• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.05a
• /
• pp.887-890
• /
• 2009

To transmit information over a channel in the presence of noise, there needs some technique to code the information. One of the coding techniques used for error detection and correction close to the Shannon limit is Low Density Parity Code. LDPC and decoding characteristic features by sum-product algorithm are matched for the performance to Turbo Code, RA(Repeat Accumulate) code, in case of very long code length of LDPC surpass their performance. This paper explains LDPC coding scheme of image data and decoding scheme, implements LDPC decoder in FPGA.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.18 no.1
• /
• pp.15-29
• /
• 2024

A fetoscope is an optical endoscope, which is often applied in fetoscopic laser photocoagulation to treat twin-to-twin transfusion syndrome. In an operation, the clinician needs to observe the abnormal placental vessels through the endoscope, so as to guide the operation. However, low-quality imaging and narrow field of view of the fetoscope increase the difficulty of the operation. Introducing an accurate placental vessel segmentation of fetoscopic images can assist the fetoscopic laser photocoagulation and help identify the abnormal vessels. This study proposes a method to solve the above problems. A novel encoder-decoder network with a dual-path structure is proposed to segment the placental vessels in fetoscopic images. In particular, we introduce a channel attention mechanism and a continuous convolution structure to obtain multi-scale features with their weights. Moreover, a switching connection is inserted between the corresponding blocks of the two paths to strengthen their relationship. According to the results of a set of blood vessel segmentation experiments conducted on a public fetoscopic image dataset, our method has achieved higher scores than the current mainstream segmentation methods, raising the dice similarity coefficient, intersection over union, and pixel accuracy by 5.80%, 8.39% and 0.62%, respectively.

• Journal of Communications and Networks
• /
• v.1 no.3
• /
• pp.182-188
• /
• 1999

The optimal joint maximum-likelihood sequence estima-for using an array of antennas is derived for synchronous direct sequence-code division multiple access (DS-CDMA) system. Each user employs a rate 1/n convolutional code for channel coding for the additive white Gaussian noise (AWGN) channel. The array re-ceiver structure is composed of beamformers in the users' direc-tions followed by a bank of matched filters. The decoder is imple-mented using a Viterbi algorithm whose states depend on the num-ber of users and the constraint length of the convolutional code. The asymptotic array multiuser coding gain(AAMCG)is defined to encompass the asymptotic multiuser coding gain and the spatial information on users' locations in the system. We derive the upper and lower bounds of the AAMCG. As an example, the upper and lower bounds of AAMCG are obtained for the two user case where each user employes the maximum free distance convolutional code with rate 1/2. The enar-far resistance property is also investigated considering the number of antenna elements and user separations in the space.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.8 no.8
• /
• pp.1613-1619
• /
• 2004

Turbo codes of long block sizes have been known to show very good performance in an AWGN channel and the turbo code has been strongly recommended as error correction code for W-CDMA in 3GPP(3rd Generation Partnership Project). Recently, turbo codes of short block sizes suitable for real time communication systems have attracted a lot of attention. Thus, in this paper we consider the turbo code of 1/3 code rate and short frame size of 192 bits in ITU-R channel model. We analyzed the performance of W-CDMA systems of 10MHz bandwidths employing RAKE receiver with not only MRC diversity but also SOVA-based turbo code.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
• /
• v.4 no.1
• /
• pp.20-27
• /
• 1993

In this paper, a model of TCM / M-PSK with set partitioning and a model of the combined M-ary PSK system with pragmatic coding for digital radio communication are realized. The equations of error probability for TCM / M-PSK system and the combined M-ary PSK system with pragmatic coding are derived on the conditions of the Rayleigh fading with the AWGN. It is found that the combined M-ary PSK systemwith pragmatic coding in the AWGN channel can not be applied to the fading channel since uncoded bits cause parallel:parallel paths in the trellis diagram to degrade the performance of the system. However, the use of pragmatic coding in the AWGN channel could simplify the given system since single convolutional encoder / decoder is required.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.9 no.6
• /
• pp.768-775
• /
• 1998

The equalizer is the central part of the receiver and its performance significantly affects the overall performance of the system in the mobile communication. A proposed equalizer is composed of the channel estimator, MLSE based on the Viterbi algorithm and GMSK decoder. The approximation of GMSK with QPSK has great impact on the equalizer design, because it allows us to use the existing simple and efficient algorithms for designing optimal QPSK equalizer. In order to estimate efficiently channel, we use a sliding window algorithm based on energy calculation and cross-correlator. And also a tuning scheme is presented in order to improve the equalizer performance. Simulation results indicate that a proposed equalizer meets the GSM standards easily in terms of performance.

