• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2005.10a
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• pp.37-42
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• 2005

It is necessary that Fail-Safe Seaway technology providing a continuous navigation solution though fault of navigation system is occurred in sea. This paper focus on signal processing of GPS receiver, one of receivers using the software radio technology to implement a integrated radio navigation system including satellite-based and ground-based navigation systems. It is difficult to implement the software GPS receivers using a commercial processor bemuse of the heavy computational burden for processing the GPS signals in real time. This paper proposes an efficient multi-bit GPS signal processing scheme to reduce the computational burden for processing the GPS signals in the software GPS receiver. The proposed scheme uses a compression concept of the multi-bit replica signals and patterned look-up table method to generate the correlation value between the GPS signals and the replica signals.

• 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 Integrative Natural Science
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• v.4 no.3
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• pp.192-196
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• 2011

New method to encode multiple photonic features of Bragg type reflector on silicon wafer has been investigated. Multiple bit encodes of distributed Bragg reflector features have been prepared by electrochemical etching of crystalline silicon by using various square wave current densities. Optical characterization of multi-encoding of distributed Bragg reflectors on porous silicon was achieved by Ocean optics 2000 spectrometer for the search of possible applications of multiple bit encoding of distributed Bragg reflectors such as multiplexed assays and chemical sensors. The morphology and cross-sectional structure of multi-encoded distributed Bragg reflectors was investigated by field emission scanning electron micrograph.

• The Journal of the Acoustical Society of Korea
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• v.30 no.2
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• pp.86-91
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• 2011

In the underwater communications, the acoustic channel is in poor communication conditions, such as long propagation delay, narrow bandwidth, and high bit-error rate. For these bad acoustic channels, we propose an efficient automatic repeat request (ARQ) for multi-hop underwater network by using the concepts of concurrent bi-directional transmission, multiple sub-packets, and overhearing data packet instead of the acknowledgement signal. Our results show that the proposed ARQ significantly reduces the transmission latency especially in high BER compared with the existing Stop and Wait ARQ.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.4
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• pp.822-833
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• 1996

The video coding for very low bit rate has recently received considerable attention, but conventional block based transform coding schemes suffer from the blocking effect for the constraints of bit rates. In this paper, we present a video coding sysem suing multi-resolution motion estimation/compensation with variable size block(VMRME/C) and multi-resolution vector quantization(MRVQ) in wavelet transform domain for very low bit rate coding. It is shown that the presented scheme has better performance in the peak signal-to-nose ratio(RSNR) by 0.2-0.6 dB as well as subjective quality than that of conventional block based transform video coding techniques(especially, H. 263 which is DCT based video coding).

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.510-521
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• 2002

MBE (multi-band excitation) coder can achieve high qualify synthetic speech below 4.0 kbps. There are, however, significant differences of the fine structure between the original spectrum and the synthetic spectrum. They are mainly due to the exclusive partition of voiced and unvoiced regions in frequency domain and the decision procedure based on the experimental threshold. This paper proposes MMBE (mixed multi-band excitation) speech model to overcome drawbacks of a MBE coder. In addition, two analysis methods, which do not need my decision procedure based on a threshold, are presented. Both voiced and unvoiced components can be mixed over all the frequency axis in the MMBE speech model. To illustrate the potential of the proposed speech model, we develop a 2.6 kbps MMBE coder and compare it with a 2.9 kbps MBE coder by both objective and subjective methods. The results have shown that the proposed coder has a better performance even at a lower bit-rate compared with the MBE coder.

• The Journal of the Convergence on Culture Technology
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• v.6 no.3
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• pp.427-430
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• 2020

The multi-channel analog signal to digital signal conversion is increasing in the field of IoT and medical measurement equipments. It has chip area and power consumption constraints to use a few single or 2_channel ADC for multi_channel application. This paper described to design and implement a proposed comb filter for multi-channel, 24bit ADC. The function of proposed comb filter is verified by matlab simulation and the FPGA test board. It was fabricated using SK Hynix 0.35㎛ CMOS standard process. The performance and chip size is compared with the existing design method that uses integrator/differentiator and FIR construction. The proposed comb filter is expected to use the IoT product and medical measurement equipments that require multi-channel, low power consumption and small hardware size.

• Proceedings of the IEEK Conference
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• 2004.06b
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• pp.533-536
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• 2004

In this paper, CMOS A/D converter with 10bit 32MSPS at 3.3V is designed for HPNA 2.0. In order to obtain the resolution of 10bit and the character of high-speed operation, we present multi-stage type architecture. That consist of sample and hold(S&H), 4bit flash ADC and 4bit Multiplier D/A Converter (MADC) also the Overflow and Underflow for timing error correct of Digital Correct ion Logic (DCL). The proposed ADC is based on 0.35um 3-poly 5-metal N-well CMOS technology. and it consumes 130mW at 3.3V power supply.

• Journal of Advanced Navigation Technology
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• v.11 no.4
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• pp.408-414
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• 2007

Research advances in the areas of micro-sensor device and wireless network technology, has made it possible to develop energy efficient and low cost wireless sensor nodes. In this paper, the forward error control (FEC) scheme for lower power consumption and excellent BER(Bit Error Rate) performance during transmission propose in multi-hop wireless sensor network based on FSK modem. The FEC technique uses extra processing power related to encoding and decoding, it is need complex functions to be built into the sensor node. The probability of receiving a correct bit and codeword for relaying a frame over h nodes to the sink node is calculated as a function of channel parameter, number of hops, number of bits transmitted and the distance between the different nodes.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.1
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• pp.71-76
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• 1997

In this paper, we describe simulation results of subjective assessments with bit rate variation of multi-channel audio codec system for the services of HDTV satellite broadcasting services. Based on this experiment results, we also describe the specification and subjective performance results for 4-channel audio codec. For multi-channel, bit rates are 384,320,256,128kbps and the results show that 320kbps bit rate is needed to compare with the original and the reproduced signal. Based on this, for 384kbps for 4-channel audio codec, three items that achieve a diffgrade worse than -0.5 are due to the noise of analog output module. This system is satisfied for the audio codec of the HDTV system.