• Journal of Communications and Networks
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• 제7권3호
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• Proceedings of the KIEE Conference
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• 대한전기학회 2005년도 학술대회 논문집 정보 및 제어부문
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• pp.484-486
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• 2005

Passenger safety is a primary concern of railway system but, it has been urgent issue that dozens of people are killed every year when they falloff from train platforms. Recently, advancements in IT have enabled applying vision sensors to railway environments, such as CCTV and stereo camera sensors. In this paper, we propose a stereoscopic video coding scheme for subway accident monitoring system. The proposed scheme is designed for providing flexible video among various displays, such as control center, station employees and train driver. We uses MPEG-2 standard for coding the left-view sequence and IBMDC for predicting the P- and B-types of frames of the right-view sequence. IBMDC predicts matching block by interpolating both motion and disparity predicted macroblocks. To provide efficient stereoscopic video service. we define both temporally and spatially scalable layers for each eye's-view by using the concept of Spatio-Temporal scalability. According to the experimental results. we expect the proposed functionalities will play a key role in establishing highly flexible stereoscopic video codec for ubiquitous display environment where devices and network connections are heterogeneous.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 한국해양정보통신학회 2005년도 춘계종합학술대회
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• pp.405-408
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• 2005

This paper proposed the method to images of losses using restorable wavelet transformation. The algorithm proposed in this work stars by processing the pre-quantizer on the original images to organize an image that matches the gray level. The wavelet transformation filter to the original image which is already pre-quantized in order to segment bands. Considering the lowest coding of bands influencing the most to the overall condition of the reconstructed image, it only uses the Huffman coding using prediction. Reconstructed images by proposed algorithm showed higher PSNR when coding images of JPEG or non pre-quantized images. Applying pre-quantizer can control the peak errors and is expected to be useful at mass image compression.

• Journal of the Institute of Electronics Engineers of Korea SP
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• 제47권2호
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• pp.11-19
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• 2010

Most of deblocking filters assumes that the block boundaries are accurately known and the coding information like quantization parameters are available. In some applications such as commercial television, however, many external video inputs without known block boundary and coding information arc given. If a decompressed video sequence heavily degraded with blocking artifacts is given through the external video port, it is absolutely necessary to detect block boundaries and control the strength of deblocking filtering by analysing the given images. This paper presents an efficient method to find the block boundaries and estimate the strength of the blocking artifacts without the knowledge of coding information. In addition, the confidence of the estimated blocking artifact information is also evaluated to control the adaptive deblocking filter robustly. Experiment results show that the estimated block boundary locations and strength relative strength and confidence information are practically good enough to reduce the blocking artifacts without prior knowledge.

• Journal of KIISE:Computing Practices and Letters
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• 제15권2호
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

• Journal of the Institute of Electronics Engineers of Korea SD
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• 제44권11호
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• pp.127-136
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• 2007

An abrupt scene-change frame is one that is hardly correlated with the previous frames. In that case, because an intra-coded frame has less distortion than an inter-coded one, almost all macroblocks are encoded in intra mode. This breaks up the rate control flow and increases the number of bits used. Since the reference software for H.264 takes no special action for a scene-change frame, several studies have been conducted to solve the problem using the quadratic R-D model. However, since this model is more suitable for inter frames, the existing schemes are unsuitable for computing the QP of the scene-change intra frame. In this paper, an improved rate control scheme accounting for the characteristics of intra coding is proposed for scene-change frames. The proposed scheme was validated using 16 test sequences. The results showed that the proposed scheme performed better than the existing H.264 rate control schemes. The PSNR was improved by an average of 0.4-0.6 dB and a maximum of 1.1-1.6 dB. The PSNR fluctuation was also in proved by an average of 18.6 %.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제18권1호
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• pp.110-122
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• 1993

When voice is transmitted through packet switching network, there needs a overload control, that is, a control for the congestion which lasts short periods and occurrs in local extents. In this thesis, we analyzed the performance of the statistical packet voice multiplexer using the overload control strategy by bit dropping. We assume that the voice is coded accordng to (4,2) embedded ADPCM and that the voice packet is generated and transmitted according to the procedures in the CCITT recomendation G. 764. For the performance analysis, we must model the superposed packet arrival process to the multiplexer as exactly as possible. It is well known that interarrival times of the packets are highly correlated and for this reason MMPP is more suited for the modelling in the viewpoint of accuracy. Hence the packet arrival process in modeled as MMPP and the matrix geometric method is used for the performance analysis. Performance analysis is similar to the MMPP IG II queueing system. But the overload control makes the service time distribution G dependent on system status or queue length in the multiplexer. Through the performance analysis we derived the probability generating function for the queue length and using this we derived the mean and standard deviation of the queue length and waiting time. The numerical results are verified through the simulation and the results show that the values embedded in the departure times and that in the arbitrary times are almost the same. Results also show bit dropping reduces the mean and the variation of the queue length and those of the waiting time.

• Proceedings of the KIEE Conference
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• 대한전기학회 1987년도 전기.전자공학 학술대회 논문집(II)
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• pp.1438-1440
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• 1987

In this paper, we discussed the quantization technique in the DCT domain employing a vector quantizer (VQ), and described the relations between the DCT and the VQ. And, we introduced a zonal coding technique for the OCT coefficients based on the classified VQ technique proposed in [2]. We shall show that this technique reduced the coding complexity about 30% while maintaining the same image quality as shown in [2]. A result of simulation with a natural image is also presented.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제30권10A호
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• pp.949-957
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• 2005

In this paper, the performances of a single-user DS-UWB system applying two simple proposed channel estimation schemes are introduced, according to the newly updated DS-UWB PHY Layer standard from IEEE P802.15.3a. The performances of error control coding, different combining schemes in selective Rake receiver for DS-UWB system are analyzed. Both of the two channel estimation schemes using data-independent structure work well in DS-UWB system with few pilot bits. For the purpose of channel estimation and reduces the number of pilot bits, we apply a pilot symbol spreaded with $2{^8}-1\;or\;2{^9}-1$ periods of m-sequence for different channel estimation schemes.

• Proceedings of the IEEK Conference
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.797-798
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• 2008

Visual quality of video telephony over 3G networks is not satisfactory due to limited bandwidths. Therefore, it is worthwhile to enhance subjective visual quality based on ROI coding. In this paper, we propose a rate-control algorithm for video telephony with ROI based H.264|AVC coding. The QP values are increasingly assigned in non-ROI away from the ROI so that graceful degradation of visual quality can be achieved, which is more visually pleasant.