Earticle

본 논문에서는 송신부에서 다중 안테나를 사용할 때, 안테나 선택 과정에서 논리적 정보를 담아 전송하는 시스템이다. 송신부는 보내고자 하는 데이터 스트림 일부분을 안테나 선택 테이블에 적용하여 일부분을 제외한 데이터 스트림을 송 신한다. 수신부는 송신된 데이터 스트림과, 선택된 안테나 테이블에 의해 실제적으로는 송신되지 않은 데이터 스트림을 복구하여 송신부가 보내고자 했던 전체 데이터 스트림을 도출하는 시스템이다. 채널 용량을 최대화하기 위하여 SVD (Singular value decomposition)을 적용 후 송신할 안테나를 선택하였다. 선택하는 송신 안테나 수에 따라 달라지는 채널 용량을 실험하였으며, 안테나 선택에 의한 논리적 전송에 따라 증가하는 시스템 처리량을 확인하였다.

Wireless communication chipsets have fixed transmission rate and communication distance. Although there are many kinds of chipsets with throughput and distance purpose, they cannot support various types of wireless applications. This paper provides theoretic research results in order to support various wireless applications requiring different throughput, delayed quality-of-service (QoS), and different communication distances by using a wireless communication chipset with fixed rate and transmission power. As a performance metric, the probability for a data frame that successfully receives at a desired receiver is adopted. Based on this probability, the average number of transmission in order to make a successful frame transmission is derived. Equations are utilized to analyze the performance of a single-hop with channel coding and a dual-hop without error correction matter transmission system. Our results revealed that single-hop transmission assisted by channel coding could extend its communication distance. However, communication range extending effect of the single-hop system was limited. Accordingly, dual-hop transmission is needed to overcome the communication distance limit of a chipset.

The paper proposes reliable channel for medical uses via Physical Layer. The purpose is to serve medical data transmission from WBAN through the UMTS. This study is focusing on applying coding theory to carry high reliability and robustness, in order to satisfy various QoS requirements of medical data. The aim is to build Medical Network Channel that transmits WBAN medical data through the UMTS. The design of extra channel coding is considered as concatenated techniques. The method applies adaptive outer code with inner UMTS code. The accomplishment of QoS is investigated with numerical results proven under AWGN and Rayleigh fading.

In this paper we propose high rate turbo code (HRTC) that exploits the potential correlation present between the data of the neighboring sensor nodes in a wireless sensor network. The problem of implosion and overlap, faced by the sensor nodes, gives rise to data redundancy in the sensor network. This paper combines source -channel coding wi th data aggregation using HRTC to reduce the redundant information produced amongst the nodes and to make the network more resourceful and reliable. This paper also includes the design of the proposed HRTC that uses deterministic interleaver to ensure good permutations.

The concatenated codes or the convolutional codes which are often adopted in the communication systems are originally designed to overcome the random independent noise. As such, the performance is severely degraded over the bursty impulsive noise channels. In this paper we compare the bit error rate (BER) performance of the various different channel coding schemes with interleaver over the bursty impulsive noise channels. In particular, we focus on the existence of the bit level and/or the byte level interleavers and show from the extensive simulations that the performance can be different depending on the existence of the interleaver. From our simulations we can see that the concatenated codes with the byte interleaver between the inner and outer codes and the bit interleaver after the inner code gives the best performance among them. For the schemes which include the bit level interleaver, we also observe the interleaving gain versus the interleaver span so that it can be a good guideline to design the communication systems with the concatenated codes.

Many kinds of multi-purpose wireless communication chips exist. However, their transmission power, throughput and error correction schemes are fixed. It is hard to apply them to various wireless applications with various quality-of-service. However, bit error occurs frequently in that transmission speed, which results in corrupted digital image when we use the chip module for image transmission. Unfortunately, this chip module does not have any error correction functionality. Hence, transmission speed should be reduced for reliable wireless data transmission. In this paper, a channel coding scheme is adopted outside of the chip module. Especially, the convolutional code is adopted as an error correction code. When the channel coding is adopted, frame error probability is analyzed considering practical chip setup parameters. By using the channel coding added outside of the chip module, reliable image transmission becomes available. Using multiple chip modules and channel coding together, throughput enhancement with trusted wireless transmission can be implemented.

