• The KIPS Transactions:PartA
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• v.14A no.3 s.107
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• pp.173-184
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• 2007

Garbage collection in the Java virtual machine is a core function that relieves application programmers of difficulties related to memory management. In this paper, we evaluate and analyze the performance of GenGC and GenRGC, garbage collectors for embedded Java virtual machines. For performance evaluation, we employ CVM, a real embedded Java virtual machine developed by Sun Microsystems, Inc., as a platform and also use a widely-used SpecJVM98 as a set of benchmark programs. To compare the performance of GenGC and GenRGC, we first evaluate the time of garbage collection and the delay time caused by garbage collection. Second, for more detailed performance analysis of GenRGC, we evaluate the time of garbage collection and the delay time caused by garbage collection while changing the sizes of a block and a frame. Third, we analyze the size of storage space required for performing GenRGC, and show GenRGC to be suitable for embedded environment with a limited mont of memory. Since CVM is the most representative one of embedded Java virtual machines, this performance study is quite meaningful in that we can predict the performance of garbage collectors in real application environments more accurately.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.1 s.307
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• pp.27-38
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• 2006

The main purpose of rate control is to achieve the highest video quality when bandwidth or storage capacity is limited. For this purpose, we need a rate control algorithm which is adaptively controlled by the motion information of sequences, scene change, buffer capacity and time-varing bandwitdh channels. A rate-control method in the encoder requires the accurate estimation of target bit for each frame and the low end-to-end delay for transmitting video data by intelligent selection of encoding parameters. In this paper, we suggest three kinds of linear relation in the encoder to satisfy the characteristics of rate control. The first relation is that between the percentage of zero quantized transformed coefficients(p) and coded bits. Second relation is that between the PSNR of encoded frame and its Quantization parameter(QP). Finally, we can find out a linear approximation between QP and p. According to the experimental analysis, the proposed method results in an efficient rate control in terms of the bit estimation, the buffer capacity, and PSNR compared with the existing rate control in the H.264 JM 9.3.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.12
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• pp.108-116
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• 2009

In this paper, we propose a Ethernet based audio distribution system with a new L1/L2 hybrid switching scheme, and evaluate its performance. The proposed scheme not only offers guaranteed low latency and jitter characteristics that are essentially required for the distribution of high-quality uncompressed audio traffic, and but also provide an efficient transmission of data traffic on the Ethernet environment. The audio distribution system with a proposed scheme consists of a master node and a number of relay nodes, and all nodes are mutually connected as a daisy-chain topology through up and downlinks. The master node generates an audio frame for each cycle of 125us, and the audio frame has 24 time slotted audio channels for carrying stereo 24 channels of 16-bit PCM sampled audio. On receiving the audio frame from its upstream node via the downlink, each intermediate node inserts its audio traffic to the reserved time slot for itself, then relays again to next node through its physical layer(L1) transmission - repeating. After reaching the end node, the audio frame is loopbacked through the uplink. On repeating through the uplink, each node makes a copy of audio slot that node has to receive, then play the audio. When the audio transmission is completed, each node works as a normal L2 switch, thus data frames are switched during the remaining period. For supporting this L1/L2 hybrid switching capability, we insert a glue logic for parsing and multiplexing audio and data frames at MII(Media Independent Interlace) between the physical and data link layers. The proposed scheme can provide a good delay performance and transmission efficiency than legacy Ethernet based audio distribution systems. For verifying the feasibility of the proposed L1/L2 hybrid switching scheme, we use OMNeT++ as a simulation tool with various parameters. From the simulation results, one can find that the proposed scheme can provides outstanding characteristics in terms of both jitter characteristic for audio traffic and transmission efficiency of data traffics.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.97-105
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• 2010

This paper proposes an effective medium access mechanism which is based on IEEE 802.11 to transmit an urgent message, The main features of the proposed mechanism are as follows. First, when it comes to channel access to have high priority, it has a shorter interval between frames than that specified in standard document. Second, we use fixed window size instead of back-off window with an exponential increase. Performance evaluation of proposed mechanism is executed by simulation and compare with the node using the specified mechanism in standard document. Performance evaluation results show that according to increasing competition the nodes using proposed method have less accessing time than the conventional methods. Also, the proposed method can improve processing time because of the decreasing transmission delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.11C
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• pp.641-648
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• 2011

Frequency domain equalization is the most promising technology that has relatively low complexity in multipath channel. A frame of single carrier system with frequency domain equalization (SC-FDE) has cyclic prefix to mitigate effect of delay spread. After synchronization and equalization procedure on the SC-FDE system, common phase offset (CPO) that can introduce performance degradation caused by phase mismatch between transmitter and receiver oscillators is remained. In this paper, common phase offset tracking in frequency domain is proposed. To track CPO, constant amplitude zero autocorrelation code sequence as training sequence is adopted. By using numerical results, performance of mean square error is evaluated. The results show that MSE of CPO has similar performance compare to the time-domain estimation and there is no need of domain conversion.

