• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.3
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• pp.22-30
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• 2010

This paper proposes a combination system of Adaptive Modulation and Coding (AMC) and Multiple Input Multiple Output (MIMO), which improves the throughput and has a better reliability. In addition, the system includes Precoding, Antenna Subset Selection and MIMO Mode Selection scheme. Finally, we make a performance analysis of the proposed system. The principal environmental parameters for the simulation experiment consist of a frequency non-selective rayleigh fading channel and a Spreading Factor (SF) of 16. Other parameters may be included in order to fulfill the requirements of the HSDP A Standard. The proposed system has a higher throughput and more reliability than the conventional system, which does not include MIMO Mode Selection scheme, Precoding or Antenna Subset Selection. According to the simulation results, the proposed system reaches the maximum throughput at 8dB, presentlng an improvement of 6dB and twice higher throughput, respect to the conventional system. Specifically, at the point of -6dB, the conventional system reaches 2.5Mbps, while the proposed system reaches 6.4Mbps at the same SNR. Also, at the point of 2dB, each system reaches 7.5Mbps (conventional system) and 15.3Mbps (proposed system), with near twice the difference. According to the results exposed above, we can conclude that the system proposed in this paper has, as the greatest contribution, the improvement of the throughput, especially, the average throughput.

• Journal of the Korea Society of Computer and Information
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• v.19 no.3
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• pp.63-71
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• 2014

We analyze the execution time of Reed-Solomon coding, which is the MAC-layer forward error correction scheme used in CDMA2000 1xEV-DO broadcast services, under different air channel conditions. The results show that the time constraints of MPEG-4 cannot be guaranteed by Reed-Solomon decoding when the packet loss rate (PLR) is high, due to its long computation time on current hardware. To alleviate this problem, we propose three error control schemes. Our static scheme bypasses Reed-Solomon decoding at the mobile node to satisfy the MPEG-4 time constraint when the PLR exceeds a given boundary. Second, dynamic scheme corrects errors in a best-effort manner within the time constraint, instead of giving up altogether when the PLR is high; this achieves a further quality improvement. The third, video-aware dynamic scheme fixes errors in a similar way to the dynamic scheme, but in a priority-driven manner which makes the video appear smoother. Extensive simulation results show the effectiveness of our schemes compared to the original FEC scheme.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

• 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 Journal of Korean Institute of Communications and Information Sciences
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• v.39A no.4
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• pp.195-205
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• 2014

The ultra-high definition television (UHDTV) has been considered as a next generation broadcsating service. However the conventional digital terrestrial transmission system cannot afford the required transmission data rate of UHDTV, and thus adopting ultra-high order constellation, such as 4096-QAM, into the conventional DTT systems has been studied. In particular, when the ultra-high order constellation is adopted into the digital video broadcasting-2nd generation terrestrial (DVB-T2) unequal-error protection (UEP) properties of a codeword of an error correction coding and ultra-high order constellations should be properly matched by bit mapper in order to enhance the decoding performance. Because long codeword results in a heavy computational complexity to design the bit mapper, the DVB-T2 divided it into cascaded blocks, the bit interleaver and the bit-to-cell DEMUX, and there have been many researches related to each block. However, there are few published study related to design methodology of bit interleaver. In this respect, this paper proposes a design methodology of the bit interleaver and presents bit interleavers of 1024-QAM and 4096-QAM according to the proposed design algorithm. The newly designed interleavers improved the decoding performance of the error correction coding by maximally 0.6 dB SNR over both of AWGN and random fading channel.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.9 s.351
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• pp.149-157
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• 2006

Turbo code, a kind of error correction coding technique, has been used in the field of digital mobile communication system. As the number of iterations increases, it can achieves remarkable BER performance over AWGN channel environment. However, if the number of iterations is increased in the several channel environments, any further iteration results in very little improvement, and requires much delay and computation in proportion to the number of iterations. To solve this problems, it is necessary to device an efficient criterion to stop the iteration process and prevent unnecessary delay and computation. In this paper, it proposes an efficient and simple criterion for stopping the iteration process in turbo decoding. By using variance values of LLR in turbo decoder, the proposed algerian can largely reduce the average number of iterations without BER performance degradation in all SNR regions. As a result of simulation, the average number of iterations in the upper SNR region is reduced by about $34.66%{\sim}41.33%$ compared to method using variance values of extrinsic information. the average number of iterations in the lower SNR region is reduced by about $13.93%{\sim}14.45%$ compared to CE algorithm and about $13.23%{\sim}14.26%$ compared to SDR algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.12B
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• pp.1565-1573
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• 2011