• ETRI Journal
• /
• v.26 no.5
• /
• pp.375-383
• /
• 2004

This paper proposes a common interleaving method associated with independent channel-encoding among transmitter antenna branches in orthogonal frequency and code division multiplexing based on multiple-input multiple-output (MIMO) multiplexing to achieve an extremely high throughput such as 1 Gbps using a 100 MHz bandwidth. This paper also investigates the average packet error rate performance as a function of the average received signal energy per bit-to-background noise power spectrum density ratio $(E_b/N_0)$. We found that the loss in the required average received $E_b/N_0$ of the proposed method is only within approximately 0.3 dB in up to a 12-path Rayleigh fading channel, using 16QAM and Turbo coding with a coding rate of 5/6. We also clarify that even for a large fading correlation among antenna branches, 1 Gbps is still possible by increasing the transmission power. Therefore, the proposed method reduces the processing rate to 1/4 in the turbo decoder with only a slight loss in the required average received $E_b/N_0$.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.43 no.3 s.309
• /
• pp.9-16
• /
• 2006

Many image and video compression algorithms work by splitting the image into blocks and producing variable-length code bits for each block data. If variable-length code data are transmitted consecutively over error-prone channel without any error protection technique, the receiving decoder cannot decode the stream properly. So the standard image and video compression algorithms insert some redundant information into the stream to provide some protection against channel errors. One of redundancies is resynchronization marker, which enables the decoder to restart the decoding process from a known state in the event of transmission errors, but its usage should be restricted not to consume bandwidth too much. The Error Resilient Entropy Code(EREC) is well blown method which can regain synchronization without any redundant information. It can work with the overall prefix codes, which many image compression methods use. This paper proposes EREREC method to improve FEREC(Fast Error-Resilient Entropy Coding). It first calculates initial searching position according to bit lengths of consecutive blocks. Second, initial offset is decided using statistical distribution of long and short blocks, and initial offset can be adjusted to insure all offset sequence values can be used. The proposed EREREC algorithm can speed up the construction of FEREC slots, and can improve the compressed image quality in the event of transmission errors. The simulation result shows that the quality of transmitted image is enhanced about $0.3{\sim}3.5dB$ compared with the existing FEREC when random channel error happens.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.11 s.353
• /
• pp.8-15
• /
• 2006

In this paper, we propose the iterative receiver for LDPC-coded OFDMA systems in uplink environments. Applying the Wiener filtering to pilot symbols, an initial channel estimation can be performed effectively. To reduce the complexity of the Wiener filtering, we approximate Wiener filtering coefficients to pre-determined coefficients according to estimated correlation of channel. After an LDPC decoding process, soft symbol derived by extrinsic information of decoder outputs is used to estimate channel. we also derive the error variance of channel estimation and maximum ratio combined results. Using combined results, the channel correlation is re-estimated. Then the proper Wiener filtering coefficients are chosen according to the re-estimated result of the channel correction. Using a computer simulation, we show that the proposed receiver structure has the better performance than the receiver using only pilot symbols.

• Journal of Korea Multimedia Society
• /
• v.12 no.9
• /
• pp.1201-1209
• /
• 2009

Recently, the convergence of the telecommunications industry through the development of network media has brought along interactive data services in addition to the traditional broadcast services by providing direct connections between the users and the broadcasters. This can be easily achieved by the combination of existing network technology and multimedia technology, but in reality, as seen in the actual implementation of IPVT, it's difficult to provide satisfactory levels of service to users due to issues of network bandwidth and problems of streaming system. In particular, a network which is able to fully support Qos/QoE, and also a fast enough channel changing technology that is satisfactory to the user, are prerequisites for IPTV success. Therefore this thesis proposes a fast channel changing technique to delivery enhanced IPTV user experience. Recent IP set-top-boxes use only a small portion of the CPU to feed Ethernet packets to the hardware decoder and play them. Thus, by making IP set-top-boxes share cached channel information and content with each other, reducing the load on the server and enhancing the channel changing time is made possible.