• KSII Transactions on Internet and Information Systems (TIIS)
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• v.2 no.4
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• pp.209-221
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• 2008

In this paper, we propose a subcarrier allocation method and a composite linear minimum mean square error (LMMSE) channel estimator to increase spectrum efficiency in orthogonal frequency division multiplexing (OFDM) transmit diversity. The pilot symbols for OFDM transmit (Alamouti) diversity are exclusively allocated in two OFDM symbols in different antennas, which causes serious degradation of spectrum efficiency. To reduce the number of pilot symbols, our subcarrier allocation method uses repetition-coded data symbols, and the proposed channel estimator maintains good bit error rate (BER) performance.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.470-470
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• 2004

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.275-275
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• 2004

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.743-754
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• 2000

In this paper, we propose wavelet transform image compression method using shuffling and bit plane correlation. Proposed method is that original image decompose into multiresolutions using biorthogonal wavelet transform with linear phase response property and decomposed subbands are classified by maximum classification gain. And classified data sets in each subband are quantized using arbitrary set optimum bit allocation method. Quantized data sets in each subband are shuffled and context based bit plane arithmetic encoded .In context based bit plane arithmetic encoding, the context for each subband is not assigned uniformly, but assigned according to maximum correlation direction. Our results are comparable, or superior for some images at low rates, to published state-of-the-art coders.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.4 s.310
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• pp.113-122
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• 2006

In this paper, we propose a fuzzy method to control bit-rate in the DCT Domain. The method consists of a bit-rate allocation with fuzzy measures and a least-distortion bit-rate reduction. Fuzzy measures are calculated by the code length, the discontinuity ambiguity, and the neighborhood momentum in each DCT block. These measures are summed with weights and form a reduction fuzziness to indicate the degree of preferable reduction. Using the reduction fuzziness. each DCT block is filtered by the least-distortion reduction method to adjust the bit-rate for the target bandwidth. In the experiment, we show the results that the transcoded video quality by the method is better and the bandwidth is more regular than those of existing methods in both visually and quantitatively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1487-1504
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• 1997

In this paper, the two-stage variable block-size multiresolution motion algorithm is proposed for an interframe coding scheme in the wavelet decomposition. An optimal bit allocagion between motion vectors and the prediction error in sense of minimizing the total bit rate is obtained by the proposed algorithm. The proposed algorithm consists of two stages for motion estimatation and only the first stage can be separated and run on its own. The first stage of the algorithm introduces a new method to give the lower bit rate of the displaced frame difference as well as a smooth motion field. In the second stage of the algorithm, the technique is introduced to have more accurate motion vectors in detailed areas, and to decrease the number of motion vectors in uniform areas. The algorithm aims at minimizin gthe total bit rate which is sum of the motion vectors and the displaced frame difference. The optimal bit allocation between motion vectors and displaced frame difference is accomplished by reducing the number of motion vectors in uniform areas and it is based on a botom-up construction of a quadtree. An entropy criterion aims at the control of merge operation. Simulation resuls show that the algorithm lends itself to the wavelet based image sequence coding and outperforms the conventional scheme by up to the maximum 0.28 bpp.

• Journal of the Korea Computer Industry Society
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• v.7 no.5
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• pp.513-520
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• 2006

In this paper, we propose a novel real-time frame-layer rate control algorithm using sliding window method for low bit rate video coding. The proposed rate control method performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. A new frame-layer rate-distortion model is derived, and a non-iterative optimization method is used for low computational complexity. In order to reduce the quality fluctuation, we use a sliding window scheme which does not require the pre-analysis process. Therefore, the proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better visual and PSNR performance than the existing TMN8 rate control method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.7A
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• pp.731-742
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• 2010

We propose an incremental relaying protocol in conjunction with opportunistic communication for differential modulation with an aim to make efficient use of the degrees of freedom of the channels by exploiting a imited feedback signal from the destination. In particular, whenever the direct link from the source to the destination is not favorable to decoding, the destination will request the help from the opportunistic relay (if any). The performance of the proposed system is derived in terms of average bit error probability and achievable spectral efficiency. The analytic results show that the system assisted by the opportunistic relaying can achieve full diversity at low SNR regime and exhibits a 30㏈ gain relative to direct transmission, assuming single-antenna terminals. We also determine the effect of power allocation on the bit error probability BEP) performance of our relaying scheme. We conclude with a discussion on the relationship between the given thresholds and channel resource savings. Monte-Carlo simulations are performed to verify the analysis.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.4
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• pp.25-32
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• 1997

In this paper, we propose an adaptive BTC algorithm using the characteristics of the error signals. The BTC algorithm has a avantage that it is low computational complexity, but a disadvantage that it produces the ragged edges in the reconstructed images for th esloping regions beause of coding the input with 2-level signals. Firstly, proposed methods classify the input into low, medium, and high activity blocks based on the variance of th einput. Using 1-level quantizer for low activity block, 2-level for medium, and 4-level for high, it is adaptive methods that reduce bit rates and the inherent quantization noises in the 2-level quantizer. Also, in case of processing high activity block, we propose a new quantization level allocation algorithm using the characteristics of the error signals between the original signals and the reconstructed signals used by 2-level quantizer, in oder that reduce bit rates superior to the conventional 4-level quantizer. Especially, considering the characteristics of input block, we reduce the bit rates without incurrng the visual noises.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.63-67
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• 1997

The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.