• Journal of the Institute of Convergence Signal Processing
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• v.2 no.2
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• pp.62-67
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• 2001

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need a large time delay over the reception process. Moreover, Turbo Code has been known as the robust ending method with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications such as IMT-2000. Therefore, in this paper, we proposed of the method to improve the conventional performance with the parallel concatenated 4-New Turbo Decoder using MAP a1gorithm in spite of complexity increasement. In the real-time video and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and fading channels.

• Journal of Broadcast Engineering
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• v.22 no.1
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• pp.15-27
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• 2017

CRA(Conditional Replenishment Algorithm) was proposed to improve the visual quality of SC-MMH(Service Compatible 3DTV using Main and Mobile Hybrid delivery) which is a kind of hybrid 3DTV system and has been standardized by ATSC. In SC-MMH system, reference view and additional view may have different resolutions and/or encoding methods. To reconstruct 3D view, additional view needs to be enlarged as same as reference view. Although the performance of CRA is quite satisfactory, there may exist some blocky artifacts in the enlarged view since it adopts block-shaped processing unit with quad-tree structure. In this paper, we analyze the main causes of blocky artifacts in CRA and show these artifacts can be successfully suppressed by applying the deblocking filter at receiver side.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.135-143
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• 1998

Several equalizers are applied to the DVD-ROM(Digital Versatile Disc-Read Only Memory) system. Because imperfect writing process may cause nonlinear distortion in the reply signal, neural equalizers which have strong nonlinear mapping capability are applied to the system to compensate the nonlinear distortion. Experimental results to verify that the combination of decision-feedback type equalizers and modulation code is formidable are also given. The experimental results shwo that the SNR gain of the neural equalizers over the conventional equalizers becomes much as the nonlinearity in the channel increases. Furthermore, the decision-feedback type equalizers outperform the equalizers which have no decision-feedback in eliminating ISI(Intersymbol Interference) of random data sequence but there is no performance gain of the decision-feedback type equalizers over the equalizers without decision-feedback when these are applied to compensate the ISI of modulation-encoded data sequence.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.6
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• pp.54-62
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• 2007

In this paper, we significantly reduces the amount of computation for intra 16$\times$16 mode decision in H.264 by applying the fast algorithm, which obtains the transformed prediction residual with fewer computations. By extending the existing intra 4$\times$4 mode decision, we propose the new algorithm for fast intra 16$\times$16 mode decision. The proposed algorithm uses partial transform coefficients which consist of one DC and three adjacent AC coefficients after 4$\times$4 transform in the intra 16$\times$16 mode decision. Theoretical analysis and experimental results show that the proposed algorithm can reduce computations up to 50% in the intra 16$\times$16 mode decision process with unnoticeable degradation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1579-1588
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• 2014

The LAN method is the most widely used in domestic high-speed internet access and rapidly moving to 1 Gbps Ethernet from 100 Mbps one to provide high-speed services such as UHD TV. The 1000BASE-T PHY with 4 pairs UTP transmits a PAM-5 signal at the 125 MHz clock per each pair to achieve 1 Gbps rate. In order to correct errors over the channel, the transmitter uses a TCM which is combined the convolutional encoder and PAM-5, and the receiver uses the Viterbi decoder. In this paper, we implement a Viterbi decoder which can correct two pair errors and operate at the least 125 MHz clock speed. Finally, we will verify the error correction function and the operating speed of the implemented decoder with a logic analyzer.

• The Transactions of the Korea Information Processing Society
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• v.2 no.5
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• pp.677-690
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• 1995

In this paper, we propose the high efficiency data compression capable of error control using block-sorting, move to front(MTF) and arithmetic code with variable length in to fixed out. First, the substring with is parsed into length N is shifted one by one symbol. The cyclic shifted rows are sorted in lexicographical order. Second, the MTF technique is applied to get the reference of locality in the sorted substring. Then the preprocessed sequence is coded using VF(variable to fixed) arithmetic code which can be limited the error propagation in one codeword. The key point is how to split the fixed length codeword in proportion to symbol probabilities in VF arithmetic code. We develop the new VF arithmetic coding that split completely the codeword set for arbitrary source alphabet. In addition to, an extended representation for symbol probability is designed by using recursive Gray conversion. The performance of proposed method is compared with other well-known source coding methods with respect to entropy, compression ratio and coding times.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2C
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• pp.57-64
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• 2011

A two-way relay channel is a bidirectional cooperative communication channel between two nodes using a relay. In many cooperative communication schemes, a relay transmits its data to each node using separate channels. However, in the two-way relay channel, a relay can broadcast the network-coded signal to both nodes in a same time slot, which can increase the system throughput. In this paper, a new cooperative network error correcting scheme using distributed turbo code in a two-way relay channel is proposed. The proposed scheme not only increases the system throughput using network code but also improves the performance by utilizing the LLR information from relay node and other user node through distributed turbo code. Also, a power allocation scheme is investigated for various channel conditions to improve the system performance.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.6
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• pp.99-110
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• 2005

These days, many image processing techniques have been studied for effective image compression. Among those, The 2D image filtering is widely used for 2D image processing. The 2D image filtering can be implemented by performing the 1D linear filter separately in the horizontal and vertical direction. Efficiency of image compression depends on what filtering method is used. Generally, circular convolution is widely used in 2D image filtering for image processing. However it doesn't consider correlations at the boundary region of image, therefore effective filtering can not be performed. To solve this problem. I proposed new convolution technique using loop convolution which satisfies the 'alias-free' and 'error-free' requirement in the reconstructed image. This method could provide more effective compression performance than former methods because it used highly-correlated data when performed at the boundary region. In this paper, Sub-band Coding(SBC) was adopted to analyze efficiency of proposed filtering technique, and the simulator developed by Java-based language was used to examine the performance of proposed method.

• Journal of the Korea Society of Computer and Information
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• v.13 no.2
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• pp.193-201
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• 2008

Using Network Coding in P2P systems yields great benefits, e.g., reduced download delay. The core notion of Network Coding is to allow encoding and decoding at intermediate nodes, which are prohibited in the traditional networking. However, improper implementation of Network Coding may reduce the overall performance of P2P systems. Network Coding cannot work with general arithmetic operations, since its arithmetic is over a Finite Field and the use of an efficient Finite Field arithmetic algorithm is the key to the performance of Network Coding. Also there are other important performance parameters in Network Coding such as Field size. In this paper we study how those factors influence the performance of Network Coding based systems. A set of experiments shows that overall performance of Network Coding can vary 2-5 times by those factors and we argue that when developing a network system using Network Coding those performance parameters must be carefully chosen.