• Journal of Broadcast Engineering
• /
• v.12 no.2
• /
• pp.102-111
• /
• 2007

Currently, the efficient adaptive video streaming services and coding techniques are required to ensure quality of service (QoS) in diverse multimedia environments. To maintain optimal video quality in these environments, it is required to support scalable video coding (SVC) technique and proper QoS metric methods. In this paper, we propose a new quality metric that takes in to the temporal, the SNR, the spatial scalability on SVC. Experimental results shows that proposed method has a higher correlation with the subjective quality. We expect that the proposed metric plays an important role in deciding the coding parameter to optimize the quality of the SVC based video.

• KIISE Transactions on Computing Practices
• /
• v.22 no.10
• /
• pp.479-487
• /
• 2016

People can listen to almost every type of music by music streaming services without possessing music. Ironically it is difficult to choose what to listen to. A music recommendation system helps people in making a choice. However, existing recommendation systems have high computation complexity and do not consider context information. Emotion is one of the most important context information of music. Lyrics can be easily computed with various language processing techniques and can even be used to extract emotion of music from itself. We suggest a music-level similarity evaluation method using emotion and structure. Our result shows that it is important to consider semantic information when we evaluate similarity of music.

• Journal of KIISE:Computer Systems and Theory
• /
• v.35 no.9_10
• /
• pp.466-475
• /
• 2008

Owing to the improved wireless communication technologies, it is possible to provide streaming service of multimedia with PDAs and mobile phones in addition to desktop PCs. Since mobile client devices have low computing power and low network bandwidth due to wireless network, the transcoding technology to adapt media for mobile client devices considering their characteristics is necessary. Transcoding servers transcode the source media to the target media within corresponding grades and provide QoS in real-time. In particular, an effective load balancing policy for transcoding servers is inevitable to support QoS for large scale mobile users. In this paper, the transcoding load estimation algorithm is proposed for load balance on the distributed transcoding environments. The proposed algorithm estimates transcoding time from transcoding server information, movie information and target transcoding bit-rate. The estimated transcoding time is proved based on experiments.

• Journal of the Institute of Electronics Engineers of Korea CI
• /
• v.44 no.6
• /
• pp.43-50
• /
• 2007

In this paper we propose a method that identifies users at H.264 streaming using watermarking with fingerprints. The watermark can efficiently reduce the potential danger of forgery or alteration. Especially a biometric watermark has various advantages. Among entire biometric characteristics, the fingerprint is the most convenient and economical. In this paper we propose a novel fingerprint-based watermarking technique that can survive under very low bit-rate compression. The proposed algorithm consists of enhancement of a fingerprint image, the watermark generation using the extracted feature coordinates, watermark insertion using discrete wavelet transform, and authentication. The proposed algorithm can achieve robust watermark extraction against 0.264 compressed videos.

• Journal of Digital Contents Society
• /
• v.3 no.1
• /
• pp.47-57
• /
• 2002

Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).

• Journal of Broadcast Engineering
• /
• v.16 no.5
• /
• pp.738-749
• /
• 2011

In this paper, we study QoS(Quality of Service) control protocols and the effect using MAC/PHY parameters of client device in mobile network. We proposes the way of controling the bit-rate by estimating the channel condition of the client with measured MAC/PHY parameters which is sent from the client. With the proposed method, more accurate available bit-rate can be estimated compared to conventional protocol, RTCP(Real-time Transport Control Protocol). The accurate bit-rate estimation can decrease wasted bit-rate and transport delay. In the result of the advantages, the transported video quality can be enhanced. In this paper, we show the effects of enhancement using client's the field data measured in WiMAX.

