• 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.

• ETRI Journal
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• v.36 no.2
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• pp.206-213
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• 2014

This paper presents a novel 90 GHz band 16-quadrature amplitude modulation (16-QAM) orthogonal frequency-division multiplexing (OFDM) communication system. The system can deliver 6 Gbps through six channels with a bandwidth of 3 GHz. Each channel occupies 500 MHz and delivers 1 Gbps using 16-QAM OFDM. To implement the system, a low-noise amplifier and an RF up/down conversion fourth-harmonically pumped mixer are implemented using a $0.1-{\mu}m$ gallium arsenide pseudomorphic high-electron-mobility transistor process. A polarization-division duplex architecture is used for full-duplex communication. In a digital modem, OFDM with 256-point fast Fourier transform and (255, 239) Reed-Solomon forward error correction codecs are used. The modem can compensate for a carrier-frequency offset of up to 50 ppm and a symbol rate offset of up to 1 ppm. Experiment results show that the system can achieve a bit error rate of $10^{-5}$ at a signal-to-noise ratio of about 19.8 dB.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.76-82
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• 2001

In this paper, a novel quantization method for wideband signal codec with 7 kHz bandwidth is proposed. In the transform-based wideband signal codecs, the signal is transformed to frequency domain and the spectral coefficients in each frequency band are quantized based on human perceptual model, followed by Huffman coding. However, the property of each band varies with frequency, and the codec has poor performance when all bands are quantized with the same method. Therefore, a selective quantization method is proposed, which analyzes the band property and selects the quantization domain between frequency domain and time domain based on the quantization efficiency. It is confirmed that the proposed method has better performance than the quantizer of G722.1 codec.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.3 s.309
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• pp.54-60
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• 2006

This paper presents an effective algorithm for transcoding between the Enhanced Variable Rate Codec(EVRC) and G.729A. The simplest way to communicate between heterogeneous speech networks is the cascade connection of two different codecs, called tandem coding. However, tandem coding not only produces high computational loads, but also makes long delay, These problems can be solved by using the transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion and algorithm for reduction of delay. Experimental results show the proposed algorithm produces lower computational complexity, shorter algorithm delay, and similar speech quality when compared with the tandem algorithm.

• Journal of Advanced Marine Engineering and Technology
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• v.38 no.9
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• pp.1125-1130
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• 2014

In this paper, we propose a sound reception system against surrounding noise for ships based on digital signal processing technologies. In order to suppress unwanted surrounding noises, a digital band-pass filter is designed, which the pass-band of the filter is between 70Hz to 820Hz. Also, we develope a sound direction indicating algorithm with 4 microphones. After filtering the audio signals from 4 microphones, the developed sound direction indicating algorithm can indicate 8 directions. In addition, we implement prototype board for the sound reception using a digital signal processor chip and audio codecs, and verify the proposed algorithm.

• JSTS:Journal of Semiconductor Technology and Science
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• v.13 no.5
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• pp.430-442
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• 2013

This paper presents a fast multi-reference frame integer motion estimator for H.264/AVC. The proposed system uses the previously proposed fast multi-reference frame algorithm. The previously proposed algorithm executes a full search area motion estimation in reference frames 0 and 1. After that, the search areas of motion estimation in reference frames 2, 3 and 4 are minimized by a linear relationship between the motion vector and the distances from the current frame to the reference frames. For hardware implementation, the modified algorithm optimizes the search area, reduces the overlapping search area and modifies a division equation. Because the search area is reduced, the amount of computation is reduced by 58.7%. In experimental results, the modified algorithm shows an increase of bit-rate in 0.36% when compared with the five reference frame standard. The pipeline structure and the memory controller are also adopted for real-time video encoding. The proposed system is implemented using 0.13 um CMOS technology, and the gate count is 1089K with 6.50 KB of internal SRAM. It can encode a Full HD video ($1920{\times}1080P@30Hz$) in real-time at a 135 MHz clock speed with 5 reference frames.

• ETRI Journal
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• v.34 no.5
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• pp.649-654
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• 2012

This paper presents a novel 16-quadrature-amplitude-modulation (QAM) E-band communication system. The system can deliver 10 Gbps through eight channels with a bandwidth of 5 GHz (71-76 GHz/81-86 GHz). Each channel occupies 390 MHz and delivers 1.25 Gbps using a 16-QAM. Thus, this system can achieve a bandwidth efficiency of 3.2 bit/s/Hz. To implement the system, a driver amplifier and an RF up-/down-conversion mixer are implemented using a $0.1{\mu}m$ gallium arsenide pseudomorphic high-electron-mobility transistor (GaAs pHEMT) process. A single-IF architecture is chosen for the RF receiver. In the digital modem, 24 square root raised cosine filters and four (255, 239) Reed-Solomon forward error correction codecs are used in parallel. The modem can compensate for a carrier-frequency offset of up to 50 ppm and a symbol rate offset of up to 1 ppm. Experiment results show that the system can achieve a bit error rate of $10^{-5}$ at a signal-to-noise ratio of about 21.5 dB.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.1
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• pp.242-259
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• 2015

Three-dimensional video coding is one of the main challenges restricting the widespread applications of 3D video and free viewpoint video. In this paper, a novel fractal coding algorithm with motion-vector-field-based motion estimation for depth map sequence is proposed. We firstly add pre-search restriction to rule the improper domain blocks out of the matching search process so that the number of blocks involved in the search process can be restricted to a smaller size. Some improvements for motion estimation including initial search point prediction, threshold transition condition and early termination condition are made based on the feature of fractal coding. The motion-vector-field-based adaptive hexagon search algorithm on the basis of center-biased distribution characteristics of depth motion vector is proposed to accelerate the search. Experimental results show that the proposed algorithm can reach optimum levels of quality and save the coding time. The PSNR of synthesized view is increased by 0.56 dB with 36.97% bit rate decrease on average compared with H.264 Full Search. And the depth encoding time is saved by up to 66.47%. Moreover, the proposed fractal depth map sequence codec outperforms the recent alternative codecs by improving the H.264/AVC, especially in much bitrate saving and encoding time reduction.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.258-262
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• 2009

In this paper, we propose an effective memory reduction algorithm to reduce the amount of reference frame buffer and memory bandwidth in video encoder and decoder. In general video codecs, decoded previous frames should be stored and referred to reduce temporal redundancy. Recently, reference frames are recompressed for memory efficiency and bandwidth reduction between a main processor and external memory. However, these algorithms could hurt coding efficiency. Several algorithms have been proposed to reduce the amount of reference memory with minimum quality degradation. They still suffer from quality degradation with fixed-bit allocation. In this paper, we propose an adaptive block-based min-max quantization that considers local characteristics of image. In the proposed algorithm, basic process unit is $8{\times}8$ for memory alignment and apply an adaptive quantization to each $4{\times}4$ block for minimizing quality degradation. We found that the proposed algorithm could improve approximately 37.5% in coding efficiency, compared with an existing memory reduction algorithm, at the same memory reduction rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.575-584
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• 2009

In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.