• Journal of Broadcast Engineering
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• v.14 no.5
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• pp.573-588
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• 2009

In this paper, we propose a novel video quality degradation monitoring scheme titled VR-VQMS(Visual Rhythm based Video Quality Monitoring Scheme) over an IPTV service prone to packet losses during network transmission. Proposed scheme quantifies the amount of quality degradation due to packet losses, and can be classified into a RR(reduced-reference) based quality measurement scheme exploiting visual rhythm data of H.264-encoded video frames at a media server and reconstructed ones at an Set-top Box as feature information. Two scenarios, On-line and Off-line VR-VQMS, are proposed as the practical solutions. We define the NPSNR(Networked Peak-to-peak Signal-to-Noise Ratio) modified by the well-known PSNR as a new objective quality metric, and several additional objective and subjective metrics based on it to obtain the statistics on timing, duration, occurrence, and amount of quality degradation. Simulation results show that the proposed method closely approximates the results from 2D video frames and gives good estimation of subjective quality(i.e.,MOS(mean opinion score)) performed by 10 test observers. We expect that the proposed scheme can play a role as a practical solution to monitor the video quality experienced by individual customers in a commercial IPTV service, and be implemented as a small and light agent program running on a resource-limited set-top box.

• Journal of Broadcast Engineering
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• v.25 no.4
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• pp.609-619
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• 2020

In accordance with a demand for high quality video streaming, it needs high data rate in limited bandwidth and more traffic congestion occurs. In particular, when providing real time video service, packet loss rate and bit error probability increase significantly. To solve these problems, a raptor code, which is one of FEC(Forward Error Correction) techniques, is pervasively used in the application layers as a method for improving real-time service quality. In this paper, we propose a method of determining image transmission parameters based on various deep neural networks to increase transmission efficiency at a similar level of image quality by using raptor codes. The proposed neural network uses the packet loss rate, video encoding rate and data rate as inputs, and outputs raptor FEC parameters and packet sizes. The results of the proposed method present that the throughput is 1.2% higher than that of the existing multimedia transmission technique by optimizing the transmission efficiency at a PSNR(Peak Signal-to-Noise Ratio) level similar to that of the existing technique.

• Journal of Communications and Networks
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• v.1 no.2
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• pp.134-142
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• 1999

We have developed fully functional Video Server and Client applications which can transmit, receive, decompress and display compressed video over various networks. Our video trans-port allows dynamic rate control feedback, loss detection, and repair requests from Clients to the Server. Our experiments show how feedvack-before-degradation scheme for rate adaptation maintains good display frame-rate for video playback. We show how the playback degradation(reduction in display frame-rate) oc-curs and what happens if corrective measures are not taken to im-prove the situation. The degradation is attributed to the increased internal kernel buffering which consumes scarce CPU resources. We demontrate with our experimental results that ServerNet, with improved hardware delivery guarantees, can significantly reduce host CPU resource consumption while serving video streams. We present the maximum number of streams which can be served for each of ATM and ServerNet interconnects. The appropriate user-level packet size for the video server are also determined for each case.

• Journal of Broadcast Engineering
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• v.8 no.3
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• pp.259-267
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• 2003

In this paper, we propose a media processor to provide interactive services in mobile multimedia broadcasting environments. The proposed system Is designed to support various functionalities, such as generation of MPEG-4 IOD (Initial Object Descriptor)/OD(Object Descriptor)/BIFS (Binary Format for Scene) data, encapsulation of MPEG-4 AVC (Advanced Video Coding)/BSAC (Bit Sliced Arithmetic Coding) stream and generated IOD/OD/BIFS data into SL (Sync Layer) packet, packetization of SL packet into TS (Transport Stream) packet and multiplexing. The proposed media processor can provide MPEG-4 based interactive services for users.

• The Journal of the Korea Contents Association
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• v.12 no.4
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• pp.69-75
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• 2012

In this paper, to provide various services of QoS, and moreover applying traffic ratio to CAC(Call Admission Control) algorithm tested how long average data rate and the average packet delay time. When CAC algorithm is not applied, traffic mixture ratio is 1:1:4:4, the FTP Service 0.4, web services 0.6, streaming service 0.7, the packet delay requirements are not satisfied. On the other hand CAC Algorithm is applied, all the service of packet delay are satisfied with arrival rate. Therefore, we can make sure that applying of CAC of traffic control WWW, FTP, Video, VoIP can guarantee the various services of QoS.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.5
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• pp.1037-1050
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• 2000