• Journal of Korea Multimedia Society
• /
• v.13 no.3
• /
• pp.349-358
• /
• 2010

Adaptability is an important issue in video streaming over mobile environments, since the clients may request videos with great differences in their workload. In this paper, we propose the issues in limited bandwidth scheduling for efficient MPEG-4 video stream transmission over a mobile or wireless network. In the phase of admission control, the amount of bandwidth allocated to serve a video request is the mean bandwidth requirement of its requested video. The dynamic allocation of bandwidth in the phase of scheduling depends on the playback buffer levels of the clients with an objective to make it more adaptive to the playback situation of individual clients. In the proposed RTA scheduling protocol, more bandwidth may be allocated temporarily to the client whose buffer level is low. By employing the buffer level based scheduling policy, this protocol attempts to maximize the real-time performance of individual playback while minimizing the impact of transient overloading. Extensive simulation experiments have been performed to investigate the performance characteristics of the RTA protocol as comparing with BSBA protocol. This RTA protocol shows the better performance by transferring more frames than BSBA protocol.Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2016.10a
• /
• pp.114-116
• /
• 2016

Recently, SDN and NFV technology have been developed actively and provide enormous flexibility of network provisioning. The future network services would generally involve many different types of services such as hologram games, social network live streaming videos and cloud-computing services, which have dynamic service requirements. To provision networks for future services dynamically and efficiently, SDN/NFV orchestrators must clearly understand the service requirements. Currently, network provisioning relies heavily on QoS parameters such as bandwidth, delay, jitter and throughput, and those parameters are necessary to describe the network requirements of a service. However it is often difficult for users to understand and use them proficiently. Therefore, in order to maintain interoperability and homogeneity, it is required to have a service abstraction layer between users and orchestrators. The service abstraction layer analyzes ambiguous user's requirements for the desired services, and this layer generates corresponding refined services requirements. In this paper, we present our initial effort to design a Smart Service Abstraction Layer (SmSAL) for future network architecture, which takes advantage of machine learning method to analyze ambiguous and abstracted user-friendly input parameters and generate corresponding network parameters of the desired service for better network provisioning. As an initial proof-of-concept implementation for providing viability of the proposed idea, we implemented SmSAL with a decision tree model created by learning process with previous service requests in order to generate network parameters related to various audio and video services, and showed that the parameters are generated successfully.

• KIPS Transactions on Computer and Communication Systems
• /
• v.4 no.4
• /
• pp.141-146
• /
• 2015

Data have been increased enormously due to the development of IT technology such as recent smart equipments, social network services and streaming services. To meet these environments the technologies that can treat mass data have received attention, and the typical one is Hadoop. Hadoop is on the basis of open source, and it has been designed to be used at general purpose computers on the basis of Linux. To initial Hadoop nearly no security was introduced, but as the number of users increased data that need security increased and there appeared new version that introduced Kerberos and Token system in 2009. But in this method there was a problem that only one secret key can be used and access permission to blocks cannot be authenticated to each user, and there were weak points that replay attack and spoofing attack were possible. Hence, to supplement these weak points and to maintain efficiency a protocol on the basis of group key, in which users are authenticated in logical group and then this is reflected to token, is proposed in this paper. The result shows that it has solved the weak points and there is no big overhead in terms of efficiency.

• Applied Chemistry for Engineering
• /
• v.20 no.5
• /
• pp.557-560
• /
• 2009

In this study, a new wet etching method and the solution for thinning the glass with the thickness below $100{\mu}m$ were investigated. For the preparation of etching solution with low hydrofluoric acid, it was effective to use $NH_4F$ or $NH_4HF_2$ as a main ingredient with the addition of sulfuric acid or nitric acid. Influence of the composition of mixed acid solution and the temperature on the etching rate was investigated. The addition of anionic surfactant provides the function to prevent the adhesion of sludge generated by the etching reaction. A new wet etching pilot device equipped with streaming generation parts was used to test etching of commercial non-alkali glass and soda lime glass. The non-alkali glass with the thickness of 640 ${\mu}m$ and soda lime glass with the thickness of $500{\mu}m$ were etched to $45{\mu}m$ and $100{\mu}m$, respectively, by using the pilot device. After the etching by pilot device, the roughness degree of the glass surface was maintained at $0.01{\sim}0.02{\mu}m$.