• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.3
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• pp.44-53
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• 2010

The H.264/AVC has been designed to use 9 directional intra prediction modes for removing spatial redundancy. It also employs high correlation between neighbouring block modes in sending mode information. For indication of the mode, smaller bits are assigned for higher probable modes and are compressed by predicting the mode with minimum value between two prediction modes of neighboring two blocks. In this paper, we calculated the statistical probability of prediction modes of the current block to exploit the correlation among the modes of neighboring two blocks with several test video sequences. Then, we made the probable prediction table that lists 5 most probable candidate modes for all possible combinatorial modes of upper and left blocks. By using this probability table, one of 5 higher probable candidate modes is selected based on RD-optimization to reduce computational complexity and determines the most probable mode for each cases for improving compression performance. The compression performance of the proposed algorithm is around 1.1%~1.50%, compared with JM14.2 and we achieved 18.46%~36.03% improvement in decoding speed.

• Journal of KIISE:Computer Systems and Theory
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• v.30 no.1
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• pp.33-49
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• 2003

Recently, there have been large storage demands for manipulating multimedia data. To solve the tremendous storage demands, one of the major researches is the SAN(Storage Area Network) that provides the local file requests directly from shared-disk storage and also eliminates the server bottlenecks to performance and availability. SAN also improve the network latency and bandwidth through new channel interface like FC(Fibre Channel). But to manipulate the efficient storage network like SAN, traditional local file system and distributed file system are not adaptable and also are lack of researches in terms of a metadata structure for large-scale inode object such as file and directory. In this paper, we describe the architecture and design issues of our shared-disk file system and provide the efficient bitmap for providing the well-formed block allocation in each host, extent-based semi flat structure for storing large-scale file data, and two-phase directory structure of using Extendible Hashing. Also we describe a detailed algorithm for implementing the file system's device driver in Linux Kernel and compare our file system with the general file system like EXT2 and shard disk file system like GFS in terms of file creation, directory creation and I/O rate.

• Journal of the Korea Society of Computer and Information
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• v.17 no.1
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• pp.1-10
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• 2012

Improving operating frequency of processors is no longer today's issues; a multiprocessor technique which integrates many processors has received increasing attention. Currently, high-performance processors that integrate 64 or 128 cores are developing for large data processing over 2, 4, or 8 processor cores. This paper proposes an optimal many-core processor for synthesizing guitar sounds. Unlike the previous research in which a processing element (PE) was assigned to support one of guitar strings, this paper evaluates the impacts of mapping different numbers of PEs to one guitar string in terms of performance and both area and energy efficiencies using architectural and workload simulations. Experimental results show that the maximum area energy efficiencies were achieved at PEs=24 and 96, respectively, for synthesizing guitar sounds with sampling rate of 44.1kHz and 16-bit quantization. The synthesized sounds were very similar to original guitar sounds in their spectra. In addition, the proposed many-core processor was 1,235 and 22 times better than TI TMS320C6416 in area and energy efficiencies, respectively.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.106-115
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• 2003

Diverse multimedia services will be deployed at hand on 3G-and-beyond multi-service CDMA systems in order to satisfy different quality of service (QoS) according to traffic types. In order to use appropriate resources efficiently the call admission control (CAC) as a major resource control mechanism needs to be used to take care of efficient utilization of limited resources. In this paper, we propose a QoS-aware CAC (QCAC) that is enabled to provide service fairness and service differentiation in accordance with priority order and that applies the different thresholds in received power considering different QoS requirements such as different bit error rates (BER) when adopting total received power as the ceil load estimation. The proposed QCAC calculates the different thresholds of the different traffic types based on different required BER applies it for admission policy, and can get service fairness and differentiation in terms of call dropping probability as a main performance metric. The QCAC is aware of the QoS requirement per traffic type and allows admission discrimination according to traffic types in order to minimize the probability of QoS violation. Also the CAC needs to consider the resource allocation schemes such as complete sharing (CS), complete partitioning (CP), and priority sharing(PS) in order to provide fairness and service differentiation among traffic types. Among them, PS is closely related with the proposed QCAC having differently calculated threshold per each traffic type according to traffic priority orders.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1
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• pp.97-102
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• 2000

Subband coding is to divide the signal frequency band into a set of uncorrelated frequency bands by filtering and then to encode each of these subbands using a bit allocation rationale matched to the signal energy in that subband. The actual coding of the subband signal can be done using waveform encoding techniques such as PCM, DPCM and vector quantizer(VQ) in order to obtain higher data compression. Most researchers have focused on the error in the quantizer, but not on the overall reconstruction error and its dependence on the filter bank. This paper provides a thorough analysis of subband codecs and further development of optimum filter bank design using vector quantizer. We compute the mean squared reconstruction error(MSE) which depends on N the number of entries in each code book, k the length of each code word, and on the filter bank coefficients. We form this MSE measure in terms of the equivalent quantization model and find the optimum FIR filter coefficients for each channel in the M-band structure for a given bit rate, given filter length, and given input signal correlation model. Specific design examples are worked out for 4-tap filter in 2-band paraunitary filter bank structure. These optimum paraunitary filter coefficients are obtained by using Monte Carlo simulation. We expect that the results of this work could be contributed to study on the optimum design of subband codecs using vector quantizer.