In this paper, we present a MIMO-OFDM broadband power line communication (BPLC) for Smart Grid and its associated applications and analyze its performance over the 3-phase 4-wire power line channels. For practical BPLC system simulation, we adopt the statistical power line fading channel model instead of the existing deterministic fading channel models (Zimmermann model, MTL model, and so on). In this paper, we implement $2{\times}2$ and $3{\times}3$ MIMO schemes using 3-phase 4-wire power lines. We investigate the capacity loss and BER performance of the proposed MIMO system by considering the effect of cross-talk between antenna paths. We choose space-frequency coding in order to reduce frequency interference between subcarriers and employ maximum ratio combining (MRC) that achieves both multiple antenna path diversity gain and multiple fading path diversity gain. We evaluate the proposed system performance through computer simulation in terms of the impulse noise index and the capacity loss ratio and compare the different signal combining schemes including MRC, equal gain combing (EGC), and selection combining (SC).

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.7 s.349
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• pp.143-149
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• 2006

Turbo code, a kind of error correction coding technique, has been used in the field of digital mobile communication system. As the number of iterations increases, it can achieves remarkable BER performance over AWGN channel environment. However, if the number of iterations is increased in the several channel environments, any further iteration results in very little improvement, and requires much delay and computation in proportion to the number of iterations. To solve this problems, it is necessary to device an efficient criterion to stop the iteration process and prevent unnecessary delay and computation. In this paper, it proposes an efficient and simple criterion for stopping the iteration process in turbo decoding. By using sign changed ratio of extrinsic information values in turbo decoder, the proposed algorithm can largely reduce the average number of iterations without BER performance degradation. As a result of simulations, the average number of iterations is reduced by about $12.48%{\sim}22.22%$ compared to CE algorithm and about $20.43%{\sim}54.02%$ compared to SDR algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10C
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• pp.1387-1394
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• 2004

Turbo code, a kind of error correction coding technique, has been used in the field of digital mobile communication systems. And it is well known about the fact that turbo code has better the BER performance as the number of decoding iterations increases in the AWGN channel environment. However, as the number of decoding iterations is increased under the several channel environments, any further iteration results in very little improvement, and it requires much delay, computation and power consumption in proportion to the number of decoding iterations. In this paper, it proposes the efficient iterative decoding stop criterion algorithm which can largely reduce the average number of decoding iterations of turbo code. Through simulations, it is verifying that the proposed algorithm can efficiently stop the iterative decoding by using the variance value of error probability for the soft output value, and can largely reduce the average number of decoding iterations without BER performance degradation. As a result of simulation, the average number of decoding iterations for the proposed algorithm is reduced by about 2.25% ~14.31％ and 3.79％ ~14.38% respectively compared to conventional schemes, and power consumption is saved in proportion to the number of decoding iterations.

• Journal of Navigation and Port Research
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• v.39 no.4
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• pp.293-297
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• 2015

Underwater acoustic communications has been limited use for military purposes in the past. However, the fields of underwater applications expend to detection, submarine and communication in recent. The excessive multipath encountered in underwater acoustic communication channel is creating inter symbol interference, which is limiting factor to achieve a high data rate and bit error rate performance. To improve the performance of a received signal in underwater communication, many researchers have been studied for channel coding scheme with excellent performance at low SNR. In this paper, we applied BCJR decoder based ( 2,1,7 ) convolution codes and to compensate for the distorted data induced by the multipath, we applying the turbo equalization method. Through the underwater experiment on the Gyeungcheun lake located in Mungyeng city, we confirmed that turbo equalization structure of BCJR has better performance than hard decision and soft decision of Viterbi decoding. We also confirmed that the error rate of decoder input is less than error rate of $10^{-1}$, all the data is decoded. We achieved sucess rate of 83% through the experiment.