A goal of a BISDN network is to provided integrated transport for a wide range of applications such as teleconferencing, Video On Demand etc. There require multipoint communications in addition to conventional point-to-point connections. Therefore multicast capabilities are very essential in multimedia communications. In this paper, we propose a new multicast cell scheduling method on the Parallel Multicast Packet Switch with Ring network: PMRN which are based on separated HOL. In this method, we place two different HOLs, one for unicast cells and the other for multicast cells. Then using non-FIFO scheduling, we can schedule both unicast cells and multicast cells which are available at the time in the input buffer. The simulation result shows that this method reduces the delay in the input buffer and increases the efficiency of both point-to-point network and ring network and finally enhances the bandwidth of the overall packet switch. A goal of a BISDN network is to provided integrated transport for a wide range of applications such as teleconferencing, Video On Demand etc. There require multipoint communications in addition to conventional point-to-point connections. Therefore multicast capabilities are very essential in multimedia communications. In this paper, we propose a new multicast cell scheduling method on the Parallel Multicast Packet Switch with Ring network: PMRN which are based on separated HOL. In this method, we place two different HOLs, one for unicast cells and the other for multicast cells. Then using non-FIFO scheduling, we can schedule both unicast cells and multicast cells which are available at the time in the input buffer. The simulation result shows that this method reduces the delay in the input buffer and increases the efficiency of both point-to-point network and ring network and finally enhances the bandwidth of the overall packet switch.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.6
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• pp.736-745
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• 2002

In network delivery of compressed video, packets may be lost if the channel is unreliable like Internet. Such losses tend to of cur in burst like continuous bit-stream error. In this paper, we propose an effective error-concealment approach to which an error resilient video encoding approach is applied against burst errors and which reduces a complexity of error concealment at the decoder using data hiding. To improve the performance of error concealment, a temporal and spatial error resilient video encoding approach at encoder is developed to be robust against burst errors. For spatial area of error concealment, block shuffling scheme is introduced to isolate erroneous blocks caused by packet losses. For temporal area of error concealment, we embed parity bits in content data for motion vectors between intra frames or continuous inter frames and recovery loss packet with it at decoder after transmission While error concealment is performed on error blocks of video data at decoder, it is computationally costly to interpolate error video block using neighboring information. So, in this paper, a set of feature are extracted at the encoder and embedded imperceptibly into the original media. If some part of the media data is damaged during transmission, the embedded features can be extracted and used for recovery of lost data with bi-direction interpolation. The use of data hiding leads to reduced complexity at the decoder. Experimental results suggest that our approach can achieve a reasonable quality for packet loss up to 30% over a wide range of video materials.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1039-1051
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• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.7
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• pp.3465-3479
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• 2017

Information-Centric Vehicular Ad Hoc Network (IC-VANET) is a promising network architecture for the future intelligent transport system. Video streaming applications over IC-VANET not only enrich infotainment services, but also provide the drivers and pedestrians real-time visual information to make proper decisions. However, due to the characteristics of wireless link and frequent change of the network topology, the packet loss seriously affects the quality of video streaming applications. In this paper, we propose a REceiver-Driven loss reCOvery Mechanism (REDCOM) to enhance video dissemination over IC-VANET. A Markov chain based estimation model is introduced to capture the real-time network condition. Based on the estimation result, the proposed REDCOM recovers the lost packets by requesting additional forward error correction packets. The REDCOM follows the receiver-driven model of IC-VANET and does not require the infrastructure support to efficiently overcome packet losses. Experimental results demonstrate that the proposed REDCOM improves video quality under various network conditions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6C
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• pp.471-479
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• 2008

In this paper, we propose a novel algorithm to automatically generate an RTP timestamp value that is required for the RTP packetization in order to transmit SVC video over various If networks such as Internet. Unlike the conventional single layer coding algorithms such as H.263, MPEG-4 and H.264, SVC generates a multi-layered single bitstream which is composed of a base layer and one or more enhancement layers in order to simultaneously provide temporal, spatial, and SNR scalability. Especially, in order to provide temporal scalability based on hierarchical B-picture prediction structure, the encoding (or transmission) and display order of pictures in SVC coding is completely decoupled. Thus, the timestamp value to be specified at the header of each RTP packet in video transmission does not increase monotonically according to the display time instant of each picture. Until now, no method for automatically generating an RTP timestamp when SVC video is loaded in a RTP packet has teen introduced. In this paper, a novel automatic RTP timestamp generation method exploiting the TID (temporal ID) field of the SVC NAL unit header is proposed to accommodate the SVC video transmission.