representative interval. I was implemented that the ACLMS-MPC method and evaluate the SNR of APC-LMS in coding condition in 8kbps. As a result, SNR of ACLMS-MPC was 15.0dB for female voice and 14.3dB for male voice respectively. Therefore, I found that ACLMS-MPC was improved by 0.3dB~1.8dB for male voice and 0.3dB~1.6dB for female voice compared to existing MPC, ACFBD-MPC and LMS-MPC. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or internet phone. In the future, I will study the evaluation of the sound quality of 6.9kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source.

• Journal of Internet Computing and Services
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• v.18 no.5
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• pp.17-22
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• 2017

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. This paper present PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) used V/UV/S( Voiced / Unvoiced / Silence ) switching, position compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. Also, I was implemented that the PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) system and evaluate the SNRseg of PCFBD-MPC in coding condition of 8kbps. As a result, SNRseg of PCFBD-MPC was 13.4dB for female voice and 13.8dB for male voice respectively. In the future, I will study the evaluation of the sound quality of 8kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or a smart phone.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.5
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• pp.501-509
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• 2011

Software defined radio(SDR) has a goal that places the analog-to-digital converter(ADC) as near the antenna as possible. But current technique actually can't do analog-to-digital converting about RF band signals. So one method is studying that samples RF band signals to IF band. One of the ways Sub-Sampling technique can convert signals from RF band to IF band without oscillator. If Sub-Sampling technique is used, over 2 bands can convert signals from RF band to IF band. But due to the filter performance in RF band, it is possible to generate interference between signals that is converted in low frequency band. The effect degrades performance. In this paper, we propose one method that uses time division multiplexing(TDM) method as a solution to avoid interference between signals. By doing TDM and Sub-Sampling at the same time that method can get signals without large changes of structures.