A new decoding algorithm that modifies error searching soft decision method is presented. The proposed algorithm improves the decoding error performance caused by repeated searching. It is shown that the simplicity inherent in the hard decision decoding is still maintained. This paper proposes an error correcting binary channel code which promotes symbol transition. This scheme exploits the error correcting capability of the preparatory code. It is demonstrated that the proposed code combine good symbol transition property with their own error correcting capability.

In this paper, delay-limited transmission of Quasi-stationary sources over block fading channels is considered. Considering distortion outage probability as the performance measure, two source and channel coding schemes with power adaptive transmission are presented. The delay limited transmission of a quasi-stationary source over a block fading channel is from the perspective of source and channel coding design and it utilizes performance scaling laws. Because In a Quasi -stationary sources time utilization takes a major role. For that the first one is optimized for fixed rate transmission, and hence enjoys simplicity of implementation. The second one is a high performance scheme, which also benefits from optimized rate adaptation with respect to source and channel states. In high SNR regime, the performance scaling laws in terms of outage distortion exponent and asymptotic outage distortion gain are derived, where two schemes with fixed transmission power and adaptive or optimized fixed rates are considered as benchmarks for comparisons. Various analytical and numerical results are provided which demonstrate a superior performance for source and channel optimized rate and power adaptive scheme.

A selection method of possible code words in channel coding systems is presented. The proposed method is for channel coding of digital communication systems. Measuring the error correction capability with reunion junction, the proposed method has superior performance and low implementation complexity.

In this paper the model having D8PSK modulation applied by channel coding of Convolution Turbo Code (CTC) was designed and analyzed to improve the performances of wireless communication system for data link between ground equipment and aircraft. The suggested modem was differential 8PSK modulation system and channel coding of Convolution Turbo Code (CTC) system, and TDMA system was used for transmission. The hardware was embodied for the test and done by using 2 FPGA’s and DSP. The function of FPGA has the one of modulation and demodulation of signal, acquisition and maintaining of TDMA synchronization and generating timing signal. As the test results, it was recognized that the transmission rate can be maintained though the code rate is increased based on 28.8 Kbps and the performance of SNR gets 15dB when BER performance criteria is 10-6. And, it was recognized that the performance of BLER satisfies 10-4 based on BER 10-6. And, it was also recognized that the performance of SNB was improved by 4dB comparing with the case which channel coding was not used from the test results.

In this paper, we proposed a progressive time varying source channel coding system for transmitting image over wireless channels. Transmission of compressed image data over noisy channel is an important problem and has been investigated in a variety of scenarios. the core results obtained by a systematic method of instantaneous rate allocation between the progressive source coder and progressive channel coder .It is developed by closed form of expression for end-to-end distortion , rate allocation respectively in static channels. It is extended the static result to an algorithm for fading channels. It is introduced set DCT (blocks) approach is adapted to perform sub band decomposition followed by SPIHT (Set partitioning in Hierarchical tree)

This paper introduces a new paradigm of physical layer security through channel coding for wireless networks. The well known spread spectrum based physical layer security in wireless network is applicable when code division multiple access (CDMA) is used as wireless air link interface. In our proposal, we incorporate the proposed security protocol within channel coding as channel coding is an essential part of all kind of wireless communications. Channel coding has a built-in security in the sense of encoding and decoding algorithm. Decoding of a particular codeword is possible only when the encoding procedure is exactly known. This point is the key of our proposed security protocol. The common parameter that required for both encoder and decoder is generally a generator matrix. We proposed a random selection of generators according to a security key to ensure the secrecy of the networks against unauthorized access. Therefore, the conventional channel coding technique is used as a security controller of the network along with its error correcting purpose.