• The KIPS Transactions:PartC
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• v.12C no.6 s.102
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• pp.881-890
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• 2005

Over for the past 10 years, the increase in geometric progression for the internet traffic, has allowed the IP protocol framework to be the most important network technology. In addition, the internet service is being developed as a service mode differentiated, aiming to support the new-mode real-time multimedia services such as internet phone, video conference, cyber reality, and internet game, focusing on offering a latest service. These days, aiming to solve the need for broad bandwidth along with guaranteeing QoS, the WDM technology of offering multiple gigabit wavelengths is emerging as the core technology of next-generation optical internet backbone network. In the next-generation optical internet backbone network based on WDM, the QoS framework is one of fore subjects aiming to offer a service of guaranteeing QoS This study analyzes the requirements of performance related to QoS framework in IP Subnet and in WDM optical backbone network, and suggests optical QoS service framework differentiated. in order to guarantee end-to-end QoS through the next-generation optical internet backbone network, using GMPLS control protocol.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.2
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• pp.327-334
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• 2016

In this paper, we propose a hardware architecture of transform and quantization for high-perfornamce HEVC(High Efficiency VIdeo Coding) encoder. HEVC transform decides the transform mode by comparing RDCost to search for the best mode of them. But, RDCost is computed using the bit-rate and distortion which is computed by transform, quantization, de-quantization, and inverse transform. Due to the many calculations and encoding time, it is hard to process high resolution and high definition image in real-time. This paper proposes the method of transform mode decision by comparing sum of coefficient after transform only. We use BD-PSNR and BD-Bitrate which is performance indicator. Based on the experimental result, We confirmed that the decision of transform mode can process images with no significant change in the image quality. We reduced hardware area by assigning different values at the same output according to the transform mode and overlapping coefficient multiplied as much as possible. Also, we raise performance by implementing sequential pipeline operation. In view of the larger process that we used compared with the process of reference paper, Our design has reduced by half the hardware area and has increased performance 2.3 times.

• Journal of the Korea Society of Computer and Information
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• v.18 no.11
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• pp.1-12
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• 2013

In multi-core processors, Last Level Cache(LLC) can reduce the speed gap between the memory and the core. For this reason, LLC has big impact on the performance of processors. LLC is composed of shared cache and private cache. In computer architecture community, most researchers have mainly focused on the management techniques for shared cache, while management techniques for private cache have not been widely researched. In conventional private LLC, memory is statically assigned to each core, resulting in serious performance degradation when the workloads are not fairly distributed. To overcome this problem, this paper proposes the replacement policy for managing private cache of LLC efficiently. As proposed core-aware cache replacement policy can reconfigure LLC dynamically, hit rate of LLC is increases drastically. Moreover, proposed policy uses 2-bit saturating counters to improve the performance. According to our simulation results, the proposed method can improve hit rates by 9.23% and reduce the access time by 12.85% compared to the conventional method.

• Journal of Broadcast Engineering
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• v.24 no.1
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• pp.118-131
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• 2019

In order to provide 8K UHD contents of terrestrial broadcasting with a large capacity, the terrestrial broadcasting system has various problems such as limited bandwidth and so on. To solve these problems, UHD contents transmission technology has been actively studied, and an 8K UHD broadcasting system using terrestrial broadcasting network and communication network has been proposed. The proposed technique is to solve the limited bandwidth problem of terrestrial broadcasting network by segmenting 8K UHD contents and transmitting them to heterogeneous networks through hierarchical separation. Through the terrestrial broadcasting network, the base layer corresponding to FHD and the additional enhancement layer data for 4K UHD are transmitted, and the additional enhancement layer data corresponding to 8K UHD is transmitted through the communication network. When 8K UHD contents are provided in such a way, user can receive up to 4K UHD broadcasting by terrestrial channels, and also can receive up to 8K UHD additional communication networks. However, in order to transmit the 4K UHD contents within the allocated bit rate of the domestic terrestrial UHD broadcasting, the compression rate is increased, so a certain level of image deterioration occurs inevitably. Due to the nature of UHD contents, video quality should be considered as a top priority over other factors, so that video quality should be guaranteed even within a limited bit rate. This requires packet scheduling of content generators in the broadcasting system. Since the multiplexer sends out the packets received from the content generator in order, it is very important to make the transmission time and the transmission rate of the process from the content generator to the multiplexer constant and accurate. Therefore, we propose a variable transmission scheduler between the content generator and the multiplexer to guarantee the image quality of a certain level of UHD contents in this paper.

• 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.