• Korean Journal of Optics and Photonics
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• v.16 no.4
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• pp.313-318
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• 2005

We propose an active stabilization method for the receiver of NRZ-DPSK transmission. The 1-bit delayed Mach-Zehlder interferometer is thermally controlled to maintain the largest DC component power ratio between the constructive and destructive output ports, for the optimum transmission condition. This method is very cost effective since no additional components are required. Experimental results show that the proposed scheme guarantees error free performance even when there was ~ 1 GHz optical carrier frequency fluctuation in 10 Gbps transmission.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04a
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• pp.453-456
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• 2001

21세기에 새로운 분야로 대두되고 있는 가상현실은 많은 사람들의 흥미를 끌고 있다. 상상 속에서나 가능하던 일들을 현실감과 입체감을 통해 실제처럼 느낄 수 있게 해준다는 것이 가상현실의 가장 큰 매력일 것이다. 가상현실을 요하는 멀티미디어 기기들의 활발한 시장진출로 3D 효과를 가진 오디오/비디오의 하드웨어 구현이 불가피하다. 본 연구에서는 휴대용 기기들에서 실시간 3D 입체음향 효과를 얻을 수 있도록 하드웨어를 구성하였다. 기존의 입체음향 기술에서 사용되는 콘볼루션 방법은 계산량이 많기 때문에 실시간 구현이 어렵다. 그러나 제안된 방식은 FFT를 사용하여 주파수 영역에서 처리함으로써 계산량을 줄여 하나의 프로세서로도 실시간 처리가 가능하도록 하였다. 저가/저전력/소형화조건을 요구하는 휴대용 기기에서 3D 입체 음향 효과를 얻을 수 있는 것이다. DSP는 TI(Texas Instruments)사의 16비트 고정소수점(fixed-point) 프로세서인 TMS320C5416을 사용한다. 구현된 3D 입체음향 칩은 입체음향을 필요로 하는 휴대용 MP3 Player, 가전용 오디오/비디오 등에 응용될 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.1
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• pp.85-100
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• 1991

In the discussion of digital mobile communication systems the speech coder based on the high quality low bit rate is an essential part of topics to overcome the limited availability of radio spectrum, which will enhance the communication services. In this paper we present the implementation and performance evaluation of 13kbps RPE LTP speech coder. An implementation of a real time full duplex coder with 75% of DSP loading rate using a single DSP chip has been shown, and also the fixed point simulations for H/W implementation has been performed. Finally, analysis result for relative bit importance of each transmitting parameter has been shown for channel coding.

• Journal of the Korea Society for Simulation
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• v.27 no.4
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• pp.27-45
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• 2018

The terrestrial communication network and the underwater acoustic communication network have very different communication characteristics each other in operational environments, communication media, propagation delay, frequency bandwidth, transmission speed, bit error rate, and so on. These different characteristics cause some different address schemes and different maximum transmission units and, as a result, these differences must form certainly obstacles to the intercommunication between a terrestrial communication network and an underwater acoustic communication network. In this paper, we presents a method to use the virtual addresses to resolve the interconnection problem caused by different address schemes between a terrestrial communication network and an underwater acoustic communication network, and, through a mathematical modeling, we analyze the performance on the message transceiving delay time in the underwater environment.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.949-952
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• 2009

In this paper, a watermarking method based on the trellis code with multi-layer is proposed. An image is divided $8{\times}8$ block with no overlapping, and compute the discrete cosine transform(DCT) of each block, and the 12 medium-frequency AC terms from each block are extracted. Next it is compared with gaussian random vectors with zero mean and unit variance. As these processing, the embedding vectors with minimum linear correlation can be obtained by Viterbi algorithm at each layer of trellis coding. To evaluate the performance of proposed method, the average bit error rate of watermark message is calculated from different several images.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.365-373
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• 2010

For effective underwater digital communications, a frame structure is used, which includes pilots in time and frequency domains for channel estimation at a receiver. To estimate channel precisely, the each pilot should be located less than coherence time and coherence bandwidth. This paper measured underwater communication environments to provide coherence time and coherence bandwidth. Based on the measurement, the paper exhibits the calculated coherence time and coherent bandwidth is adequate by computer simulations.

• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.269-275
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• 2007

Pervasive Multi-channel audio systems are being realized due to advances in digital technology. This paper proposes an efficient system that serially connects individual speakers with bidirectional digital communication capability by means of SoC design. In particular, each speaker can identify the bit stream assigned to the speaker and convert it into analog audio. Furthermore, the speaker can self-diagnose the speaker functionality by utilizing the designed capability to measure frequencies of various square wave test signals. The proposed system running on 200MHz clock yielded restoration of analog output signal with latency of only $500{\mu}s$ compared to directly driving the speakers in a traditional way.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.12
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• pp.3235-3245
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• 1996

A DDFS(Direct Digital Frequency Synthesizer) used in spread spectrum communication systems must need fast switching speed, high resolution(the step size of the synthesizer), small size and low power. The chip has been designed with four parallel sine look-up table to achieve four times throughput of a single DDFS. To achieve a high processing speed DDFS chip, a 24-bit pipelined CMOS technique has been applied to the phase accumulator design. To reduce the size of the ROM, each sine ROM of the DDFS is stored 0-.pi./2 sine wave data by taking advantage of the fact that only one quadrant of the sine needs to be stored, since the sine the sine has symmetric property. And the 8 bit of phase accumulator's output are used as ROM addresses, and the 2 MSBs control the quadrants to synthesis the sine wave. To compensate the spectrum purity ty phase truncation, the DDFS use a noise shaper that structure like a phase accumlator. The system input clock is divided clock, 1/2*clock, and 1/4*clock. and the system use a low frequency(1/4*clock) except MUX block, so reduce the power consumption. A 107MHz DDFS(Direct Digital Frequency Synthesizer) implemented using 0.8.mu.m CMOS gate array technologies is presented. The synthesizer covers a bandwidth from DC to 26.5MHz in steps of 1.48Hz with a switching speed of 0.5.mu.s and a turing latency of 55 clock cycles. The DDFS synthesizes 10 bit sine waveforms with a spectral purity of -65dBc. Power consumption is 276.5mW at 40MHz and 5V.

• Journal of the Korean Institute of Telematics and Electronics D
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• v.35D no.10
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• pp.39-50
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• 1998

Wide-band sharp-transition filters are widely used in applications such as wireless CODEC design or medical systems. Since these filters suffer from large sensitivity and roundoff noise, large word-length is required for the VLSI implementation, which increases the hardware size and the power consumption of the chip. In this paper, a low-power implementation technique for digital filters with wide-band sharp-transition characteristics is proposed using CPL (Complementary Pass-Transistor Logic), LWDF (Lattice Wave Digital Filter) and a modified DIFIR (Decomposed & Interpolated FIR) algorithm. To reduce the short-circuit current component in CPL circuits due to threshold voltage reduction through the pass transistor, three different approaches can be used: cross-coupled PMOS latch, PMOS body biasing and weak PMOS latch. Of the three, the cross-coupled PMOS latch approach is the most realistic solution when the noise margin as well as the energy-delay product is considered. To optimize CPL transistor size with insight, the empirical formulas for the delay and energy consumption in the basic structure of CPL circuits were derived from the simulation results. In addition, the filter coefficients are encoded using CSD (Canonic Signed Digit) format and optimized by a coefficient quantization program. The hardware cost is minimized further by a modified DIFIR algorithm. Simulation result shows that the proposed method can achieve about 38% reductions in power consumption compared with the conventional method.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.6
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• pp.884-889
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• 2010

Recently, a music genre classification has been studied. However, experts use different criteria to classify each of these classifications is difficult to derive accurate results. In addition, when the emergence of a new genre of music genre is a newly re-defined. Music as a genre rather than to separate search should be classified as emotional words. In this paper, the feelings of people on the basis of brightness and darkness tries to categorize music. The proposed classification system by applying VCM(Variance Considered Machines) is the contrast of the music. In this paper, we are using three kinds of musical characteristics. Based on surveys made throughout the learning, based on musical attributes(beat, timbre, note) was used to study in the VCM. VCM is classified by the trained compared with the results of the survey were analyzed. Note extraction using the MATLAB, sampled at regular intervals to share music via the FFT frequency analysis by the sector average is defined as representing the element extracted note by quantifying the height of the entire distribution was identified. Cumulative frequency distribution in the entire frequency rage, using the difference in Timbre and were quantified. VCM applied to these three characteristics with the experimental results by comparing the survey results to see the contrast of the music with a probability of 95.4% confirmed that the two separate.