In this paper, we analyze the performance of transmitter and receiver of W-CDMA system using the channel coding and space time code in downlink over ITU-R realistic channel model. We can improve the data rate and the reliability of communications by a channel code, and we can get a diversity gain by using the space time code over fading channels. We also apply the RAKE receiver to improve a performance in multi-path fading channel environment.

In this paper, we analyze the performance of transmitter and receiver of W-CDMA system using the channel coding and space time code in downlink over ITU-R realistic channel model. We can improve the reliability of communications and increase the data rate by channel coding, and we can get a diversity gain by using the space time code over fading channels. We also apply the RAKE receiver to improve tile performance in multi-Path fading channel environment.

This paper proposes a new joint channel coding algorithm based on principal component analysis. A conventional joint channel coder using passive downmixing undergoes a reduction of both the primary-to-ambient energy ratio (PAR) of the downmix signal and the panning gain ratio of the primary source. The proposed system preserves the PAR of the downmix signal by using active downmixing which reflects spatial characteristic. The proposed system also improves the accuracy of the panning gain ratio estimation. Computer simulations and subjective listening tests verify the performance of the proposed system.

This paper proposes a coding scheme of control channel for mobile packet communication to maximize the minimum Hamming distance, which is based on shifting of basis vectors of Reed Muller code with optimized dynamic puncturing and/or(partial) sequence repetition. Since the maximum likelihood decoding can be performed using the extremely simple FHT(Fast Hadamard Transformation), it is suitable for real time optimum decoding of control channel information with very little complexity. We show applications of the proposed coding method to TFCI(Transport Format Combination Indicator) code in split and normal modes of 3GPP W-CDMA system. We also discuss how this method can reduce rate indication error over AWGN(Additive White Gaussian Noise) as well as fading channels when the proposed coding scheme is applied to 1xEV-DV system of $3^{rd}$ generation partnership project 2(3GPP2) to indicate the data rate transmitted on the reverse traffic channel by a Mobile Station(MS).

The scalable extension of the H.264/AVC video coding standard (SVC) demonstrates superb adaptability in video communications. Joint source and channel coding (JSCC) has been shown to be very effective for such scalable video consisting of parts of different significance. In this paper, a new JSCC scheme for SVC transmission over packet loss channels is proposed which performs two-dimensional optimization on the quality layers of each frame in a rate-distortion (R-D) sense as well as on the temporal hierarchical structure of frames under dependency constraints. To compute the end-to-end R-D points of a frame, a novel reduced trellis algorithm is developed with a significant reduction of complexity from the existing Viterbi-based algorithm. The R-D points of frames are sorted under the hierarchical dependency constraints and optimal JSCC solution is obtained in terms of the best R-D performance. Experimental results show that our scheme outperforms the existing scheme of [13] with average quality gains of 0.26 dB and 0.22 dB for progressive and non-progressive modes respectively.

Recently, realistic visual services such as 3DTV, UHDTV, 3D cinema, 3D gaming, augmented reality (AR), virtual reality (VR) and so on are introduced, and realistic audio services are getting more important to make the realistic visual services more realistic. Generally, the realistic audio services have a limitation that users should have multi-channel speaker layout for generating the realistic audio sound, so the binaural sound generation method that provides the users the realistic sound through a stereo headphone has been introduced. But, since the binaural sound has a constant sound scene, it can not reflect the users’ movement in the AR or the VR service. In this paper, we proposed a sound scene control scheme of binaural sound in the multi-channel audio coding for generating a more realistic audio sound in the VR or AR service. In the proposed method, multi-channel gain factors and a stereo down-mix signal are modified based on the new multi-channel layout to be arranged according to a user’s head movement. The experimental results show that the proposed sound scene control scheme of the binaural sound in the multi-channel audio coding can successfully generate the binaural sound with the controlled sound scene while maintaining the good sound quality.