• Journal of Internet of Things and Convergence
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• v.10 no.2
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• pp.51-60
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• 2024

Aiming at the problem of low contrast and difficult to recognize video images in low-light environment, we propose an adaptive contrast compensation enhancement algorithm based on human visual perception. First of all, the video image characteristic factors in low-light environment are extracted: AL (average luminance), ABWF (average bandwidth factor), and the mathematical model of human visual CRC(contrast resolution compensation) is established according to the difference of the original image's grayscale/chromaticity level, and the proportion of the three primary colors of the true color is compensated by the integral, respectively. Then, when the degree of compensation is lower than the bright vision precisely distinguishable difference, the compensation threshold is set to linearly compensate the bright vision to the full bandwidth. Finally, the automatic optimization model of the compensation ratio coefficient is established by combining the subjective image quality evaluation and the image characteristic factor. The experimental test results show that the video image adaptive enhancement algorithm has good enhancement effect, good real-time performance, can effectively mine the dark vision information, and can be widely used in different scenes.

• Journal of the Korea Society of Computer and Information
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• v.21 no.4
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• pp.17-24
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• 2016

In this paper, we propose a AH LSU(Advanced Hybrid QoS Routing Link State Update) Algorithm that improves the performance of Hybrid LSU(Hybrid QoS Link State State Update) Algorithm with statistical information of flow holding time in network. AH LSU algorithm has had both advantages of LSU message control in periodic QoS routing LSU algorithm and QoS routing performance in adaptive LSU algorithm. It has the mechanism that calculate LSU message transmission priority using the flow of statistical request bandwidth and available bandwidth and include MLMR(Meaningless LSU Message Removal) mechanism. MLMR mechanism can remove the meaningless LSU message generating repeatedly in short time. We have evaluated the performance of the MLMR mechanism, the proposed algorithm and the existing algorithms on MCI simulation network. We use the performance metric as the QoS routing blocking rate and the mean update rate per link, it thus appears that we have verified the performance of this algorithm.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.3
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• pp.477-493
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• 2011

The IEEE 802.16 mobile multi-hop relay (MMR) task group 'j' (TGj) has introduced the multi-hop relaying concept in the IEEE 802.16 Wireless MAN, wherein a relay station (RS) is employed to improve network coverage and capacity. Several RSs can be deployed between a base station and mobile stations, and configured to form a tree-like multi-hop topology. In such architecture, we consider the problem of a path selection through which the mobile station in and outside the coverage can communicate with the base station. In this paper, we propose a new path selection algorithm that ensures more efficient distribution of resources such as bandwidth among the relaying nodes for improving the overall performance of the network. Performance of our proposed scheme is compared with the path selection algorithms based on loss rate and the shortest path algorithm. Based on the simulation results using ns-2, we show our proposal significantly improves the performance on throughput, latency and bandwidth consumption.

• Proceedings of the IEEK Conference
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• summer
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• pp.233-236
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• 2004

This paper proposes new method for the enhancement of nonexistent high frequency spectral contents from low sample rate audio signal. For example, Due to the protocol constraint, the audio bandwidth of MP3 is restricted to 16Khz. Although band-restricted MP3 audio provide savings of storage space and network bandwidth, it suffers a major problem of a loss in high frequency fidelity such as localization, ambient information, and bright nature of audio. This paper provides a new mathematical analysis for the adaptive estimation of the high frequency contents based on the nature of the input low sample rate audio. Proposed method can be worked globally to any kind of audio such as speech and music that are restricted by sampling rate and bandwidth.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.121-124
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• 2003

Passive optical networks (PONs) will be the pervasive choice in the design of next-generation access networks. One possible solution to implementing Passive optical access network is to rely on wavelength-division multiplexing (WDM). In this paper, we solve the problem of providing real-time service to both hard and soft real-time messages in conjunction with a conventional vest-effort service in WDM. we propose an adaptive scheduling algorithm to schedule and manage the message transmissions in the optical access network. Proposed algorithm is MLF-MQF. Also, we suggest dynamic bandwidth algorithm(DBA) in Loop-back WDM-PON (LPB-PON).our mechanism based on bandwidth reservation protocol.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.23-24
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• 2007

Ethernet passive optical network (EPON) is a technology to solve the access network bottleneck and to realize fiber to the home (FTTH). In EPON, the dynamic bandwidth allocation (DBA) is important because all users share common line in upstream. In this paper, we propose new DBA named as modified EPON dynamic scheduling algorithm (MEDSA) and simulate new DBA using Matlab.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1962-1965
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• 2004

In this paper, the new algorithm for blind adaptive decorrelating parallel interference canceller detector in direct-sequence code division multiple access (DS-CDMA) synchronous communication systems is proposed. The goal of this paper is to improve the performance of the blind adaptive decorrelating parallel interference cancellation detector (BAD/PIC). The proposed blind adaptive decorrelating detector is using optimum step-size technique bootstrap algorithm as an initial stage of PIC, which does not require a training sequence. Therefore, this algorithm has a superior view of utilizing bandwidth and reduces the complexity of computation of inversion cross-correlation matrix. The computer simulation results show that the bit error rate performance of the proposed algorithm for the new structure of detector is better than that of the other detectors such as matched filters, the conventional PIC, and the blind adaptive decorrelating PIC detector.

• The KIPS Transactions:PartB
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• v.17B no.6
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• pp.413-420
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• 2010

In this paper, we propose a scheme for dynamic allocating network bandwidth based on the playback buffer levels of the clients in a distributed mobile multimedia system. In this scheme, the amount of bandwidth allocated to serve a video request depends on the buffer level of the requesting client. If the buffer level of a client is low or high temporarily, more or less bandwidth will be allocated to serve it with an objective to make it more adaptive to the playback situation of this client. By employing the playback buffer level based bandwidth allocation policy, fair services can also be provided to the clients. In order to support high quality video playbacks, video frames must be transported to the client prior to their playback times. The main objectives in this bandwidth allocation scheme are to enhance the quality of service and performance of individual video playback such as to minimize the number of dropped video frames and at the same time to provide fair services to all the concurrent video requests. The performance of the proposed scheme is compared with that of other static bandwidth allocation scheme through extensive simulation experiments, resulting in the 4-9% lower ratio of frames dropped according to the buffer level.

• The Journal of the Korea Contents Association
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• v.10 no.9
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• pp.36-44
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• 2010

In this paper, we propose video quality control system(VQCS) which can control video service quality through the monitoring of end-to-end available bandwidth for video streaming service like IPTV in NGN convergence network. Various multimedia services such as video, voice and gaming service can be provided by IPTV, and these services require large amounts of bandwidth. At this time, video quality degradation like video jerkiness, block distortion and blurring is caused when network available bandwidth is insufficient. Available bandwidth monitoring method is need to stably control video streaming quality. So, we periodically calculate the amount of the packets in link and measure available bandwidth by using total length field in IP header at terminal. Scalability extractor in network selects suitable video streaming data rate based on available bandwidth and transports video streaming with adaptive data rate to prevent video quality deterioration.