• Journal of Institute of Control, Robotics and Systems
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• v.17 no.1
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• pp.31-37
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• 2011

Recently gantry-type robot with 3 axes rectangular coordinates have been studied in the many industrial production equipment and machinery fields. To acquire a good handling and motion performance of this robot, reducing the settling-time and securing the accurate-transfer positioning under high-speed conditions should be required. However when robot is moved in high-speed, the large inertia of robot can lead to serious vibration of robot's head. The time-delayed control characteristics of this robot can also lead to tracking error. In this research, the analysis of the effects of higher order positional-profile is carried out to assure high-speed performance and stiffness specifications. To remove the residual vibration caused by kinematic coupling effect of dual-servo gantry, we develop a dual-servo gantry of rotary type that moving frame of x-axis rotates about z-axis. In order to decrease the tracking error, the 3 type lead-filter through system identification was applied respectively. From the experimental results, it was shown that zero-order series leader-filter has the best performance about tracking error and settling time.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.4
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• pp.720-740
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• 2011

Wi-Fi-enabled portable devices have recently been introduced into the consumer electronics market. These devices download or upload content, from or to a host machine, such as a personal computer, a laptop, a home gateway, or a media server. This paper investigates the fairness among multiple Wi-Fi-enabled portable devices in a home network when they are simultaneously communicated with the host machine. First, we present that, a simple IEEE 802.11-based home network suffers from unfairness, and the fairness is exaggerated by the wireless link errors. This unfairness is due to the asymmetric response of the TCP to data-packet loss and to acknowledgment-packet loss, and the wireless link errors that occur in the proximity of any node; the errors affect other wireless devices through the interaction at the interface queue of the home gateway. We propose a QoS-provisioning framework in order to achieve per-device fairness and service differentiation. For this purpose, we introduce the medium access price, which denotes an aggregate value of network-wide traffic load, per-device link usage, and per-device link error rate. We implemented the proposed framework in the ns-2 simulator, and carried out a simulation study to evaluate its performance with respect to fairness, service differentiation, loss and delay. The simulation results indicate that the proposed method enforces the per-device fairness, regardless of the number of devices present and regardless of the level of wireless link errors; furthermore it achieves high link utilization with only a small amount of frame losses.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.319-329
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• 2016

Internet Video Coding (IVC) is a royalty-free codec currently being developed in MPEG. Coding efficiency of IVC codec has been steadily enhanced and it was reported that the performance of Committee Draft (CD) version is comparable to H.264/AVC High Profile (HP) in terms of objective and subjective qualities. In this paper, we present some coding tools that have been proposed for enhancing the coding efficiency of IVC during the developing process in MPEG along with brief overview of IVC codec architecture and coding algorithms. The coding tools include both of normative tools and informative tools such as non-reference P frame coding, DC mode intra prediction, Lagrange multiplier selection, and extension of chroma intra prediction modes. Improvement obtained by each tool is presented in terms of algorithm and coding gain based on the experiments. As a result of the experiment, the coding tools give the average bit saving of 8.8%, 0.4%, 0.4%, and 0.0%, respectively, in the low-delay coding mode.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.1012-1021
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• 2005

This paper describes the design and implementation of an efficient multi-point multimedia conference system using IP grouping. Existing multi-point multimedia conference systems are difficult for multi-user to perform efficient cooperation due to bandwidth limitation for data transmission of video, audio and documentation. In the case that multi-user uses limited bandwidth, smooth cooperation does not accomplish due to transmission delay for the real-time transmission of image and speech data. A hybrid transfer method which is mixed with distributed and centralized methods is used for smooth cooperation, and the network bandwidth is reduced by forming multi-user conference systems of IP grouping in this paper. Also, adaptive image frame variations are used to solve bottleneck effect according to the number of users. An efficient multi-user conference system is designed to support audio quality.